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.Globaliv