From infos at madovsky.org  Mon Aug  1 00:13:55 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 31 Jul 2011 16:13:55 -0400
Subject: [Freeswitch-users] mod_rtmp question
Message-ID: <794224500D9C43C3860909EE801A3DC8@e1705>

I use freeswitch in ODBC mode.
in sofia endpoint if I type
"sofia status profile internal"
I get all registrations from all nodes

but in rtmp endpoint
"rtmp status profile internal reg"

I get registrations of the local node.

is it a normal behaviour ?
if yes which rtmp command can give the same
result of sofia with ODBC ?

Thanks

Franck





-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110731/594c8336/attachment.html 

From bracken_dave at yahoo.com  Mon Aug  1 09:02:25 2011
From: bracken_dave at yahoo.com (Dave Bracken)
Date: Sun, 31 Jul 2011 22:02:25 -0700 (PDT)
Subject: [Freeswitch-users] 3
Message-ID: <1312174945.61733.yint-ygo-j2me@web114509.mail.gq1.yahoo.com>

Make the right choice.. http://kodaifresh.com/page.php?mgoogleId=69h8


From krice at freeswitch.org  Mon Aug  1 09:05:15 2011
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 01 Aug 2011 00:05:15 -0500
Subject: [Freeswitch-users] 3
In-Reply-To: <1312174945.61733.yint-ygo-j2me@web114509.mail.gq1.yahoo.com>
Message-ID: <CA5B9E3B.19FFB%krice@freeswitch.org>


Hey Dave,

Your email got hax0red

On 8/1/11 12:02 AM, "Dave Bracken" <bracken_dave at yahoo.com> wrote:

> Make the right choice.. http://SPAMMER/page.php?mgoogleId=69h8
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From steveayre at gmail.com  Mon Aug  1 11:48:47 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 1 Aug 2011 08:48:47 +0100
Subject: [Freeswitch-users] mod_rtmp question
In-Reply-To: <794224500D9C43C3860909EE801A3DC8@e1705>
References: <794224500D9C43C3860909EE801A3DC8@e1705>
Message-ID: <CAFiqYu=zOqWe6Gxecg74WCr_LXq2Ca0JoE8dbGJpDrk-5U=6nw@mail.gmail.com>

That's because mod_sofia stores registrations in a the database (which in
your case must be a shared ODBC one) while mod_rtmp currently has everything
in memory. Yes it's expected, and there's no workaround at the moment other
than running the ESL command on all nodes.

Don't assume all modules work in the same way...

-Steve



On 31 July 2011 21:13, Madovsky <infos at madovsky.org> wrote:

> **
> I use freeswitch in ODBC mode.
> in sofia endpoint if I type
> "sofia status profile internal"
> I get all registrations from all nodes
>
> but in rtmp endpoint
> "rtmp status profile internal reg"
>
> I get registrations of the local node.
>
> is it a normal behaviour ?
> if yes which rtmp command can give the same
> result of sofia with ODBC ?
>
> Thanks
>
> Franck
>
>
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110801/b73b475e/attachment-0001.html 

From michael.knop at hcu-hamburg.de  Mon Aug  1 13:52:41 2011
From: michael.knop at hcu-hamburg.de (michael knop)
Date: Mon, 01 Aug 2011 11:52:41 +0200
Subject: [Freeswitch-users] ptime
Message-ID: <4E367769.9000602@hcu-hamburg.de>

Hi all,

I?m using FreeSWITCH Version 1.0.head (git-0fc8050 2011-07-31 22-14-06 
-0500) and I?m dealing with a problem 
<http://pastebin.freeswitch.org/16913> that sounds like the one that is 
described in the Jira bug report 
<http://jira.freeswitch.org/browse/FS-2003>.

Is the patch to reset the timestamp detector every time FreeSWITCH gets 
a new sdp or reply to invite still active?

Thanks,

micha


From dhairya.blogs at gmail.com  Mon Aug  1 08:51:25 2011
From: dhairya.blogs at gmail.com (Dhairya Vora)
Date: Mon, 1 Aug 2011 10:21:25 +0530
Subject: [Freeswitch-users] unable to connect to freeswitch server using
	x-lite
In-Reply-To: <CAKzWOxU=KFh8wcGHrypZKQ5oJ7X0Ua9fmDjuDabNAF==t2h-3w@mail.gmail.com>
References: <CAHq18AxZ1Ky_5RhaOpANTWkAhZCOuKhmoe3dpd49VWCk67V5Vw@mail.gmail.com>
	<CAKzWOxU=KFh8wcGHrypZKQ5oJ7X0Ua9fmDjuDabNAF==t2h-3w@mail.gmail.com>
Message-ID: <CA+P_fE_CQMhmvRgCr1f1qM=ihCsSPtgZTS7Ay47X+Y2ejNs66A@mail.gmail.com>

Thanks a lot....
It worked....
I initially just switched of the selinux from
/var/log/nginx/ and making it
SELINUX=disabled

but now (after doing lokkit), I could connect.

Thanks a lot.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110801/d718f7f2/attachment.html 

From bino at indoakses-online.com  Mon Aug  1 11:29:35 2011
From: bino at indoakses-online.com (bino oetomo)
Date: Mon, 01 Aug 2011 14:29:35 +0700
Subject: [Freeswitch-users] How to for video call ?
Message-ID: <4E3655DF.4060207@indoakses-online.com>

Dear All ..
I just learn to setup a FreeSwitch server.
Well .. actualy I set it up using FusionPBX

Now .. I have to extention .. that is 2001 and 2003

2003 is my friends Mac station , using X-Lite software
2001 is my windows station using linphone.

Both station is set to use H263 as Video Codec.

I try to trace the video fail in my Freeswitch server, using :

root at lapp freeswitch/conf# tail -f ../log/freeswitch.log |grep FAIL 
|grep video

And here is the result

Dialplan: sofia/internal/2003 at fusionpbx.int Regex (FAIL) [video_record] 
destination_number(2001) =~ /^\*9193$/ break=on-false
Dialplan: sofia/internal/2003 at fusionpbx.int Regex (FAIL) 
[video_playback] destination_number(2001) =~ /^\*9194$/ break=on-false
=============================================
Dialplan: sofia/internal/2001 at fusinpbx.int Regex (FAIL) [video_record] 
destination_number(2003) =~ /^\*9193$/ break=on-false
Dialplan: sofia/internal/2001 at fusinpbx.int Regex (FAIL) [video_playback] 
destination_number(2003) =~ /^\*9194$/ break=on-false

The first 2 part is happen when 2003 dial 2001, and the other 2 part is 
vice versa.


Here is a snippet of my vars.xml :
root at lapp freeswitch/conf# more vars.xml

<!-- Codecs -->
<X-PRE-PROCESS cmd="set" 
data="global_codec_prefs=G722,PCMU,PCMA,GSM,H263,H264,H261,H263-1998,H263-2000"/>
<X-PRE-PROCESS cmd="set" 
data="outbound_codec_prefs=G722,PCMU,PCMA,GSM,H263,H264,H261,H263-1998,H263-2000"/>


Kindly please tell me what to check / edit.

Sincerely
-bino-




From jcgpoza at gmail.com  Mon Aug  1 18:17:01 2011
From: jcgpoza at gmail.com (Dissident)
Date: Mon, 1 Aug 2011 07:17:01 -0700 (PDT)
Subject: [Freeswitch-users] Sip Headers advice (Not parsing properly)
In-Reply-To: <1311249742856-6606461.post@n2.nabble.com>
References: <1311249742856-6606461.post@n2.nabble.com>
Message-ID: <1312208221935-6641160.post@n2.nabble.com>

Problem solved!!!

 It seems that this doc is misleading ->
http://wiki.freeswitch.org/wiki/Channel_Variables#Channel_Variable_Manipulation
not all sip headers are gonna be parsed only the ones which start with X- or
P- the rest are not parsed because they are not supposed to exist, however
in my case they do and the only solution I've found to parse my CISCO
headers is
modifying the mod_sofia, file sofia.c line 6515, I added the last OR
condition.

 } else if (!strncasecmp(un->un_name, "X-", 2) || !strncasecmp(un->un_name,
"P-", 2)
   || !strncasecmp(un->un_name, "Cisco", 5)) {

and now I can retrieve this header from the dialplan just like this:

        ${sip_h_Cisco-Guid}"/>

Thanks to Swk.






--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Sip-Headers-advice-Not-parsing-properly-tp6606461p6641160.html
Sent from the freeswitch-users mailing list archive at Nabble.com.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110801/7e7a36a5/attachment.html 

From msc at freeswitch.org  Mon Aug  1 19:24:16 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 1 Aug 2011 08:24:16 -0700
Subject: [Freeswitch-users] Help setting up SIP reg
In-Reply-To: <CAE=3qObVuLOhp2GD29Da-GfzB_qkbkyDP4kPmRxjdbJi7XeirQ@mail.gmail.com>
References: <CAE=3qOYPpTuRTvq7RTJeUgzsCtvrfCjMiKFGu1fg0PHyZRE+-g@mail.gmail.com>
	<CAFiqYu=c8mY5BMWPumy1CgZHZ81kMRPgoNiF-6mjFAkW1q30fA@mail.gmail.com>
	<CAE=3qOZxwERT3CQEFHetO_o2s=aQUwYZ54TmMRPx9eYD=A7NoQ@mail.gmail.com>
	<CAKzWOxUEsGUKojF-vvstL6FvcDMhePqYGdXEQXQ4x-EJ81Bk6w@mail.gmail.com>
	<CAE=3qOacZh50iepxJKJD5UvtesxZAsN7RnPKNe1zdLO9VmFsuA@mail.gmail.com>
	<CAE=3qOZsNuiimFBZJTf4SbPRA8bRq_hJfPUunEnT8rxZj3qTnA@mail.gmail.com>
	<CAE=3qObVuLOhp2GD29Da-GfzB_qkbkyDP4kPmRxjdbJi7XeirQ@mail.gmail.com>
Message-ID: <CAKzWOxUyZoBS5Evp4b61vP8fm64Lef0n01ti3KmVraQO_BqYtA@mail.gmail.com>

Do they challenge you (digest auth) or do they have your IP address on a
white list? That's a critical piece of information that only your provider
can supply.

-MC

On Fri, Jul 29, 2011 at 9:31 PM, lloyd thomas <lloydie.t at gmail.com> wrote:

> OK Inbound working with:
>
>   <!--2circles -->
>     <list name="domains" default="allow">
>       <node type="allow" cidr="80.40.150.150/32"/>
>     </list>
>
> Just need to sort outbound.
>
>
> On 30 July 2011 04:59, lloyd thomas <lloydie.t at gmail.com> wrote:
>
>> Hi, dialling in produces the following error.
>>
>> 2011-07-30 04:56:07.818936 [DEBUG] sofia.c:6517 IP 80.40.150.150 Rejected
>> by acl "domains". Falling back to Digest auth.
>> 2011-07-30 04:56:07.826367 [WARNING] sofia_reg.c:1246 SIP auth challenge
>> (INVITE) on sofia profile 'internal' for [01869******@172.16.XXX.XXX] from
>> ip 80.40.150.150
>>
>>
>>
>> On 30 July 2011 04:34, lloyd thomas <lloydie.t at gmail.com> wrote:
>>
>>> I am registering with a them. I could not find suitable example in
>>> http://wiki.freeswitch.org/wiki/SIP_Provider_Examples which
>>>
>>>
>>> On 29 July 2011 21:57, Michael Collins <msc at freeswitch.org> wrote:
>>>
>>>> Are you registering with the provider or are they registering with you?
>>>> If they register with you then a user example is appropriate. If you are
>>>> registering with them then all you need is a gateway configured.
>>>> -MC
>>>>
>>>>
>>>> On Fri, Jul 29, 2011 at 1:40 PM, lloyd thomas <lloydie.t at gmail.com>wrote:
>>>>
>>>>> Sorry, example is not clear to me.
>>>>> I don't understand why a user config is relevant to sip registration
>>>>> for a provider.
>>>>> An example will help me more. Maybe CIDR attribute in a sip_profile
>>>>> gateway could help.
>>>>>
>>>>>
>>>>> On 29 July 2011 19:55, Steven Ayre <steveayre at gmail.com> wrote:
>>>>>
>>>>>> Look at the cidr attribute in the user directory to authenticate by
>>>>>> IP:
>>>>>> http://wiki.freeswitch.org/wiki/Acl#Users
>>>>>>
>>>>>> -Steve
>>>>>>
>>>>>> On 29 July 2011 19:38, lloyd thomas <lloydie.t at gmail.com> wrote:
>>>>>>
>>>>>>> *Hi I need a little help setting up a SIP registration for a
>>>>>>> provider that does not use auth.*
>>>>>>>
>>>>>>> *All I have is info below.*
>>>>>>> **
>>>>>>>
>>>>>>> *
>>>>>>> *
>>>>>>>
>>>>>>>
>>>>>>> SBC/Proxy IP:        80.40.150.150:5060
>>>>>>>
>>>>>>> Authentication:      Trusted IP ? 88.221.85.33
>>>>>>>
>>>>>>> Assigned DDI:        01869******, 01869******
>>>>>>>
>>>>>>> DTMF Method:         RFC2833
>>>>>>>
>>>>>>> Status:              Live
>>>>>>>
>>>>>>> No. of trunks:       2x
>>>>>>>
>>>>>>> Session Timer:       1800
>>>>>>>
>>>>>>> Profile*:*             Generic (35060)
>>>>>>>
>>>>>>>
>>>>>>> Apparently the following is used for *
>>>>>>>
>>>>>>> [vibe]
>>>>>>>
>>>>>>> type = friend
>>>>>>>
>>>>>>> host = 80.40.150.150
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110801/b3f7082c/attachment-0001.html 

From cmcureau at gmail.com  Mon Aug  1 19:46:46 2011
From: cmcureau at gmail.com (Chris Cureau)
Date: Mon, 1 Aug 2011 10:46:46 -0500
Subject: [Freeswitch-users] Help with choppy audio after attended
	transfer
In-Reply-To: <CAKbxfG-UOWoisXbbZaEH0Ruzm7=e2oobyvwqM1vJ-pQFtDuOow@mail.gmail.com>
References: <CAL0fNgWOuhAeMgLS3UsD4tSqdDd2FGSqXw4T-H9AnfVLBGOr5Q@mail.gmail.com>
	<CAKbxfG-UOWoisXbbZaEH0Ruzm7=e2oobyvwqM1vJ-pQFtDuOow@mail.gmail.com>
Message-ID: <CAL0fNgVu9EP4jFO9XzYw+SbZKiz8W7rN03A_6TWhOi-9cJf9bA@mail.gmail.com>

Anthony,

Thanks for answering...and sorry for the delay.  I've already checked all of
the ptime settings I can, and all phones plus freeswitch are set to use 20ms
packetization.  I've even set "scrooge" in the codec negotiation, but I keep
running into this issue.  I've updated my post with "sofia global siptrace
on".

I am assuming that the ptime issue happens around line 2462 (
http://pastebin.freeswitch.org/16935)


   1. 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:4711 Audio Codec
   Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:30:64000]
   2. 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:2753 Already using
   PCMU
   3. 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:4819 Set 2833 dtmf
   send payload to 101
   4. 2011-08-01 09:12:37.332892 [DEBUG] sofia.c:5599 Processing updated SDP
   5. 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:3042 Audio params are
   unchanged for sofia/internal/sip:1003 at 10.0.1.205:5060.
   6. 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:3052sofia/internal/sip:
   1003 at 10.0.1.205:5060 Setting audio receive payload in Re-INVITE to 0

Could this be an issue with the Aastra'a firmware?  Or maybe the MOH is
being processed at 30ms instead of 20ms, and the negotiation is not updated
somehow?

I don't mean to sound ignorant, but I'm really at a loss here...and thanks
again for any help!

Cheers,
Chris

On Fri, Jul 29, 2011 at 10:44 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> probably ptime related thing.
> you should have included the sip trace "sofia global siptrace on"
>
>
> On Fri, Jul 29, 2011 at 12:28 AM, Chris Cureau <cmcureau at gmail.com> wrote:
> > I'm having some issues with extremely choppy audio after a call has been
> > sent to another extension via an automated transfer.  The audio is great
> > when the call is answered.  Shortly after, the transfer button is pressed
> > and the incoming call hears music on hold.  The music on hold is sent to
> the
> > caller sounds fine as does the conversation between extensions.  When the
> > transfer is completed, the caller hears what sounds like someone speaking
> > through a fan (though slower) but incoming audio sounds fine.
> >
> > Since it's such a large log, I posted it to the FreeSWITCH pastebin:
> > http://pastebin.freeswitch.org/16911
> >
> > I'm thinking that it has something to do with the transition from MOH to
> the
> > internal extension, but I can't figure out what is happening.
> >
> > Any ideas?
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110801/9f036991/attachment.html 

From gavin.henry at gmail.com  Mon Aug  1 21:02:05 2011
From: gavin.henry at gmail.com (Gavin Henry)
Date: Mon, 1 Aug 2011 18:02:05 +0100
Subject: [Freeswitch-users] Still having problems making calls to phones
 behind double nat
In-Reply-To: <201107131141.59013.justlikeef@gmail.com>
References: <201107131141.59013.justlikeef@gmail.com>
Message-ID: <CAA8_NKDxuH8kpdzQ=L++ojsnfPYBeqHyxAywq+0pSCrYhcNUXA@mail.gmail.com>

On 13 July 2011 16:41, Rob Hutton <justlikeef at gmail.com> wrote:
> I am still unable to figure this one out and would appreciate any help.
>
> I have two phones at a remote location that can call out through the switch and have two way audio, but no one can call into them. ?The switch returns a 606 User Not registered even though show registrations seems to indicate that they are.
>
> I don't see anything in the sip trace to indicate that it actually tried to contact the destination user, but INFO keepalives are going back and forth...
>
> Configuration and sofia log are here: http://pastebin.freeswitch.org/16784
>

Hi,

What are the remote locations ext numbers to help grok your paste?

Thanks.

-- 
http://www.suretecsystems.com/services/openldap/
http://www.surevoip.co.uk


From lloydie.t at gmail.com  Mon Aug  1 23:35:48 2011
From: lloydie.t at gmail.com (lloyd thomas)
Date: Mon, 1 Aug 2011 20:35:48 +0100
Subject: [Freeswitch-users] Help setting up SIP reg
In-Reply-To: <CAKzWOxUyZoBS5Evp4b61vP8fm64Lef0n01ti3KmVraQO_BqYtA@mail.gmail.com>
References: <CAE=3qOYPpTuRTvq7RTJeUgzsCtvrfCjMiKFGu1fg0PHyZRE+-g@mail.gmail.com>
	<CAFiqYu=c8mY5BMWPumy1CgZHZ81kMRPgoNiF-6mjFAkW1q30fA@mail.gmail.com>
	<CAE=3qOZxwERT3CQEFHetO_o2s=aQUwYZ54TmMRPx9eYD=A7NoQ@mail.gmail.com>
	<CAKzWOxUEsGUKojF-vvstL6FvcDMhePqYGdXEQXQ4x-EJ81Bk6w@mail.gmail.com>
	<CAE=3qOacZh50iepxJKJD5UvtesxZAsN7RnPKNe1zdLO9VmFsuA@mail.gmail.com>
	<CAE=3qOZsNuiimFBZJTf4SbPRA8bRq_hJfPUunEnT8rxZj3qTnA@mail.gmail.com>
	<CAE=3qObVuLOhp2GD29Da-GfzB_qkbkyDP4kPmRxjdbJi7XeirQ@mail.gmail.com>
	<CAKzWOxUyZoBS5Evp4b61vP8fm64Lef0n01ti3KmVraQO_BqYtA@mail.gmail.com>
Message-ID: <CAE=3qObsDBDTsKY2ycK_NierbS_A+-f896yMW4OArTkgauYgRA@mail.gmail.com>

I think they have my IP on a white list.

On 1 August 2011 16:24, Michael Collins <msc at freeswitch.org> wrote:

> Do they challenge you (digest auth) or do they have your IP address on a
> white list? That's a critical piece of information that only your provider
> can supply.
>
> -MC
>
>
> On Fri, Jul 29, 2011 at 9:31 PM, lloyd thomas <lloydie.t at gmail.com> wrote:
>
>> OK Inbound working with:
>>
>>   <!--2circles -->
>>     <list name="domains" default="allow">
>>       <node type="allow" cidr="80.40.150.150/32"/>
>>     </list>
>>
>> Just need to sort outbound.
>>
>>
>> On 30 July 2011 04:59, lloyd thomas <lloydie.t at gmail.com> wrote:
>>
>>> Hi, dialling in produces the following error.
>>>
>>> 2011-07-30 04:56:07.818936 [DEBUG] sofia.c:6517 IP 80.40.150.150 Rejected
>>> by acl "domains". Falling back to Digest auth.
>>> 2011-07-30 04:56:07.826367 [WARNING] sofia_reg.c:1246 SIP auth challenge
>>> (INVITE) on sofia profile 'internal' for [01869******@172.16.XXX.XXX] from
>>> ip 80.40.150.150
>>>
>>>
>>>
>>> On 30 July 2011 04:34, lloyd thomas <lloydie.t at gmail.com> wrote:
>>>
>>>> I am registering with a them. I could not find suitable example in
>>>> http://wiki.freeswitch.org/wiki/SIP_Provider_Examples which
>>>>
>>>>
>>>> On 29 July 2011 21:57, Michael Collins <msc at freeswitch.org> wrote:
>>>>
>>>>> Are you registering with the provider or are they registering with you?
>>>>> If they register with you then a user example is appropriate. If you are
>>>>> registering with them then all you need is a gateway configured.
>>>>> -MC
>>>>>
>>>>>
>>>>> On Fri, Jul 29, 2011 at 1:40 PM, lloyd thomas <lloydie.t at gmail.com>wrote:
>>>>>
>>>>>> Sorry, example is not clear to me.
>>>>>> I don't understand why a user config is relevant to sip registration
>>>>>> for a provider.
>>>>>> An example will help me more. Maybe CIDR attribute in a sip_profile
>>>>>> gateway could help.
>>>>>>
>>>>>>
>>>>>> On 29 July 2011 19:55, Steven Ayre <steveayre at gmail.com> wrote:
>>>>>>
>>>>>>> Look at the cidr attribute in the user directory to authenticate by
>>>>>>> IP:
>>>>>>> http://wiki.freeswitch.org/wiki/Acl#Users
>>>>>>>
>>>>>>> -Steve
>>>>>>>
>>>>>>> On 29 July 2011 19:38, lloyd thomas <lloydie.t at gmail.com> wrote:
>>>>>>>
>>>>>>>> *Hi I need a little help setting up a SIP registration for a
>>>>>>>> provider that does not use auth.*
>>>>>>>>
>>>>>>>> *All I have is info below.*
>>>>>>>> **
>>>>>>>>
>>>>>>>> *
>>>>>>>> *
>>>>>>>>
>>>>>>>>
>>>>>>>> SBC/Proxy IP:        80.40.150.150:5060
>>>>>>>>
>>>>>>>> Authentication:      Trusted IP ? 88.221.85.33
>>>>>>>>
>>>>>>>> Assigned DDI:        01869******, 01869******
>>>>>>>>
>>>>>>>> DTMF Method:         RFC2833
>>>>>>>>
>>>>>>>> Status:              Live
>>>>>>>>
>>>>>>>> No. of trunks:       2x
>>>>>>>>
>>>>>>>> Session Timer:       1800
>>>>>>>>
>>>>>>>> Profile*:*             Generic (35060)
>>>>>>>>
>>>>>>>>
>>>>>>>> Apparently the following is used for *
>>>>>>>>
>>>>>>>> [vibe]
>>>>>>>>
>>>>>>>> type = friend
>>>>>>>>
>>>>>>>> host = 80.40.150.150
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110801/f78d74ed/attachment-0001.html 

From msc at freeswitch.org  Mon Aug  1 23:50:11 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 1 Aug 2011 12:50:11 -0700
Subject: [Freeswitch-users] Help setting up SIP reg
In-Reply-To: <CAE=3qObsDBDTsKY2ycK_NierbS_A+-f896yMW4OArTkgauYgRA@mail.gmail.com>
References: <CAE=3qOYPpTuRTvq7RTJeUgzsCtvrfCjMiKFGu1fg0PHyZRE+-g@mail.gmail.com>
	<CAFiqYu=c8mY5BMWPumy1CgZHZ81kMRPgoNiF-6mjFAkW1q30fA@mail.gmail.com>
	<CAE=3qOZxwERT3CQEFHetO_o2s=aQUwYZ54TmMRPx9eYD=A7NoQ@mail.gmail.com>
	<CAKzWOxUEsGUKojF-vvstL6FvcDMhePqYGdXEQXQ4x-EJ81Bk6w@mail.gmail.com>
	<CAE=3qOacZh50iepxJKJD5UvtesxZAsN7RnPKNe1zdLO9VmFsuA@mail.gmail.com>
	<CAE=3qOZsNuiimFBZJTf4SbPRA8bRq_hJfPUunEnT8rxZj3qTnA@mail.gmail.com>
	<CAE=3qObVuLOhp2GD29Da-GfzB_qkbkyDP4kPmRxjdbJi7XeirQ@mail.gmail.com>
	<CAKzWOxUyZoBS5Evp4b61vP8fm64Lef0n01ti3KmVraQO_BqYtA@mail.gmail.com>
	<CAE=3qObsDBDTsKY2ycK_NierbS_A+-f896yMW4OArTkgauYgRA@mail.gmail.com>
Message-ID: <CAKzWOxXYi-yO9DNDw1rHCduKq=h9q=BddFmn2dKuqALKb=t1fA@mail.gmail.com>

Okay, so what happens when you dial out? Sorry, it's been a few days and I
don't recall where we left off. Be sure to include console log w/ siptrace
on pastebin.freeswitch.org.

-MC

On Mon, Aug 1, 2011 at 12:35 PM, lloyd thomas <lloydie.t at gmail.com> wrote:

> I think they have my IP on a white list.
>
>
> On 1 August 2011 16:24, Michael Collins <msc at freeswitch.org> wrote:
>
>> Do they challenge you (digest auth) or do they have your IP address on a
>> white list? That's a critical piece of information that only your provider
>> can supply.
>>
>> -MC
>>
>>
>> On Fri, Jul 29, 2011 at 9:31 PM, lloyd thomas <lloydie.t at gmail.com>wrote:
>>
>>> OK Inbound working with:
>>>
>>>   <!--2circles -->
>>>     <list name="domains" default="allow">
>>>       <node type="allow" cidr="80.40.150.150/32"/>
>>>     </list>
>>>
>>> Just need to sort outbound.
>>>
>>>
>>> On 30 July 2011 04:59, lloyd thomas <lloydie.t at gmail.com> wrote:
>>>
>>>> Hi, dialling in produces the following error.
>>>>
>>>> 2011-07-30 04:56:07.818936 [DEBUG] sofia.c:6517 IP 80.40.150.150
>>>> Rejected by acl "domains". Falling back to Digest auth.
>>>> 2011-07-30 04:56:07.826367 [WARNING] sofia_reg.c:1246 SIP auth challenge
>>>> (INVITE) on sofia profile 'internal' for [01869******@172.16.XXX.XXX] from
>>>> ip 80.40.150.150
>>>>
>>>>
>>>>
>>>> On 30 July 2011 04:34, lloyd thomas <lloydie.t at gmail.com> wrote:
>>>>
>>>>> I am registering with a them. I could not find suitable example in
>>>>> http://wiki.freeswitch.org/wiki/SIP_Provider_Examples which
>>>>>
>>>>>
>>>>> On 29 July 2011 21:57, Michael Collins <msc at freeswitch.org> wrote:
>>>>>
>>>>>> Are you registering with the provider or are they registering with
>>>>>> you? If they register with you then a user example is appropriate. If you
>>>>>> are registering with them then all you need is a gateway configured.
>>>>>> -MC
>>>>>>
>>>>>>
>>>>>> On Fri, Jul 29, 2011 at 1:40 PM, lloyd thomas <lloydie.t at gmail.com>wrote:
>>>>>>
>>>>>>> Sorry, example is not clear to me.
>>>>>>> I don't understand why a user config is relevant to sip registration
>>>>>>> for a provider.
>>>>>>> An example will help me more. Maybe CIDR attribute in a sip_profile
>>>>>>> gateway could help.
>>>>>>>
>>>>>>>
>>>>>>> On 29 July 2011 19:55, Steven Ayre <steveayre at gmail.com> wrote:
>>>>>>>
>>>>>>>> Look at the cidr attribute in the user directory to authenticate by
>>>>>>>> IP:
>>>>>>>> http://wiki.freeswitch.org/wiki/Acl#Users
>>>>>>>>
>>>>>>>> -Steve
>>>>>>>>
>>>>>>>> On 29 July 2011 19:38, lloyd thomas <lloydie.t at gmail.com> wrote:
>>>>>>>>
>>>>>>>>> *Hi I need a little help setting up a SIP registration for a
>>>>>>>>> provider that does not use auth.*
>>>>>>>>>
>>>>>>>>> *All I have is info below.*
>>>>>>>>> **
>>>>>>>>>
>>>>>>>>> *
>>>>>>>>> *
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> SBC/Proxy IP:        80.40.150.150:5060
>>>>>>>>>
>>>>>>>>> Authentication:      Trusted IP ? 88.221.85.33
>>>>>>>>>
>>>>>>>>> Assigned DDI:        01869******, 01869******
>>>>>>>>>
>>>>>>>>> DTMF Method:         RFC2833
>>>>>>>>>
>>>>>>>>> Status:              Live
>>>>>>>>>
>>>>>>>>> No. of trunks:       2x
>>>>>>>>>
>>>>>>>>> Session Timer:       1800
>>>>>>>>>
>>>>>>>>> Profile*:*             Generic (35060)
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> Apparently the following is used for *
>>>>>>>>>
>>>>>>>>> [vibe]
>>>>>>>>>
>>>>>>>>> type = friend
>>>>>>>>>
>>>>>>>>> host = 80.40.150.150
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110801/ef706f6f/attachment.html 

From marketing at cluecon.com  Tue Aug  2 01:37:19 2011
From: marketing at cluecon.com (marketing at cluecon.com)
Date: Mon, 1 Aug 2011 21:37:19 +0000
Subject: [Freeswitch-users] Last Chance for ClueCon 2011: Get Your Hotel
	Today
Message-ID: <00000131874790bd-d76aba6c-cf10-4f72-a519-b6642d88aba0-000000@email.amazonses.com>

An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110801/08f8c1f9/attachment.html 

From gcd at i.ph  Tue Aug  2 02:10:32 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Tue, 2 Aug 2011 06:10:32 +0800
Subject: [Freeswitch-users] How to for video call ?
In-Reply-To: <4E3655DF.4060207@indoakses-online.com>
References: <4E3655DF.4060207@indoakses-online.com>
Message-ID: <CAHKqMb60DcNm8P8FztXyJOhmf0h6wQEMgZw_ami_vbWho2CP8Q@mail.gmail.com>

hi bino,

just change your clients' extension numbers to 1xxx e.g. 2001 to 1001 and
2003 to 1003. no need to use video_record and video_playback.  the stack
dialplan, using bridge app,  will take care of connecting video. the codecs
settings in vars.xml will negotiate h.263.

-nandy


On Mon, Aug 1, 2011 at 3:29 PM, bino oetomo <bino at indoakses-online.com>wrote:

> Dear All ..
> I just learn to setup a FreeSwitch server.
> Well .. actualy I set it up using FusionPBX
>
> Now .. I have to extention .. that is 2001 and 2003
>
> 2003 is my friends Mac station , using X-Lite software
> 2001 is my windows station using linphone.
>
> Both station is set to use H263 as Video Codec.
>
> I try to trace the video fail in my Freeswitch server, using :
>
> root at lapp freeswitch/conf# tail -f ../log/freeswitch.log |grep FAIL
> |grep video
>
> And here is the result
>
> Dialplan: sofia/internal/2003 at fusionpbx.int Regex (FAIL) [video_record]
> destination_number(2001) =~ /^\*9193$/ break=on-false
> Dialplan: sofia/internal/2003 at fusionpbx.int Regex (FAIL)
> [video_playback] destination_number(2001) =~ /^\*9194$/ break=on-false
> =============================================
> Dialplan: sofia/internal/2001 at fusinpbx.int Regex (FAIL) [video_record]
> destination_number(2003) =~ /^\*9193$/ break=on-false
> Dialplan: sofia/internal/2001 at fusinpbx.int Regex (FAIL) [video_playback]
> destination_number(2003) =~ /^\*9194$/ break=on-false
>
> The first 2 part is happen when 2003 dial 2001, and the other 2 part is
> vice versa.
>
>
> Here is a snippet of my vars.xml :
> root at lapp freeswitch/conf# more vars.xml
>
> <!-- Codecs -->
> <X-PRE-PROCESS cmd="set"
>
> data="global_codec_prefs=G722,PCMU,PCMA,GSM,H263,H264,H261,H263-1998,H263-2000"/>
> <X-PRE-PROCESS cmd="set"
>
> data="outbound_codec_prefs=G722,PCMU,PCMA,GSM,H263,H264,H261,H263-1998,H263-2000"/>
>
>
> Kindly please tell me what to check / edit.
>
> Sincerely
> -bino-
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/750dd993/attachment-0001.html 

From anthony.minessale at gmail.com  Tue Aug  2 02:12:19 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 1 Aug 2011 17:12:19 -0500
Subject: [Freeswitch-users] Sip Headers advice (Not parsing properly)
In-Reply-To: <1312208221935-6641160.post@n2.nabble.com>
References: <1311249742856-6606461.post@n2.nabble.com>
	<1312208221935-6641160.post@n2.nabble.com>
Message-ID: <CAKbxfG-Aw_28U0Q7SDbFbeeDoBJARY2WjHiE-P+z4-PsCtgpZw@mail.gmail.com>

maybe you should tell Cisco that they are supposed to start
nonstandard headers with X-


On Mon, Aug 1, 2011 at 9:17 AM, Dissident <jcgpoza at gmail.com> wrote:
> Problem solved!!! It seems that this doc is misleading ->
> http://wiki.freeswitch.org/wiki/Channel_Variables#Channel_Variable_Manipulation
> not all sip headers are gonna be parsed only the ones which start with X- or
> P- the rest are not parsed because they are not supposed to exist, however
> in my case they do and the only solution I've found to parse my CISCO
> headers is modifying the mod_sofia, file sofia.c line 6515, I added the last
> OR condition. } else if (!strncasecmp(un->un_name, "X-", 2) ||
> !strncasecmp(un->un_name, "P-", 2) || !strncasecmp(un->un_name, "Cisco", 5))
> { and now I can retrieve this header from the dialplan just like this:
> Thanks to Swk.
> ________________________________
> View this message in context: Re: Sip Headers advice (Not parsing properly)
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From gcd at i.ph  Tue Aug  2 04:38:58 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Tue, 2 Aug 2011 08:38:58 +0800
Subject: [Freeswitch-users] Fidelio
In-Reply-To: <CABH8QX2y1mu1L=9-KZk2HH-Xh9MFOS98p4ctvvQrF7AW_RJV_A@mail.gmail.com>
References: <CAHKqMb5dnE1mgdFaV65R=tk-zxvfz30=hi+3AyxezENQXVsrYw@mail.gmail.com>
	<CAB2KLyfDvLjmqUU2coOJzG6ss6xzKvsGD5OOsS+bVwKaXx+P0g@mail.gmail.com>
	<CAFiqYukzYyKZ_Tj8w8PFrM4FFpXzZiguRYNfqP8H=ah0iZLkhA@mail.gmail.com>
	<CAB2KLycjzFtLc8jTBM9377CDi488K9nDmVD=Zz-13MZ7_w-cxA@mail.gmail.com>
	<CABH8QX2y1mu1L=9-KZk2HH-Xh9MFOS98p4ctvvQrF7AW_RJV_A@mail.gmail.com>
Message-ID: <CAHKqMb5uAypVLHgqkt+GFNC_oj5wDPnFbzKHxjwr0QhcEhYJPw@mail.gmail.com>

hi Jo?o,

since you had FIAS experience, would a commercial  mod_fias
license/certification be possible just like g.729?

re companies selling FIAS connectors, i can't find one that connects FS to
Fidelio. otherwise, they would sound off in this thread.

-nandy

2011/7/6 Jo?o Mesquita <jmesquita at freeswitch.org>

> Guys, be careful because I think this document as well as the protocol are
> confidential. I had to sign an NDA with Fidelio to get my hands on it and
> pay a fee for it as well. You might as well confirm it since you all seem to
> be in the US where this type of information might be easier to get.
>
> There are LOTS of companies selling their connectors to Fidelio...
>
> One other point is that you need to have the certification with them to be
> considered compatible, otherwise, no consultant will install the connector
> on the fidelio side.
>
> Regards,
> Jo?o Mesquita
>
>
>
>
> On Tue, Jul 5, 2011 at 1:52 PM, Luis F Urrea <lfurrea at gmail.com> wrote:
>
>> Awesome! great suggestions to get started,
>>
>> There is also a FIAS simulator floating around.
>>
>> That one may be a little harder to find? :)
>>
>> On Tue, Jul 5, 2011 at 10:44 AM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> I'm assuming it's this document:
>>>
>>> ftp://ftp.veracomp.com.pl/net/nomadix/Nomadix%20-%20PMS%20info/FIAS150.pdf
>>> Quite easy to google once I had the version number.
>>>
>>> You may find the nicest approach is to write a FOSS libfias, then write
>>> an endpoint module to tie FS and libfias together. Plenty of existing
>>> endpoint modules (mod_sofia mod_skinny mod_opal mod_h323 etc) can show
>>> you examples to get you started. Don't forget to read the FS API
>>> documentation too: http://docs.freeswitch.org/
>>>
>>> I'm assuming there are no license/patent restrictions to using FIAS?
>>>
>>> Good luck!
>>>
>>> -Steve
>>>
>>>
>>>
>>>
>>> On 5 July 2011 17:30, Luis F Urrea <lfurrea at gmail.com> wrote:
>>>
>>>> Hello Nandy,
>>>>
>>>> A couple of months ago I started some research on the subject and
>>>> concluded I had to write my own interface to FS, however I haven't had the
>>>> time to get the project off ground yet.
>>>>
>>>> I do have a copy of FIAS specification version 1.5 from 2001 which is
>>>> publicly available I am sure it's not the latest but it should cover the
>>>> basics.
>>>>
>>>> Please contact me off list if you have a hard time getting it online.
>>>>
>>>> Regards
>>>>
>>>> On Mon, Jul 4, 2011 at 4:07 PM, Nandy Dagondon <gcd at i.ph> wrote:
>>>>
>>>>> hi everybody,
>>>>>
>>>>> anyone working on interfacing FS with Fidelio Hotel PMS?  i can't find
>>>>> the FIAS protocol/specs online.  is this freely available?
>>>>>
>>>>> tks,
>>>>> nandy
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/faf0c764/attachment.html 

From jmesquita at freeswitch.org  Tue Aug  2 04:56:02 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 1 Aug 2011 21:56:02 -0300
Subject: [Freeswitch-users] Fidelio
In-Reply-To: <CAHKqMb5uAypVLHgqkt+GFNC_oj5wDPnFbzKHxjwr0QhcEhYJPw@mail.gmail.com>
References: <CAHKqMb5dnE1mgdFaV65R=tk-zxvfz30=hi+3AyxezENQXVsrYw@mail.gmail.com>
	<CAB2KLyfDvLjmqUU2coOJzG6ss6xzKvsGD5OOsS+bVwKaXx+P0g@mail.gmail.com>
	<CAFiqYukzYyKZ_Tj8w8PFrM4FFpXzZiguRYNfqP8H=ah0iZLkhA@mail.gmail.com>
	<CAB2KLycjzFtLc8jTBM9377CDi488K9nDmVD=Zz-13MZ7_w-cxA@mail.gmail.com>
	<CABH8QX2y1mu1L=9-KZk2HH-Xh9MFOS98p4ctvvQrF7AW_RJV_A@mail.gmail.com>
	<CAHKqMb5uAypVLHgqkt+GFNC_oj5wDPnFbzKHxjwr0QhcEhYJPw@mail.gmail.com>
Message-ID: <CABH8QX1HHvqL_YK0rbs-m1Ky1GtbZ6OQKyKMKkAne8XV5WBSig@mail.gmail.com>

I think there are none doing this at this moment. Here's what I know tho:

1. I haven't seen the contracts or legal papers involved in the
certification
2. I've never done a connector even tho I worked with companies that did in
the past
3. I know that the homologation/certification costs money and not little
money. As far as I know, there are differences between countries when it
comes to the fee to be paid. Here is Argentina it is something close to
USD$5.000,00.
4. For you to be officially certified by them, you need to have your system
installed on at least 3 hotels with Fidelio, otherwise you just lost money.

That should pretty much give you a hint of what you are facing when it comes
to the certification process. Does your project accomodate these imposed
barriers? If they do, I might be able to help.

Regards,
Jo?o Mesquita



On Mon, Aug 1, 2011 at 9:38 PM, Nandy Dagondon <gcd at i.ph> wrote:

> hi Jo?o,
>
> since you had FIAS experience, would a commercial  mod_fias
> license/certification be possible just like g.729?
>
> re companies selling FIAS connectors, i can't find one that connects FS to
> Fidelio. otherwise, they would sound off in this thread.
>
> -nandy
>
> 2011/7/6 Jo?o Mesquita <jmesquita at freeswitch.org>
>
>> Guys, be careful because I think this document as well as the protocol are
>> confidential. I had to sign an NDA with Fidelio to get my hands on it and
>> pay a fee for it as well. You might as well confirm it since you all seem to
>> be in the US where this type of information might be easier to get.
>>
>> There are LOTS of companies selling their connectors to Fidelio...
>>
>> One other point is that you need to have the certification with them to be
>> considered compatible, otherwise, no consultant will install the connector
>> on the fidelio side.
>>
>> Regards,
>> Jo?o Mesquita
>>
>>
>>
>>
>> On Tue, Jul 5, 2011 at 1:52 PM, Luis F Urrea <lfurrea at gmail.com> wrote:
>>
>>> Awesome! great suggestions to get started,
>>>
>>> There is also a FIAS simulator floating around.
>>>
>>> That one may be a little harder to find? :)
>>>
>>> On Tue, Jul 5, 2011 at 10:44 AM, Steven Ayre <steveayre at gmail.com>wrote:
>>>
>>>> I'm assuming it's this document:
>>>>
>>>> ftp://ftp.veracomp.com.pl/net/nomadix/Nomadix%20-%20PMS%20info/FIAS150.pdf
>>>> Quite easy to google once I had the version number.
>>>>
>>>> You may find the nicest approach is to write a FOSS libfias, then write
>>>> an endpoint module to tie FS and libfias together. Plenty of existing
>>>> endpoint modules (mod_sofia mod_skinny mod_opal mod_h323 etc) can show
>>>> you examples to get you started. Don't forget to read the FS API
>>>> documentation too: http://docs.freeswitch.org/
>>>>
>>>> I'm assuming there are no license/patent restrictions to using FIAS?
>>>>
>>>> Good luck!
>>>>
>>>> -Steve
>>>>
>>>>
>>>>
>>>>
>>>> On 5 July 2011 17:30, Luis F Urrea <lfurrea at gmail.com> wrote:
>>>>
>>>>> Hello Nandy,
>>>>>
>>>>> A couple of months ago I started some research on the subject and
>>>>> concluded I had to write my own interface to FS, however I haven't had the
>>>>> time to get the project off ground yet.
>>>>>
>>>>> I do have a copy of FIAS specification version 1.5 from 2001 which is
>>>>> publicly available I am sure it's not the latest but it should cover the
>>>>> basics.
>>>>>
>>>>> Please contact me off list if you have a hard time getting it online.
>>>>>
>>>>> Regards
>>>>>
>>>>> On Mon, Jul 4, 2011 at 4:07 PM, Nandy Dagondon <gcd at i.ph> wrote:
>>>>>
>>>>>> hi everybody,
>>>>>>
>>>>>> anyone working on interfacing FS with Fidelio Hotel PMS?  i can't find
>>>>>> the FIAS protocol/specs online.  is this freely available?
>>>>>>
>>>>>> tks,
>>>>>> nandy
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110801/5f8b5084/attachment-0001.html 

From woof at iwoof.org  Tue Aug  2 05:07:00 2011
From: woof at iwoof.org (Andy Spitzer)
Date: Mon, 1 Aug 2011 21:07:00 -0400
Subject: [Freeswitch-users] Sip Headers advice (Not parsing properly)
In-Reply-To: <CAKbxfG-Aw_28U0Q7SDbFbeeDoBJARY2WjHiE-P+z4-PsCtgpZw@mail.gmail.com>
References: <1311249742856-6606461.post@n2.nabble.com>
	<1312208221935-6641160.post@n2.nabble.com>
	<CAKbxfG-Aw_28U0Q7SDbFbeeDoBJARY2WjHiE-P+z4-PsCtgpZw@mail.gmail.com>
Message-ID: <77BDF4BA-21C8-49AF-9C7A-9A02614FE2AB@iwoof.org>

Woof!

On Aug 1, 2011, at 18:12 , Anthony Minessale wrote:

> maybe you should tell Cisco that they are supposed to start
> nonstandard headers with X-

That's always been the convention, but there has been some noise about deprecating the use of the X- prefex in general  (see http://tools.ietf.org/html//draft-saintandre-xdash-03).  

For SIP, I believe the convention was never codified into any standard, and it just came along for the ride with the other RFCS that it "inherited" from (such as HTTP).  So much as I like to rag on Cisco for non-standard SIP implementations, this is not one of them.

--Woof!



From anthony.minessale at gmail.com  Tue Aug  2 05:34:18 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 1 Aug 2011 20:34:18 -0500
Subject: [Freeswitch-users] Help with choppy audio after attended
	transfer
In-Reply-To: <CAL0fNgVu9EP4jFO9XzYw+SbZKiz8W7rN03A_6TWhOi-9cJf9bA@mail.gmail.com>
References: <CAL0fNgWOuhAeMgLS3UsD4tSqdDd2FGSqXw4T-H9AnfVLBGOr5Q@mail.gmail.com>
	<CAKbxfG-UOWoisXbbZaEH0Ruzm7=e2oobyvwqM1vJ-pQFtDuOow@mail.gmail.com>
	<CAL0fNgVu9EP4jFO9XzYw+SbZKiz8W7rN03A_6TWhOi-9cJf9bA@mail.gmail.com>
Message-ID: <CAKbxfG_XUjkLJTcMFP4sHbp+KYy6fM8cTOuzyhyMRHh-DbVp=A@mail.gmail.com>

can you do make current and try latest git?


On Mon, Aug 1, 2011 at 10:46 AM, Chris Cureau <cmcureau at gmail.com> wrote:
> Anthony,
>
> Thanks for answering...and sorry for the delay.? I've already checked all of
> the ptime settings I can, and all phones plus freeswitch are set to use 20ms
> packetization.? I've even set "scrooge" in the codec negotiation, but I keep
> running into this issue.? I've updated my post with "sofia global siptrace
> on".
>
> I am assuming that the ptime issue happens around line 2462
> (http://pastebin.freeswitch.org/16935)
>
> 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:4711 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[PCMU:0:8000:30:64000]
> 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:2753 Already using PCMU
> 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:4819 Set 2833 dtmf send
> payload to 101
> 2011-08-01 09:12:37.332892 [DEBUG] sofia.c:5599 Processing updated SDP
> 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:3042 Audio params are
> unchanged for sofia/internal/sip:1003 at 10.0.1.205:5060.
> 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:3052
> sofia/internal/sip:1003 at 10.0.1.205:5060 Setting audio receive payload in
> Re-INVITE to 0
>
> Could this be an issue with the Aastra'a firmware?? Or maybe the MOH is
> being processed at 30ms instead of 20ms, and the negotiation is not updated
> somehow?
>
> I don't mean to sound ignorant, but I'm really at a loss here...and thanks
> again for any help!
>
> Cheers,
> Chris
>
> On Fri, Jul 29, 2011 at 10:44 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> probably ptime related thing.
>> you should have included the sip trace "sofia global siptrace on"
>>
>>
>> On Fri, Jul 29, 2011 at 12:28 AM, Chris Cureau <cmcureau at gmail.com> wrote:
>> > I'm having some issues with extremely choppy audio after a call has been
>> > sent to another extension via an automated transfer.? The audio is great
>> > when the call is answered.? Shortly after, the transfer button is
>> > pressed
>> > and the incoming call hears music on hold.? The music on hold is sent to
>> > the
>> > caller sounds fine as does the conversation between extensions.? When
>> > the
>> > transfer is completed, the caller hears what sounds like someone
>> > speaking
>> > through a fan (though slower) but incoming audio sounds fine.
>> >
>> > Since it's such a large log, I posted it to the FreeSWITCH pastebin:
>> > http://pastebin.freeswitch.org/16911
>> >
>> > I'm thinking that it has something to do with the transition from MOH to
>> > the
>> > internal extension, but I can't figure out what is happening.
>> >
>> > Any ideas?
>> >
>> > _______________________________________________
>> > Join us at ClueCon 2011, Aug 9-11, Chicago
>> > http://www.cluecon.com 877-7-4ACLUE
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> 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
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From gcd at i.ph  Tue Aug  2 05:36:49 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Tue, 2 Aug 2011 09:36:49 +0800
Subject: [Freeswitch-users] Fidelio
In-Reply-To: <CABH8QX1HHvqL_YK0rbs-m1Ky1GtbZ6OQKyKMKkAne8XV5WBSig@mail.gmail.com>
References: <CAHKqMb5dnE1mgdFaV65R=tk-zxvfz30=hi+3AyxezENQXVsrYw@mail.gmail.com>
	<CAB2KLyfDvLjmqUU2coOJzG6ss6xzKvsGD5OOsS+bVwKaXx+P0g@mail.gmail.com>
	<CAFiqYukzYyKZ_Tj8w8PFrM4FFpXzZiguRYNfqP8H=ah0iZLkhA@mail.gmail.com>
	<CAB2KLycjzFtLc8jTBM9377CDi488K9nDmVD=Zz-13MZ7_w-cxA@mail.gmail.com>
	<CABH8QX2y1mu1L=9-KZk2HH-Xh9MFOS98p4ctvvQrF7AW_RJV_A@mail.gmail.com>
	<CAHKqMb5uAypVLHgqkt+GFNC_oj5wDPnFbzKHxjwr0QhcEhYJPw@mail.gmail.com>
	<CABH8QX1HHvqL_YK0rbs-m1Ky1GtbZ6OQKyKMKkAne8XV5WBSig@mail.gmail.com>
Message-ID: <CAHKqMb51RbLBo+pmKwhJ82=ZWaGcyM6bd6y881Vh=4cAjztt5A@mail.gmail.com>

yes, certification is big obstacle. too much for our intended project.
 perhaps FS can apply for certification as a platform not on a
per-company/model basis. i just hope Fidelio would be open to the idea.

another option would be to ask Asterisk-FIAS connectors like PBillX the
possibility to include FreeSwitch.

2011/8/2 Jo?o Mesquita <jmesquita at freeswitch.org>

> I think there are none doing this at this moment. Here's what I know tho:
>
> 1. I haven't seen the contracts or legal papers involved in the
> certification
> 2. I've never done a connector even tho I worked with companies that did in
> the past
> 3. I know that the homologation/certification costs money and not little
> money. As far as I know, there are differences between countries when it
> comes to the fee to be paid. Here is Argentina it is something close to
> USD$5.000,00.
> 4. For you to be officially certified by them, you need to have your system
> installed on at least 3 hotels with Fidelio, otherwise you just lost money.
>
> That should pretty much give you a hint of what you are facing when it
> comes to the certification process. Does your project accomodate these
> imposed barriers? If they do, I might be able to help.
>
> Regards,
> Jo?o Mesquita
>
>
>
>
> On Mon, Aug 1, 2011 at 9:38 PM, Nandy Dagondon <gcd at i.ph> wrote:
>
>> hi Jo?o,
>>
>> since you had FIAS experience, would a commercial  mod_fias
>> license/certification be possible just like g.729?
>>
>> re companies selling FIAS connectors, i can't find one that connects FS to
>> Fidelio. otherwise, they would sound off in this thread.
>>
>> -nandy
>>
>> 2011/7/6 Jo?o Mesquita <jmesquita at freeswitch.org>
>>
>>> Guys, be careful because I think this document as well as the protocol
>>> are confidential. I had to sign an NDA with Fidelio to get my hands on it
>>> and pay a fee for it as well. You might as well confirm it since you all
>>> seem to be in the US where this type of information might be easier to get.
>>>
>>> There are LOTS of companies selling their connectors to Fidelio...
>>>
>>> One other point is that you need to have the certification with them to
>>> be considered compatible, otherwise, no consultant will install the
>>> connector on the fidelio side.
>>>
>>> Regards,
>>> Jo?o Mesquita
>>>
>>>
>>>
>>>
>>> On Tue, Jul 5, 2011 at 1:52 PM, Luis F Urrea <lfurrea at gmail.com> wrote:
>>>
>>>> Awesome! great suggestions to get started,
>>>>
>>>> There is also a FIAS simulator floating around.
>>>>
>>>> That one may be a little harder to find? :)
>>>>
>>>> On Tue, Jul 5, 2011 at 10:44 AM, Steven Ayre <steveayre at gmail.com>wrote:
>>>>
>>>>> I'm assuming it's this document:
>>>>>
>>>>> ftp://ftp.veracomp.com.pl/net/nomadix/Nomadix%20-%20PMS%20info/FIAS150.pdf
>>>>> Quite easy to google once I had the version number.
>>>>>
>>>>> You may find the nicest approach is to write a FOSS libfias, then write
>>>>> an endpoint module to tie FS and libfias together. Plenty of existing
>>>>> endpoint modules (mod_sofia mod_skinny mod_opal mod_h323 etc) can show
>>>>> you examples to get you started. Don't forget to read the FS API
>>>>> documentation too: http://docs.freeswitch.org/
>>>>>
>>>>> I'm assuming there are no license/patent restrictions to using FIAS?
>>>>>
>>>>> Good luck!
>>>>>
>>>>> -Steve
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On 5 July 2011 17:30, Luis F Urrea <lfurrea at gmail.com> wrote:
>>>>>
>>>>>> Hello Nandy,
>>>>>>
>>>>>> A couple of months ago I started some research on the subject and
>>>>>> concluded I had to write my own interface to FS, however I haven't had the
>>>>>> time to get the project off ground yet.
>>>>>>
>>>>>> I do have a copy of FIAS specification version 1.5 from 2001 which is
>>>>>> publicly available I am sure it's not the latest but it should cover the
>>>>>> basics.
>>>>>>
>>>>>> Please contact me off list if you have a hard time getting it online.
>>>>>>
>>>>>> Regards
>>>>>>
>>>>>> On Mon, Jul 4, 2011 at 4:07 PM, Nandy Dagondon <gcd at i.ph> wrote:
>>>>>>
>>>>>>> hi everybody,
>>>>>>>
>>>>>>> anyone working on interfacing FS with Fidelio Hotel PMS?  i can't
>>>>>>> find the FIAS protocol/specs online.  is this freely available?
>>>>>>>
>>>>>>> tks,
>>>>>>> nandy
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/4c99e1ec/attachment-0001.html 

From jmesquita at freeswitch.org  Tue Aug  2 05:49:48 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 1 Aug 2011 22:49:48 -0300
Subject: [Freeswitch-users] Fidelio
In-Reply-To: <CAHKqMb51RbLBo+pmKwhJ82=ZWaGcyM6bd6y881Vh=4cAjztt5A@mail.gmail.com>
References: <CAHKqMb5dnE1mgdFaV65R=tk-zxvfz30=hi+3AyxezENQXVsrYw@mail.gmail.com>
	<CAB2KLyfDvLjmqUU2coOJzG6ss6xzKvsGD5OOsS+bVwKaXx+P0g@mail.gmail.com>
	<CAFiqYukzYyKZ_Tj8w8PFrM4FFpXzZiguRYNfqP8H=ah0iZLkhA@mail.gmail.com>
	<CAB2KLycjzFtLc8jTBM9377CDi488K9nDmVD=Zz-13MZ7_w-cxA@mail.gmail.com>
	<CABH8QX2y1mu1L=9-KZk2HH-Xh9MFOS98p4ctvvQrF7AW_RJV_A@mail.gmail.com>
	<CAHKqMb5uAypVLHgqkt+GFNC_oj5wDPnFbzKHxjwr0QhcEhYJPw@mail.gmail.com>
	<CABH8QX1HHvqL_YK0rbs-m1Ky1GtbZ6OQKyKMKkAne8XV5WBSig@mail.gmail.com>
	<CAHKqMb51RbLBo+pmKwhJ82=ZWaGcyM6bd6y881Vh=4cAjztt5A@mail.gmail.com>
Message-ID: <CABH8QX0cgqbPcCtS=2Z0ron_vqX=5-YMrnkGJgqsOYXx81+x9Q@mail.gmail.com>

Like I said, if you can raise enough money to be able to get the
certification going, I can talk to the ppl I know on Micros and see if we
can get FS certified (no guarantees), but it will cost at least 5k USD, so I
really don't know if there is enough interest.

Regards,
Jo?o Mesquita



On Mon, Aug 1, 2011 at 10:36 PM, Nandy Dagondon <gcd at i.ph> wrote:

> yes, certification is big obstacle. too much for our intended project.
>  perhaps FS can apply for certification as a platform not on a
> per-company/model basis. i just hope Fidelio would be open to the idea.
>
> another option would be to ask Asterisk-FIAS connectors like PBillX the
> possibility to include FreeSwitch.
>
> 2011/8/2 Jo?o Mesquita <jmesquita at freeswitch.org>
>
>> I think there are none doing this at this moment. Here's what I know tho:
>>
>> 1. I haven't seen the contracts or legal papers involved in the
>> certification
>> 2. I've never done a connector even tho I worked with companies that did
>> in the past
>> 3. I know that the homologation/certification costs money and not little
>> money. As far as I know, there are differences between countries when it
>> comes to the fee to be paid. Here is Argentina it is something close to
>> USD$5.000,00.
>> 4. For you to be officially certified by them, you need to have your
>> system installed on at least 3 hotels with Fidelio, otherwise you just lost
>> money.
>>
>> That should pretty much give you a hint of what you are facing when it
>> comes to the certification process. Does your project accomodate these
>> imposed barriers? If they do, I might be able to help.
>>
>> Regards,
>> Jo?o Mesquita
>>
>>
>>
>>
>> On Mon, Aug 1, 2011 at 9:38 PM, Nandy Dagondon <gcd at i.ph> wrote:
>>
>>> hi Jo?o,
>>>
>>> since you had FIAS experience, would a commercial  mod_fias
>>> license/certification be possible just like g.729?
>>>
>>> re companies selling FIAS connectors, i can't find one that connects FS
>>> to Fidelio. otherwise, they would sound off in this thread.
>>>
>>> -nandy
>>>
>>> 2011/7/6 Jo?o Mesquita <jmesquita at freeswitch.org>
>>>
>>>> Guys, be careful because I think this document as well as the protocol
>>>> are confidential. I had to sign an NDA with Fidelio to get my hands on it
>>>> and pay a fee for it as well. You might as well confirm it since you all
>>>> seem to be in the US where this type of information might be easier to get.
>>>>
>>>> There are LOTS of companies selling their connectors to Fidelio...
>>>>
>>>> One other point is that you need to have the certification with them to
>>>> be considered compatible, otherwise, no consultant will install the
>>>> connector on the fidelio side.
>>>>
>>>> Regards,
>>>> Jo?o Mesquita
>>>>
>>>>
>>>>
>>>>
>>>> On Tue, Jul 5, 2011 at 1:52 PM, Luis F Urrea <lfurrea at gmail.com> wrote:
>>>>
>>>>> Awesome! great suggestions to get started,
>>>>>
>>>>> There is also a FIAS simulator floating around.
>>>>>
>>>>> That one may be a little harder to find? :)
>>>>>
>>>>> On Tue, Jul 5, 2011 at 10:44 AM, Steven Ayre <steveayre at gmail.com>wrote:
>>>>>
>>>>>> I'm assuming it's this document:
>>>>>>
>>>>>> ftp://ftp.veracomp.com.pl/net/nomadix/Nomadix%20-%20PMS%20info/FIAS150.pdf
>>>>>> Quite easy to google once I had the version number.
>>>>>>
>>>>>> You may find the nicest approach is to write a FOSS libfias, then
>>>>>> write an endpoint module to tie FS and libfias together. Plenty of
>>>>>> existing endpoint modules (mod_sofia mod_skinny mod_opal mod_h323 etc) can show
>>>>>> you examples to get you started. Don't forget to read the FS API
>>>>>> documentation too: http://docs.freeswitch.org/
>>>>>>
>>>>>> I'm assuming there are no license/patent restrictions to using FIAS?
>>>>>>
>>>>>> Good luck!
>>>>>>
>>>>>> -Steve
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> On 5 July 2011 17:30, Luis F Urrea <lfurrea at gmail.com> wrote:
>>>>>>
>>>>>>> Hello Nandy,
>>>>>>>
>>>>>>> A couple of months ago I started some research on the subject and
>>>>>>> concluded I had to write my own interface to FS, however I haven't had the
>>>>>>> time to get the project off ground yet.
>>>>>>>
>>>>>>> I do have a copy of FIAS specification version 1.5 from 2001 which is
>>>>>>> publicly available I am sure it's not the latest but it should cover the
>>>>>>> basics.
>>>>>>>
>>>>>>> Please contact me off list if you have a hard time getting it online.
>>>>>>>
>>>>>>> Regards
>>>>>>>
>>>>>>> On Mon, Jul 4, 2011 at 4:07 PM, Nandy Dagondon <gcd at i.ph> wrote:
>>>>>>>
>>>>>>>> hi everybody,
>>>>>>>>
>>>>>>>> anyone working on interfacing FS with Fidelio Hotel PMS?  i can't
>>>>>>>> find the FIAS protocol/specs online.  is this freely available?
>>>>>>>>
>>>>>>>> tks,
>>>>>>>> nandy
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110801/38d142c5/attachment.html 

From krice at freeswitch.org  Tue Aug  2 05:55:12 2011
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 01 Aug 2011 20:55:12 -0500
Subject: [Freeswitch-users] Sip Headers advice (Not parsing properly)
In-Reply-To: <77BDF4BA-21C8-49AF-9C7A-9A02614FE2AB@iwoof.org>
Message-ID: <CA5CC330.1A029%krice@freeswitch.org>

X- is not covered in any RFCs for SIP, however P- is covered...

X- is as you said kinda along for the ride


On 8/1/11 8:07 PM, "Andy Spitzer" <woof at iwoof.org> wrote:

> Woof!
> 
> On Aug 1, 2011, at 18:12 , Anthony Minessale wrote:
> 
>> maybe you should tell Cisco that they are supposed to start
>> nonstandard headers with X-
> 
> That's always been the convention, but there has been some noise about
> deprecating the use of the X- prefex in general  (see
> http://tools.ietf.org/html//draft-saintandre-xdash-03).
> 
> For SIP, I believe the convention was never codified into any standard, and it
> just came along for the ride with the other RFCS that it "inherited" from
> (such as HTTP).  So much as I like to rag on Cisco for non-standard SIP
> implementations, this is not one of them.
> 
> --Woof!
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From bino at indoakses-online.com  Tue Aug  2 07:10:10 2011
From: bino at indoakses-online.com (bino oetomo)
Date: Tue, 02 Aug 2011 10:10:10 +0700
Subject: [Freeswitch-users] How to for video call ?
In-Reply-To: <CAHKqMb60DcNm8P8FztXyJOhmf0h6wQEMgZw_ami_vbWho2CP8Q@mail.gmail.com>
References: <4E3655DF.4060207@indoakses-online.com>
	<CAHKqMb60DcNm8P8FztXyJOhmf0h6wQEMgZw_ami_vbWho2CP8Q@mail.gmail.com>
Message-ID: <4E376A92.9060201@indoakses-online.com>

Dear Nandy and All ..
On 08/02/2011 05:10 AM, Nandy Dagondon wrote:
> hi bino,
>
> just change your clients' extension numbers to 1xxx e.g. 2001 to 1001 
> and 2003 to 1003. no need to use video_record and video_playback.  the 
> stack dialplan, using bridge app,  will take care of connecting video. 
> the codecs settings in vars.xml will negotiate h.263.
>
> -nandy
>
Thankyou for your fast response

Ok ...
I try to do your sugestion

I create another 10 extension using auto generated .... from 1100 - 
1109, and for easy passwords I manualy change each extension's password 
to 1234

Next I try to connect linphones to this server.
Note that 2 linphones are :
1. PC, using ubuntu @ 192.168.10.232 for extension : 1104, and
2. Laptop , using windows @ 192.168.3.204 for extension : 1105

Steps :
A. PC Registering : looks fine, with indication in logs file as
2011-08-02 02:42:52.800123 [WARNING] sofia_reg.c:1337 SIP auth challenge 
(REGISTER) on sofia profile 'internal' for [1104 at fusionpbx.int] from ip 
192.168.10.232

B. Laptop Registering : looks fine, with indication in logs file as
2011-08-02 02:43:37.598379 [WARNING] sofia_reg.c:1337 SIP auth challenge 
(REGISTER) on sofia profile 'internal' for [1105 at fusionpbx.int] from ip 
192.168.3.204

C. Call : Itried to make a call from PC (1104) to Laptop (1105) .. it's 
failed
The Linphone report is as "User is Busy"

Detailed cut of freeswitch.log is posted at
http://pastebin.com/raw.php?i=acGRSWg6

So .. again , Kindly please give me your enlightment

Sincerely
-bino-



From freeswitch at aastral.net  Tue Aug  2 08:38:40 2011
From: freeswitch at aastral.net (Bill W.)
Date: Tue, 02 Aug 2011 00:38:40 -0400
Subject: [Freeswitch-users] ESL: not bein able to determine when SIP
 gateway is down when originating a call
In-Reply-To: <BANLkTik9R9xqEEQic9miZbvN7T8MWCF-JA@mail.gmail.com>
References: <BANLkTik9R9xqEEQic9miZbvN7T8MWCF-JA@mail.gmail.com>
Message-ID: <1Qo6kg-0001Wa-LC@mail.aastral.net>

Hey Anton,

I'm running in to this same issue.  Did you ever find a result?

Thanks,
Bill

On 5/14/11 1:17 PM, Anton VG wrote:
> I'm trying to catch an error, in case I would dial wrong (non
> existent) gateway (intentionally!)
> 
> I'm running ESL outbound listener, subscribing to all events,
> if I do bgapi 'originate' to a live gateway - there are normal events
> flow, and I
> can track what is happening.
> But if I issue originate to a gateway, which is not configured or
> simply down - there
> are no any events fired.
> 
> I only have an error on FS console
> 
> 2011-05-14 20:58:13.072927 [ERR] mod_sofia.c:4044 Invalid Gateway
> following by
> 2011-05-14 20:58:13.072927 [ERR] switch_ivr_originate.c:2447 Cannot
> create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]
> 
> But HOW to catch the given in ESL?
> 
> sofia.c seems just does not have event code for that cases
> 
> if (profile_name && !profile_found) {
> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "No Such
> Profile '%s'\n", profile_name);
>> status = SWITCH_STATUS_FALSE;
> }
> 
> logically there should be a proper way to determine that gateway is bad in my
> ESL dialplan, by catching the proper event/reply/whatever,
> For the moment i did trick: esl.api('sofia status gateway
> GatewayWhichIsDown')
> 
> When in production, and there is more than a single route, there will
> be plenty of cases, when you dial a bad gateway, so there should be a
> way for ESL dialplan to determine that a gateway is not callable for a
> moment, the reason WHY and  to retry with another one.
> 
> The trick above is bad, since:
> 1. blocking api query, before evey single gateway call attempt.
> 2. Gateway maybe known in UP state, but the state is stale, in dial in
> fact will go to DOWN gateway. So, ESL dialplan will screw in that case
> 
> Any clue?
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From anthony.minessale at gmail.com  Tue Aug  2 08:59:02 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 1 Aug 2011 23:59:02 -0500
Subject: [Freeswitch-users] ESL: not bein able to determine when SIP
 gateway is down when originating a call
In-Reply-To: <1Qo6kg-0001Wa-LC@mail.aastral.net>
References: <BANLkTik9R9xqEEQic9miZbvN7T8MWCF-JA@mail.gmail.com>
	<1Qo6kg-0001Wa-LC@mail.aastral.net>
Message-ID: <CAKbxfG_j7fRYa3yCvQAcOk=tKDr1ynWOVZqoi2geumDw80qZ0Q@mail.gmail.com>

make a patch to add a new custom sofia event and fire it


On Mon, Aug 1, 2011 at 11:38 PM, Bill W. <freeswitch at aastral.net> wrote:
> Hey Anton,
>
> I'm running in to this same issue. ?Did you ever find a result?
>
> Thanks,
> Bill
>
> On 5/14/11 1:17 PM, Anton VG wrote:
>> I'm trying to catch an error, in case I would dial wrong (non
>> existent) gateway (intentionally!)
>>
>> I'm running ESL outbound listener, subscribing to all events,
>> if I do bgapi 'originate' to a live gateway - there are normal events
>> flow, and I
>> can track what is happening.
>> But if I issue originate to a gateway, which is not configured or
>> simply down - there
>> are no any events fired.
>>
>> I only have an error on FS console
>>
>> 2011-05-14 20:58:13.072927 [ERR] mod_sofia.c:4044 Invalid Gateway
>> following by
>> 2011-05-14 20:58:13.072927 [ERR] switch_ivr_originate.c:2447 Cannot
>> create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]
>>
>> But HOW to catch the given in ESL?
>>
>> sofia.c seems just does not have event code for that cases
>>
>> if (profile_name && !profile_found) {
>> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "No Such
>> Profile '%s'\n", profile_name);
>>> status = SWITCH_STATUS_FALSE;
>> }
>>
>> logically there should be a proper way to determine that gateway is bad in my
>> ESL dialplan, by catching the proper event/reply/whatever,
>> For the moment i did trick: esl.api('sofia status gateway
>> GatewayWhichIsDown')
>>
>> When in production, and there is more than a single route, there will
>> be plenty of cases, when you dial a bad gateway, so there should be a
>> way for ESL dialplan to determine that a gateway is not callable for a
>> moment, the reason WHY and ?to retry with another one.
>>
>> The trick above is bad, since:
>> 1. blocking api query, before evey single gateway call attempt.
>> 2. Gateway maybe known in UP state, but the state is stale, in dial in
>> fact will go to DOWN gateway. So, ESL dialplan will screw in that case
>>
>> Any clue?
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From gmaruzz at gmail.com  Tue Aug  2 09:45:57 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 2 Aug 2011 07:45:57 +0200
Subject: [Freeswitch-users] How to for video call ?
In-Reply-To: <4E376A92.9060201@indoakses-online.com>
References: <4E3655DF.4060207@indoakses-online.com>
	<CAHKqMb60DcNm8P8FztXyJOhmf0h6wQEMgZw_ami_vbWho2CP8Q@mail.gmail.com>
	<4E376A92.9060201@indoakses-online.com>
Message-ID: <CALXCt0ouywkiXe-cit_q4=JmVCk9nwMdOq7Z-EM5ZG-6kcHHmw@mail.gmail.com>

Use extensions 1001 to 1020 (those are configured in the default dialplan).

-giovanni


On 8/2/11, bino oetomo <bino at indoakses-online.com> wrote:
> Dear Nandy and All ..
> On 08/02/2011 05:10 AM, Nandy Dagondon wrote:
>> hi bino,
>>
>> just change your clients' extension numbers to 1xxx e.g. 2001 to 1001
>> and 2003 to 1003. no need to use video_record and video_playback.  the
>> stack dialplan, using bridge app,  will take care of connecting video.
>> the codecs settings in vars.xml will negotiate h.263.
>>
>> -nandy
>>
> Thankyou for your fast response
>
> Ok ...
> I try to do your sugestion
>
> I create another 10 extension using auto generated .... from 1100 -
> 1109, and for easy passwords I manualy change each extension's password
> to 1234
>
> Next I try to connect linphones to this server.
> Note that 2 linphones are :
> 1. PC, using ubuntu @ 192.168.10.232 for extension : 1104, and
> 2. Laptop , using windows @ 192.168.3.204 for extension : 1105
>
> Steps :
> A. PC Registering : looks fine, with indication in logs file as
> 2011-08-02 02:42:52.800123 [WARNING] sofia_reg.c:1337 SIP auth challenge
> (REGISTER) on sofia profile 'internal' for [1104 at fusionpbx.int] from ip
> 192.168.10.232
>
> B. Laptop Registering : looks fine, with indication in logs file as
> 2011-08-02 02:43:37.598379 [WARNING] sofia_reg.c:1337 SIP auth challenge
> (REGISTER) on sofia profile 'internal' for [1105 at fusionpbx.int] from ip
> 192.168.3.204
>
> C. Call : Itried to make a call from PC (1104) to Laptop (1105) .. it's
> failed
> The Linphone report is as "User is Busy"
>
> Detailed cut of freeswitch.log is posted at
> http://pastebin.com/raw.php?i=acGRSWg6
>
> So .. again , Kindly please give me your enlightment
>
> Sincerely
> -bino-
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From peter.olsson at visionutveckling.se  Tue Aug  2 10:02:42 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 2 Aug 2011 08:02:42 +0200
Subject: [Freeswitch-users] ESL: not bein able to determine when SIP
 gateway is down when originating a call
In-Reply-To: <CAKbxfG_j7fRYa3yCvQAcOk=tKDr1ynWOVZqoi2geumDw80qZ0Q@mail.gmail.com>
References: <BANLkTik9R9xqEEQic9miZbvN7T8MWCF-JA@mail.gmail.com>
	<1Qo6kg-0001Wa-LC@mail.aastral.net>,
	<CAKbxfG_j7fRYa3yCvQAcOk=tKDr1ynWOVZqoi2geumDw80qZ0Q@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59EBABB87C@cooper>

In this case there should at least be a BACKGROUND_JOB event with the result from originate (INVALID_NUMBER_FORMAT) - are you monitoring these events? If you want a specific event fired for "gateway down" you will have to create a patch for this.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Anthony Minessale [anthony.minessale at gmail.com]
Skickat: den 2 augusti 2011 06:59
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] ESL: not bein able to determine when SIP gateway is down when originating a call

make a patch to add a new custom sofia event and fire it


On Mon, Aug 1, 2011 at 11:38 PM, Bill W. <freeswitch at aastral.net> wrote:
> Hey Anton,
>
> I'm running in to this same issue.  Did you ever find a result?
>
> Thanks,
> Bill
>
> On 5/14/11 1:17 PM, Anton VG wrote:
>> I'm trying to catch an error, in case I would dial wrong (non
>> existent) gateway (intentionally!)
>>
>> I'm running ESL outbound listener, subscribing to all events,
>> if I do bgapi 'originate' to a live gateway - there are normal events
>> flow, and I
>> can track what is happening.
>> But if I issue originate to a gateway, which is not configured or
>> simply down - there
>> are no any events fired.
>>
>> I only have an error on FS console
>>
>> 2011-05-14 20:58:13.072927 [ERR] mod_sofia.c:4044 Invalid Gateway
>> following by
>> 2011-05-14 20:58:13.072927 [ERR] switch_ivr_originate.c:2447 Cannot
>> create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]
>>
>> But HOW to catch the given in ESL?
>>
>> sofia.c seems just does not have event code for that cases
>>
>> if (profile_name && !profile_found) {
>> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "No Such
>> Profile '%s'\n", profile_name);
>>> status = SWITCH_STATUS_FALSE;
>> }
>>
>> logically there should be a proper way to determine that gateway is bad in my
>> ESL dialplan, by catching the proper event/reply/whatever,
>> For the moment i did trick: esl.api('sofia status gateway
>> GatewayWhichIsDown')
>>
>> When in production, and there is more than a single route, there will
>> be plenty of cases, when you dial a bad gateway, so there should be a
>> way for ESL dialplan to determine that a gateway is not callable for a
>> moment, the reason WHY and  to retry with another one.
>>
>> The trick above is bad, since:
>> 1. blocking api query, before evey single gateway call attempt.
>> 2. Gateway maybe known in UP state, but the state is stale, in dial in
>> fact will go to DOWN gateway. So, ESL dialplan will screw in that case
>>
>> Any clue?
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



--
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900

_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

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

!DSPAM:4e3784ca32763825510562!



From anthony.minessale at gmail.com  Tue Aug  2 10:10:31 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 2 Aug 2011 01:10:31 -0500
Subject: [Freeswitch-users] Sip Headers advice (Not parsing properly)
In-Reply-To: <CA5CC330.1A029%krice@freeswitch.org>
References: <77BDF4BA-21C8-49AF-9C7A-9A02614FE2AB@iwoof.org>
	<CA5CC330.1A029%krice@freeswitch.org>
Message-ID: <CAKbxfG8jYW2QQp1UwhFvsiVzJzs-HgQMchyfBG6QKgOwoKkWmA@mail.gmail.com>

I think the standard for X- is inherited from it's daddy: HTTP and in
turn from it's daddy: SMTP



On Mon, Aug 1, 2011 at 8:55 PM, Ken Rice <krice at freeswitch.org> wrote:
> X- is not covered in any RFCs for SIP, however P- is covered...
>
> X- is as you said kinda along for the ride
>
>
> On 8/1/11 8:07 PM, "Andy Spitzer" <woof at iwoof.org> wrote:
>
>> Woof!
>>
>> On Aug 1, 2011, at 18:12 , Anthony Minessale wrote:
>>
>>> maybe you should tell Cisco that they are supposed to start
>>> nonstandard headers with X-
>>
>> That's always been the convention, but there has been some noise about
>> deprecating the use of the X- prefex in general ?(see
>> http://tools.ietf.org/html//draft-saintandre-xdash-03).
>>
>> For SIP, I believe the convention was never codified into any standard, and it
>> just came along for the ride with the other RFCS that it "inherited" from
>> (such as HTTP). ?So much as I like to rag on Cisco for non-standard SIP
>> implementations, this is not one of them.
>>
>> --Woof!
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From tculjaga at gmail.com  Tue Aug  2 10:45:08 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 2 Aug 2011 08:45:08 +0200
Subject: [Freeswitch-users] Mod_rad_auth issue for FS working with
 FreeRadius server
In-Reply-To: <CAOTmbMzvd5OHL8z=_8vu6Orp2_prGxaw1gmXYCM2=2+pgzs_WA@mail.gmail.com>
References: <CAOTmbMzvd5OHL8z=_8vu6Orp2_prGxaw1gmXYCM2=2+pgzs_WA@mail.gmail.com>
Message-ID: <CAJoKKdX01Qc8ibs25PUELi8GrzMDNOrHb_bQvYdq3ZqA4XLy4w@mail.gmail.com>

hi,

dictionary.all is just the name of a file containing all attributes i needed
at that time.

you can include other dictionaries by putting #INCLUDE <pathname> at the end
of the dictionary file you reference in rad_auth.conf.xml.
if the INCLUDE doesn't work, just append dictionary.cisco to your dictionary
file... and make your own file.


check inline comments down below...


T.


On Sun, Jul 31, 2011 at 10:46 AM, fieldpeak <fieldpeak at gmail.com> wrote:

> Hello Gurus,
>
> i met a issue when using
> mod_rad_auth(http://wiki.freeswitch.org/wiki/Mod_rad_auth) to works
> with freeradius server+mysql for AAA, the details is below, Could
> anyone give any hints, Thanks in advance.
>
> i setup a dial plan "unitest_rad-ANI-auth" as wiki above, however,
> when i dialed 601 to trigger the dial plan, the console show errors,
> it looks "h323-conf-id" is not in the directory, then i tried to add
> this attribute to the dictionary, however, it does not help, in the
> wiki, it mentioned the rad_auth.conf.xml contains <param
> name="dictionary"
> value="/usr/local/etc/radiusclient/dictionary.all"/>, however i did
> not find the file "dictionary.all" at that directory, so i use
> dictionary. BTW, the freeradius server + mysql works well.
>

i just appended the information needed into dictionary.all file... (vendor
and attribute definition).



>
> console errors:
>
> EXECUTE sofia/internal/1001 at 124.193.106.104 auth_function(in , in
> 38516060333, in 003282, out AUTH_RESULT)
> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:301 allocate initial
> structure.
> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:313 initialzed
> configuration.
> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set authserver
> := 127.0.0.1:1812:gateway.
> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set dictionary
> := /usr/local/etc/radiusclient/dictionary.
> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set seqfile :=
> /var/run/radius.seq.
> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set mapfile :=
> /usr/local/etc/radiusclient/port-id-map.
> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set default_realm :=
> .
> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set radius_timeout :=
> 3.
> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set radius_retries :=
> 2.
> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set radius_deadtime
> := 0.
> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set bindaddr := *.
> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:371 ... radius:
> User-Name: 38516060333
> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:380 ... radius:
> User-Password: 003282
> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:391 ... radius:
> Called-station-Id is empty, ignoring...
> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:413 Handle
> attribute: h323-conf-id
> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:428 Unknown attribute:
> key:h323-conf-id, not found in dictionary
> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:538 abort sending
> radius packet.
> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:546 An error occured
> during RADIUS Authentication(RC=-1)
> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:702 An error occured
> during radius authorization.
> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  AUTH_RESULT=)
> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  AUTH_RESULT=
> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  billing_model=)
> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  billing_model=
> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  credit_amount=)
> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  credit_amount=
> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  currency=)
> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  currency=
> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  preffered_lang=)
> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  preffered_lang=
>
> added below in the dictionary(/usr/local/etc/radiusclient/dictionary):
>
> ATTRIBUTE       h323-conf-id            1008    string
>

you need the vendor definition as well


>
>
> dial plan:
> <extension name="unitest_rad-ANI-auth">
>    <condition field="destination_number" expression="^601$">
>      <action application="log" data="INFO  Before Auth "/>
>
>      <action inline="true" application="set"
> data="CALLID=h323-conf-id=${uuid}"/>
>      <action inline="true" application="set"
> data="SERVICENUM=h323-prompt-id=${destination_number}"/>
>      <action inline="true" application="set"
> data="TRANSACTIONID=h323-ivr-out=transactionID:1234"/>
>  <!--      <action inline="true" application="set"
> data="CALLINGNUMBER=${caller_id_number}"/> -->
>      <action inline="true" application="set"
> data="CALLINGNUMBER=38516060333"/>
>      <action inline="true" application="set" data="USERNAME=38516060333"/>
>  <!--      <action inline="true" application="set" data="USERNAME=209354"/>
> -->
>      <action inline="true" application="set" data="PASSWD=003282"/>
>  <!--      <action inline="true" application="set"
> data="DIALED_NUMBER=16094191500"/>  -->
>
>      <action application="sleep" data="2000"/>
>      <action application="auth_function" data="in ${DIALED_NUMBER},
> in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>
>
>
>      <action application="log" data="INFO  AUTH_RESULT=${AUTH_RESULT}"/>
>      <action application="log" data="INFO
>  billing_model=${billing_model}"/>
>      <action application="log" data="INFO
>  credit_amount=${credit_amount}"/>
>      <action application="log" data="INFO  currency=${currency}"/>
>      <action application="log" data="INFO
>  preffered_lang=${preffered_lang}"/>
>      <action application="log" data="INFO  credit_time=${credit_time}"/>
>      <action application="log" data="INFO
> h323_ivr_duration=${h323_ivr_duration}"/>
>      <action application="log" data="INFO  return_code=${return_code}"/>
>      <!-- <action application="execute_extension" data="AUTH XML default"/>
> -->
>    </condition>
>  </extension>
>
>  radius_cdr.conf.xml:
>  <configuration name="radius_cdr.conf" description="RADIUS CDR
> Configuration">
>
>        <settings>
>
>                <!-- location of the radius dictionary files -->
>
>                <param name="dictionary"
> value="/usr/local/freeswitch/conf/radius/dictionary"/>
>
>
your dictionary file need to contain all the attributes you are trying to
use or to include other dictionaries (In this case dictionary.cisco) from
the dictionary file you are referencing here.


>                <!-- number of retries for each server -->
>
>                <param name="radius_retries" value="3"/>
>
>                <!-- number of seconds to wait between retries -->
>
>                <param name="radius_timeout" value="5"/>
>
>                <!-- accounting servers, up to 8 allowed -->
>
>                <!-- value is "host:port:secret", port is optional -->
>
>                <!-- use IP ADDRESSES, not hostnames -->
>
>                <param name="acct_server" value="127.0.0.1:1813
> :testing123"/>
>
>
>        </settings>
>
> </configuration>
>
>  the FS version:
>  FreeSWITCH Version 1.0.head (git-492bc6b 2011-07-23 12-53-04 -0400)
>
>  Regards,
>  Charles
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/aa6a5bc9/attachment-0001.html 

From fieldpeak at gmail.com  Tue Aug  2 10:53:38 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Tue, 2 Aug 2011 14:53:38 +0800
Subject: [Freeswitch-users] mod_radius_cdr does not work
Message-ID: <CAOTmbMyjQOiThv0OBsZn+9y4b+8nKbityt9ZyU8JjObmknAoug@mail.gmail.com>

Dear All,

i'm trying to use mod_radius_cdr, and FS successfully loaded it, but when i
make a call, it show error below on console, could anyone can help? Thanks
in advance.

2011-08-02 14:45:32.786095 [ERR] mod_radius_cdr.c:242 failed adding
Freeswitch-Src: 1002

*debug level log on console:*

freeswitch at freeswitch> 2011-08-02 14:45:32.546104 [DEBUG] sofia.c:7030 IP
222.128.70.10 Rejected by acl "domains". Falling back to Digest auth.
2011-08-02 14:45:32.546104 [WARNING] sofia_reg.c:1337 SIP auth challenge
(INVITE) on sofia profile 'internal' for [1001 at 124.193.106.104] from ip
222.128.70.10
2011-08-02 14:45:32.786095 [DEBUG] sofia.c:7030 IP 222.128.70.10 Rejected by
acl "domains". Falling back to Digest auth.
2011-08-02 14:45:32.786095 [NOTICE] switch_channel.c:897 New Channel
sofia/internal/1002 at 124.193.106.104 [d8030a1e-552b-4a32-b7f0-175653b65eb7]
2011-08-02 14:45:32.786095 [DEBUG] sofia.c:5084 Channel sofia/internal/
1002 at 124.193.106.104 entering state [received][100]
2011-08-02 14:45:32.786095 [DEBUG] sofia.c:5095 Remote SDP:
v=0
o=- 186895328 186895344 IN IP4 192.168.200.100
s=eyeBeam
c=IN IP4 192.168.200.100
t=0 0
m=audio 6200 RTP/AVP 18 100 6 0 8 3 98 97 5 102 101
a=rtpmap:100 speex/16000
a=rtpmap:98 ilbc/8000
a=rtpmap:97 speex/8000
a=rtpmap:102 l16/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=alt:1 1 : 21373DA4 8CC12811 192.168.200.100 6200

2011-08-02 14:45:32.786095 [DEBUG] sofia.c:5258 (sofia/internal/
1002 at 124.193.106.104) State Change CS_NEW -> CS_INIT
2011-08-02 14:45:32.786095 [DEBUG] switch_core_session.c:1154 Send signal
sofia/internal/1002 at 124.193.106.104 [BREAK]
2011-08-02 14:45:32.786095 [DEBUG] switch_core_state_machine.c:325
(sofia/internal/1002 at 124.193.106.104) Running State Change CS_INIT
2011-08-02 14:45:32.786095 [DEBUG] switch_core_state_machine.c:364
(sofia/internal/1002 at 124.193.106.104) State INIT
2011-08-02 14:45:32.786095 [DEBUG] mod_sofia.c:85 sofia/internal/
1002 at 124.193.106.104 SOFIA INIT
2011-08-02 14:45:32.786095 [DEBUG] mod_sofia.c:125 (sofia/internal/
1002 at 124.193.106.104) State Change CS_INIT -> CS_ROUTING
2011-08-02 14:45:32.786095 [DEBUG] switch_core_session.c:1154 Send signal
sofia/internal/1002 at 124.193.106.104 [BREAK]
2011-08-02 14:45:32.786095 [DEBUG] switch_core_state_machine.c:364
(sofia/internal/1002 at 124.193.106.104) State INIT going to sleep
2011-08-02 14:45:32.786095 [DEBUG] switch_core_state_machine.c:325
(sofia/internal/1002 at 124.193.106.104) Running State Change CS_ROUTING
2011-08-02 14:45:32.786095 [DEBUG] switch_channel.c:1821 (sofia/internal/
1002 at 124.193.106.104) Callstate Change DOWN -> RINGING
2011-08-02 14:45:32.786095 [DEBUG] switch_core_state_machine.c:373
(sofia/internal/1002 at 124.193.106.104) State ROUTING
2011-08-02 14:45:32.786095 [DEBUG] mod_sofia.c:148 sofia/internal/
1002 at 124.193.106.104 SOFIA ROUTING
2011-08-02 14:45:32.786095 [DEBUG] mod_radius_cdr.c:156 [mod_radius_cdr]
Entering my_on_routing
2011-08-02 14:45:32.786095 [ERR] mod_radius_cdr.c:242 failed adding
Freeswitch-Src: 1002
2011-08-02 14:45:32.786095 [DEBUG] switch_core_state_machine.c:373
(sofia/internal/1002 at 124.193.106.104) State ROUTING going to sleep
2011-08-02 14:45:36.185722 [DEBUG] switch_core_session.c:855 Send signal
sofia/internal/1002 at 124.193.106.104 [BREAK]
2011-08-02 14:45:36.185722 [DEBUG] switch_core_session.c:855 Send signal
sofia/internal/1002 at 124.193.106.104 [BREAK]
2011-08-02 14:45:36.185722 [DEBUG] switch_core_session.c:855 Send signal
sofia/internal/1002 at 124.193.106.104 [BREAK]
2011-08-02 14:45:36.185722 [DEBUG] sofia.c:5084 Channel sofia/internal/
1002 at 124.193.106.104 entering state [terminated][487]
2011-08-02 14:45:36.185722 [DEBUG] switch_channel.c:2739 (sofia/internal/
1002 at 124.193.106.104) Callstate Change RINGING -> HANGUP
2011-08-02 14:45:36.185722 [NOTICE] sofia.c:5806 Hangup sofia/internal/
1002 at 124.193.106.104 [CS_ROUTING] [ORIGINATOR_CANCEL]
2011-08-02 14:45:36.185722 [DEBUG] switch_channel.c:2755 Send signal
sofia/internal/1002 at 124.193.106.104 [KILL]
2011-08-02 14:45:36.185722 [DEBUG] switch_core_session.c:1154 Send signal
sofia/internal/1002 at 124.193.106.104 [BREAK]
2011-08-02 14:45:36.185722 [DEBUG] switch_core_state_machine.c:325
(sofia/internal/1002 at 124.193.106.104) Running State Change CS_HANGUP
2011-08-02 14:45:36.185722 [DEBUG] switch_core_state_machine.c:575
(sofia/internal/1002 at 124.193.106.104) State HANGUP
2011-08-02 14:45:36.185722 [DEBUG] mod_sofia.c:452 sofia/internal/
1002 at 124.193.106.104 Overriding SIP cause 487 with 487 from the other leg
2011-08-02 14:45:36.185722 [DEBUG] mod_sofia.c:458 Channel sofia/internal/
1002 at 124.193.106.104 hanging up, cause: ORIGINATOR_CANCEL
2011-08-02 14:45:36.185722 [DEBUG] switch_core_state_machine.c:46
sofia/internal/1002 at 124.193.106.104 Standard HANGUP, cause:
ORIGINATOR_CANCEL
2011-08-02 14:45:36.185722 [DEBUG] switch_core_state_machine.c:575
(sofia/internal/1002 at 124.193.106.104) State HANGUP going to sleep
2011-08-02 14:45:36.185722 [DEBUG] switch_core_state_machine.c:356
(sofia/internal/1002 at 124.193.106.104) State Change CS_HANGUP -> CS_REPORTING
2011-08-02 14:45:36.185722 [DEBUG] switch_core_session.c:1154 Send signal
sofia/internal/1002 at 124.193.106.104 [BREAK]
2011-08-02 14:45:36.185722 [DEBUG] switch_core_state_machine.c:325
(sofia/internal/1002 at 124.193.106.104) Running State Change CS_REPORTING
2011-08-02 14:45:36.185722 [DEBUG] switch_core_state_machine.c:635
(sofia/internal/1002 at 124.193.106.104) State REPORTING
2011-08-02 14:45:36.185722 [DEBUG] mod_radius_cdr.c:424 [mod_radius_cdr]
Entering my_on_reporting
2011-08-02 14:45:36.185722 [ERR] mod_radius_cdr.c:463 failed adding
Freeswitch-Hangupcause: 487
2011-08-02 14:45:36.185722 [DEBUG] switch_core_state_machine.c:635
(sofia/internal/1002 at 124.193.106.104) State REPORTING going to sleep
2011-08-02 14:45:36.185722 [DEBUG] switch_core_state_machine.c:350
(sofia/internal/1002 at 124.193.106.104) State Change CS_REPORTING ->
CS_DESTROY
2011-08-02 14:45:36.185722 [DEBUG] switch_core_session.c:1154 Send signal
sofia/internal/1002 at 124.193.106.104 [BREAK]
2011-08-02 14:45:36.185722 [DEBUG] switch_core_session.c:1326 Session 2
(sofia/internal/1002 at 124.193.106.104) Locked, Waiting on external entities
2011-08-02 14:45:36.185722 [NOTICE] switch_core_session.c:1344 Session 2
(sofia/internal/1002 at 124.193.106.104) Ended
2011-08-02 14:45:36.185722 [NOTICE] switch_core_session.c:1346 Close Channel
sofia/internal/1002 at 124.193.106.104 [CS_DESTROY]
2011-08-02 14:45:36.185722 [DEBUG] switch_core_state_machine.c:464
(sofia/internal/1002 at 124.193.106.104) Callstate Change HANGUP -> DOWN
2011-08-02 14:45:36.185722 [DEBUG] switch_core_state_machine.c:467
(sofia/internal/1002 at 124.193.106.104) Running State Change CS_DESTROY
2011-08-02 14:45:36.185722 [DEBUG] switch_core_state_machine.c:477
(sofia/internal/1002 at 124.193.106.104) State DESTROY
2011-08-02 14:45:36.185722 [DEBUG] mod_sofia.c:363 sofia/internal/
1002 at 124.193.106.104 SOFIA DESTROY
2011-08-02 14:45:36.185722 [DEBUG] switch_core_state_machine.c:60
sofia/internal/1002 at 124.193.106.104 Standard DESTROY
2011-08-02 14:45:36.185722 [DEBUG] switch_core_state_machine.c:477
(sofia/internal/1002 at 124.193.106.104) State DESTROY going to sleep

Regards,
Charles
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/30a31642/attachment.html 

From fieldpeak at gmail.com  Tue Aug  2 11:50:07 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Tue, 2 Aug 2011 15:50:07 +0800
Subject: [Freeswitch-users] Mod_rad_auth issue for FS working with
 FreeRadius server
In-Reply-To: <CAJoKKdX01Qc8ibs25PUELi8GrzMDNOrHb_bQvYdq3ZqA4XLy4w@mail.gmail.com>
References: <CAOTmbMzvd5OHL8z=_8vu6Orp2_prGxaw1gmXYCM2=2+pgzs_WA@mail.gmail.com>
	<CAJoKKdX01Qc8ibs25PUELi8GrzMDNOrHb_bQvYdq3ZqA4XLy4w@mail.gmail.com>
Message-ID: <CAOTmbMxL3pLOuw655fyT5VzAON_N0PrDN+-i-++=oZoYU6+Fmg@mail.gmail.com>

Hi Tihomir,

Finally the answer coming, i see the hope, thanks for your reply, :)

As your advise, i only use one attribute(h323-conf-id) in my dialplan, and
only one attribute(h323-conf-id) in rad_auth.conf.xml, and using the
attached dictionary (from ciso) which contains this attribute, however, it
still prompt 'unknown attribute', so i suspected if it was reading
/usr/local/etc/radiusclient/dictionary, so i copy the same dictionary to
/usr/local/freeswitch/radius/, it did not any help at all... very strange...

Log:
2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set default_realm := .
2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set radius_timeout :=
3.
2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set radius_retries :=
2.
2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set radius_deadtime :=
0.
2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set bindaddr := *.
2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:371 ... radius: User-Name:
38516060333
2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:380 ... radius:
User-Password: 003282
2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:396 ... radius:
Called-station-Id: 16094191500
2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:413 Handle attribute:
h323-conf-id
2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:428 Unknown attribute:
key:h323-conf-id, not found in dictionary
2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:538 abort sending radius
packet.
2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:546 An error occured during
RADIUS Authentication(RC=-1)
2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:702 An error occured during
radius authorization.
EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  AUTH_RESULT=)



  <extension name="unitest_rad-ANI-auth">
    <condition field="destination_number" expression="^601$">
      <!-- <action application="log" data="INFO  Before Auth "/> -->

      <action inline="true" application="set" data="CALLID=h323-conf-id
=${uuid}"/>

      <action inline="true" application="set" data="USERNAME=1001"/>
      <action inline="true" application="set" data="PASSWD=1111"/>

      <action application="sleep" data="2000"/>
      <action application="auth_function" data="in ${DIALED_NUMBER}, in
${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>

    </condition>
  </extension>


<configuration name="rad_auth.conf" description="radius authentification
module">
  <settings>

  </settings>

  <client>
    <param name="authserver" value="127.0.0.1:1812:gateway"/>
    <param name="dictionary"
value="/usr/local/etc/radiusclient/dictionary"/>
    <param name="seqfile" value="/var/run/radius.seq"/>
    <param name="mapfile" value="/usr/local/etc/radiusclient/port-id-map"/>
    <param name="default_realm" value=""/>
    <param name="radius_timeout" value="3"/>
    <param name="radius_retries" value="2"/>
    <param name="radius_deadtime" value="0"/>
    <param name="bindaddr" value="*"/>
  </client>

  <vsas>

    <param name="h323-conf-id" id="24" value="CALLID" pec="9" expr="1"
direction="in"/>

  </vsas>
 </configuration>


2011/8/2 Tihomir Culjaga <tculjaga at gmail.com>

> hi,
>
> dictionary.all is just the name of a file containing all attributes i
> needed at that time.
>
> you can include other dictionaries by putting #INCLUDE <pathname> at the
> end of the dictionary file you reference in rad_auth.conf.xml.
> if the INCLUDE doesn't work, just append dictionary.cisco to your
> dictionary file... and make your own file.
>
>
> check inline comments down below...
>
>
> T.
>
>
> On Sun, Jul 31, 2011 at 10:46 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>
>> Hello Gurus,
>>
>> i met a issue when using
>> mod_rad_auth(http://wiki.freeswitch.org/wiki/Mod_rad_auth) to works
>> with freeradius server+mysql for AAA, the details is below, Could
>> anyone give any hints, Thanks in advance.
>>
>> i setup a dial plan "unitest_rad-ANI-auth" as wiki above, however,
>> when i dialed 601 to trigger the dial plan, the console show errors,
>> it looks "h323-conf-id" is not in the directory, then i tried to add
>> this attribute to the dictionary, however, it does not help, in the
>> wiki, it mentioned the rad_auth.conf.xml contains <param
>> name="dictionary"
>> value="/usr/local/etc/radiusclient/dictionary.all"/>, however i did
>> not find the file "dictionary.all" at that directory, so i use
>> dictionary. BTW, the freeradius server + mysql works well.
>>
>
> i just appended the information needed into dictionary.all file... (vendor
> and attribute definition).
>
>
>
>>
>> console errors:
>>
>> EXECUTE sofia/internal/1001 at 124.193.106.104 auth_function(in , in
>> 38516060333, in 003282, out AUTH_RESULT)
>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:301 allocate initial
>> structure.
>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:313 initialzed
>> configuration.
>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set authserver
>> := 127.0.0.1:1812:gateway.
>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set dictionary
>> := /usr/local/etc/radiusclient/dictionary.
>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set seqfile :=
>> /var/run/radius.seq.
>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set mapfile :=
>> /usr/local/etc/radiusclient/port-id-map.
>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set default_realm :=
>> .
>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set radius_timeout
>> := 3.
>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set radius_retries
>> := 2.
>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set radius_deadtime
>> := 0.
>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set bindaddr := *.
>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:371 ... radius:
>> User-Name: 38516060333
>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:380 ... radius:
>> User-Password: 003282
>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:391 ... radius:
>> Called-station-Id is empty, ignoring...
>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:413 Handle
>> attribute: h323-conf-id
>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:428 Unknown attribute:
>> key:h323-conf-id, not found in dictionary
>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:538 abort sending
>> radius packet.
>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:546 An error occured
>> during RADIUS Authentication(RC=-1)
>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:702 An error occured
>> during radius authorization.
>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  AUTH_RESULT=)
>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  AUTH_RESULT=
>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  billing_model=)
>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  billing_model=
>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  credit_amount=)
>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  credit_amount=
>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  currency=)
>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  currency=
>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  preffered_lang=)
>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  preffered_lang=
>>
>> added below in the dictionary(/usr/local/etc/radiusclient/dictionary):
>>
>> ATTRIBUTE       h323-conf-id            1008    string
>>
>
> you need the vendor definition as well
>
>
>>
>>
>> dial plan:
>> <extension name="unitest_rad-ANI-auth">
>>    <condition field="destination_number" expression="^601$">
>>      <action application="log" data="INFO  Before Auth "/>
>>
>>      <action inline="true" application="set"
>> data="CALLID=h323-conf-id=${uuid}"/>
>>      <action inline="true" application="set"
>> data="SERVICENUM=h323-prompt-id=${destination_number}"/>
>>      <action inline="true" application="set"
>> data="TRANSACTIONID=h323-ivr-out=transactionID:1234"/>
>>  <!--      <action inline="true" application="set"
>> data="CALLINGNUMBER=${caller_id_number}"/> -->
>>      <action inline="true" application="set"
>> data="CALLINGNUMBER=38516060333"/>
>>      <action inline="true" application="set" data="USERNAME=38516060333"/>
>>  <!--      <action inline="true" application="set"
>> data="USERNAME=209354"/> -->
>>      <action inline="true" application="set" data="PASSWD=003282"/>
>>  <!--      <action inline="true" application="set"
>> data="DIALED_NUMBER=16094191500"/>  -->
>>
>>      <action application="sleep" data="2000"/>
>>      <action application="auth_function" data="in ${DIALED_NUMBER},
>> in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>
>>
>>
>>      <action application="log" data="INFO  AUTH_RESULT=${AUTH_RESULT}"/>
>>      <action application="log" data="INFO
>>  billing_model=${billing_model}"/>
>>      <action application="log" data="INFO
>>  credit_amount=${credit_amount}"/>
>>      <action application="log" data="INFO  currency=${currency}"/>
>>      <action application="log" data="INFO
>>  preffered_lang=${preffered_lang}"/>
>>      <action application="log" data="INFO  credit_time=${credit_time}"/>
>>      <action application="log" data="INFO
>> h323_ivr_duration=${h323_ivr_duration}"/>
>>      <action application="log" data="INFO  return_code=${return_code}"/>
>>      <!-- <action application="execute_extension" data="AUTH XML
>> default"/> -->
>>    </condition>
>>  </extension>
>>
>>  radius_cdr.conf.xml:
>>  <configuration name="radius_cdr.conf" description="RADIUS CDR
>> Configuration">
>>
>>        <settings>
>>
>>                <!-- location of the radius dictionary files -->
>>
>>                <param name="dictionary"
>> value="/usr/local/freeswitch/conf/radius/dictionary"/>
>>
>>
> your dictionary file need to contain all the attributes you are trying to
> use or to include other dictionaries (In this case dictionary.cisco) from
> the dictionary file you are referencing here.
>
>
>>                <!-- number of retries for each server -->
>>
>>                <param name="radius_retries" value="3"/>
>>
>>                <!-- number of seconds to wait between retries -->
>>
>>                <param name="radius_timeout" value="5"/>
>>
>>                <!-- accounting servers, up to 8 allowed -->
>>
>>                <!-- value is "host:port:secret", port is optional -->
>>
>>                <!-- use IP ADDRESSES, not hostnames -->
>>
>>                <param name="acct_server" value="127.0.0.1:1813
>> :testing123"/>
>>
>>
>>        </settings>
>>
>> </configuration>
>>
>>  the FS version:
>>  FreeSWITCH Version 1.0.head (git-492bc6b 2011-07-23 12-53-04 -0400)
>>
>>  Regards,
>>  Charles
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>>
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/cb809979/attachment-0001.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: dictionary
Type: application/octet-stream
Size: 5564 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/cb809979/attachment-0001.obj 

From tculjaga at gmail.com  Tue Aug  2 12:07:19 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 2 Aug 2011 10:07:19 +0200
Subject: [Freeswitch-users] Mod_rad_auth issue for FS working with
 FreeRadius server
In-Reply-To: <CAOTmbMxL3pLOuw655fyT5VzAON_N0PrDN+-i-++=oZoYU6+Fmg@mail.gmail.com>
References: <CAOTmbMzvd5OHL8z=_8vu6Orp2_prGxaw1gmXYCM2=2+pgzs_WA@mail.gmail.com>
	<CAJoKKdX01Qc8ibs25PUELi8GrzMDNOrHb_bQvYdq3ZqA4XLy4w@mail.gmail.com>
	<CAOTmbMxL3pLOuw655fyT5VzAON_N0PrDN+-i-++=oZoYU6+Fmg@mail.gmail.com>
Message-ID: <CAJoKKdWo=GEx=gdyBhKoH+J38hza=6Lv56QwDw-f6c+rpobzLg@mail.gmail.com>

<param name="dictionary" value="/usr/local/etc/radiusclient/dictionary"/>

this is your path to your dictionary ... you can put any path there.

[ ~]$ cd /usr/local/etc/radiusclient/
[radiusclient]$ ls -l
total 100
-rw-r--r-- 1 root root  7329 Aug 10  2010 dictionary
-rw-r--r-- 1 root root 21152 Apr 17  2010 dictionary.all
-rw-r--r-- 1 root root 11557 Mar 14  2010 dictionary.all.working
-rw-r--r-- 1 root root 12388 Aug 10  2010 dictionary.ascend
-rw-r--r-- 1 root root  5793 Mar 14  2010 dictionary.cisco
-rw-r--r-- 1 root root  1517 Aug 10  2010 dictionary.compat
-rw-r--r-- 1 root root   599 Aug 10  2010 dictionary.merit
-rw-r--r-- 1 root root  1117 Mar 11  2010 dictionary.rfc2869
-rw-r--r-- 1 root root  2489 Aug 10  2010 dictionary.sip
-rw-r--r-- 1 root root   135 Aug 10  2010 issue
-rw-r--r-- 1 root root   410 Aug 10  2010 port-id-map
-rw-r--r-- 1 root root  3299 Aug 10  2010 radiusclient.conf
-rw------- 1 root root   299 Aug 10  2010 servers



you cannot use just cisco dictionary. ... you either include cisco
dictionary in the radius client default dictionary or you append the values
you need.


use attached dictionary ... this one works fine.

T.

On Tue, Aug 2, 2011 at 9:50 AM, fieldpeak <fieldpeak at gmail.com> wrote:

> Hi Tihomir,
>
> Finally the answer coming, i see the hope, thanks for your reply, :)
>
> As your advise, i only use one attribute(h323-conf-id) in my dialplan, and
> only one attribute(h323-conf-id) in rad_auth.conf.xml, and using the
> attached dictionary (from ciso) which contains this attribute, however, it
> still prompt 'unknown attribute', so i suspected if it was reading
> /usr/local/etc/radiusclient/dictionary, so i copy the same dictionary to
> /usr/local/freeswitch/radius/, it did not any help at all... very strange...
>
> Log:
> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set default_realm :=
> .
> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set radius_timeout :=
> 3.
> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set radius_retries :=
> 2.
> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set radius_deadtime
> := 0.
> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set bindaddr := *.
> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:371 ... radius:
> User-Name: 38516060333
> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:380 ... radius:
> User-Password: 003282
> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:396 ... radius:
> Called-station-Id: 16094191500
> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:413 Handle attribute:
> h323-conf-id
> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:428 Unknown attribute:
> key:h323-conf-id, not found in dictionary
> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:538 abort sending radius
> packet.
> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:546 An error occured during
> RADIUS Authentication(RC=-1)
> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:702 An error occured during
> radius authorization.
>
> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  AUTH_RESULT=)
>
>
>
>   <extension name="unitest_rad-ANI-auth">
>     <condition field="destination_number" expression="^601$">
>       <!-- <action application="log" data="INFO  Before Auth "/> -->
>
>       <action inline="true" application="set" data="CALLID=h323-conf-id
> =${uuid}"/>
>
>       <action inline="true" application="set" data="USERNAME=1001"/>
>       <action inline="true" application="set" data="PASSWD=1111"/>
>
>
>       <action application="sleep" data="2000"/>
>       <action application="auth_function" data="in ${DIALED_NUMBER}, in
> ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>
>
>     </condition>
>   </extension>
>
>
>
> <configuration name="rad_auth.conf" description="radius authentification
> module">
>   <settings>
>
>   </settings>
>
>   <client>
>     <param name="authserver" value="127.0.0.1:1812:gateway"/>
>     <param name="dictionary"
> value="/usr/local/etc/radiusclient/dictionary"/>
>     <param name="seqfile" value="/var/run/radius.seq"/>
>     <param name="mapfile" value="/usr/local/etc/radiusclient/port-id-map"/>
>     <param name="default_realm" value=""/>
>     <param name="radius_timeout" value="3"/>
>     <param name="radius_retries" value="2"/>
>     <param name="radius_deadtime" value="0"/>
>     <param name="bindaddr" value="*"/>
>   </client>
>
>   <vsas>
>
>
>     <param name="h323-conf-id" id="24" value="CALLID" pec="9" expr="1"
> direction="in"/>
>
>   </vsas>
>  </configuration>
>
>
>
> 2011/8/2 Tihomir Culjaga <tculjaga at gmail.com>
>
>> hi,
>>
>> dictionary.all is just the name of a file containing all attributes i
>> needed at that time.
>>
>> you can include other dictionaries by putting #INCLUDE <pathname> at the
>> end of the dictionary file you reference in rad_auth.conf.xml.
>> if the INCLUDE doesn't work, just append dictionary.cisco to your
>> dictionary file... and make your own file.
>>
>>
>> check inline comments down below...
>>
>>
>> T.
>>
>>
>> On Sun, Jul 31, 2011 at 10:46 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>>
>>> Hello Gurus,
>>>
>>> i met a issue when using
>>> mod_rad_auth(http://wiki.freeswitch.org/wiki/Mod_rad_auth) to works
>>> with freeradius server+mysql for AAA, the details is below, Could
>>> anyone give any hints, Thanks in advance.
>>>
>>> i setup a dial plan "unitest_rad-ANI-auth" as wiki above, however,
>>> when i dialed 601 to trigger the dial plan, the console show errors,
>>> it looks "h323-conf-id" is not in the directory, then i tried to add
>>> this attribute to the dictionary, however, it does not help, in the
>>> wiki, it mentioned the rad_auth.conf.xml contains <param
>>> name="dictionary"
>>> value="/usr/local/etc/radiusclient/dictionary.all"/>, however i did
>>> not find the file "dictionary.all" at that directory, so i use
>>> dictionary. BTW, the freeradius server + mysql works well.
>>>
>>
>> i just appended the information needed into dictionary.all file... (vendor
>> and attribute definition).
>>
>>
>>
>>>
>>> console errors:
>>>
>>> EXECUTE sofia/internal/1001 at 124.193.106.104 auth_function(in , in
>>> 38516060333, in 003282, out AUTH_RESULT)
>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:301 allocate initial
>>> structure.
>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:313 initialzed
>>> configuration.
>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set authserver
>>> := 127.0.0.1:1812:gateway.
>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set dictionary
>>> := /usr/local/etc/radiusclient/dictionary.
>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set seqfile :=
>>> /var/run/radius.seq.
>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set mapfile :=
>>> /usr/local/etc/radiusclient/port-id-map.
>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set default_realm
>>> := .
>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set radius_timeout
>>> := 3.
>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set radius_retries
>>> := 2.
>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set radius_deadtime
>>> := 0.
>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set bindaddr := *.
>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:371 ... radius:
>>> User-Name: 38516060333
>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:380 ... radius:
>>> User-Password: 003282
>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:391 ... radius:
>>> Called-station-Id is empty, ignoring...
>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:413 Handle
>>> attribute: h323-conf-id
>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:428 Unknown attribute:
>>> key:h323-conf-id, not found in dictionary
>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:538 abort sending
>>> radius packet.
>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:546 An error occured
>>> during RADIUS Authentication(RC=-1)
>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:702 An error occured
>>> during radius authorization.
>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  AUTH_RESULT=)
>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  AUTH_RESULT=
>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  billing_model=)
>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  billing_model=
>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  credit_amount=)
>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  credit_amount=
>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  currency=)
>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  currency=
>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  preffered_lang=)
>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  preffered_lang=
>>>
>>> added below in the dictionary(/usr/local/etc/radiusclient/dictionary):
>>>
>>> ATTRIBUTE       h323-conf-id            1008    string
>>>
>>
>> you need the vendor definition as well
>>
>>
>>>
>>>
>>> dial plan:
>>> <extension name="unitest_rad-ANI-auth">
>>>    <condition field="destination_number" expression="^601$">
>>>      <action application="log" data="INFO  Before Auth "/>
>>>
>>>      <action inline="true" application="set"
>>> data="CALLID=h323-conf-id=${uuid}"/>
>>>      <action inline="true" application="set"
>>> data="SERVICENUM=h323-prompt-id=${destination_number}"/>
>>>      <action inline="true" application="set"
>>> data="TRANSACTIONID=h323-ivr-out=transactionID:1234"/>
>>>  <!--      <action inline="true" application="set"
>>> data="CALLINGNUMBER=${caller_id_number}"/> -->
>>>      <action inline="true" application="set"
>>> data="CALLINGNUMBER=38516060333"/>
>>>      <action inline="true" application="set"
>>> data="USERNAME=38516060333"/>
>>>  <!--      <action inline="true" application="set"
>>> data="USERNAME=209354"/> -->
>>>      <action inline="true" application="set" data="PASSWD=003282"/>
>>>  <!--      <action inline="true" application="set"
>>> data="DIALED_NUMBER=16094191500"/>  -->
>>>
>>>      <action application="sleep" data="2000"/>
>>>      <action application="auth_function" data="in ${DIALED_NUMBER},
>>> in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>
>>>
>>>
>>>      <action application="log" data="INFO  AUTH_RESULT=${AUTH_RESULT}"/>
>>>      <action application="log" data="INFO
>>>  billing_model=${billing_model}"/>
>>>      <action application="log" data="INFO
>>>  credit_amount=${credit_amount}"/>
>>>      <action application="log" data="INFO  currency=${currency}"/>
>>>      <action application="log" data="INFO
>>>  preffered_lang=${preffered_lang}"/>
>>>      <action application="log" data="INFO  credit_time=${credit_time}"/>
>>>      <action application="log" data="INFO
>>> h323_ivr_duration=${h323_ivr_duration}"/>
>>>      <action application="log" data="INFO  return_code=${return_code}"/>
>>>      <!-- <action application="execute_extension" data="AUTH XML
>>> default"/> -->
>>>    </condition>
>>>  </extension>
>>>
>>>  radius_cdr.conf.xml:
>>>  <configuration name="radius_cdr.conf" description="RADIUS CDR
>>> Configuration">
>>>
>>>        <settings>
>>>
>>>                <!-- location of the radius dictionary files -->
>>>
>>>                <param name="dictionary"
>>> value="/usr/local/freeswitch/conf/radius/dictionary"/>
>>>
>>>
>> your dictionary file need to contain all the attributes you are trying to
>> use or to include other dictionaries (In this case dictionary.cisco) from
>> the dictionary file you are referencing here.
>>
>>
>>>                <!-- number of retries for each server -->
>>>
>>>                <param name="radius_retries" value="3"/>
>>>
>>>                <!-- number of seconds to wait between retries -->
>>>
>>>                <param name="radius_timeout" value="5"/>
>>>
>>>                <!-- accounting servers, up to 8 allowed -->
>>>
>>>                <!-- value is "host:port:secret", port is optional -->
>>>
>>>                <!-- use IP ADDRESSES, not hostnames -->
>>>
>>>                <param name="acct_server" value="127.0.0.1:1813
>>> :testing123"/>
>>>
>>>
>>>        </settings>
>>>
>>> </configuration>
>>>
>>>  the FS version:
>>>  FreeSWITCH Version 1.0.head (git-492bc6b 2011-07-23 12-53-04 -0400)
>>>
>>>  Regards,
>>>  Charles
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>>
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/c2f4642d/attachment-0001.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: dictionary.all
Type: application/octet-stream
Size: 21151 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/c2f4642d/attachment-0001.obj 

From fieldpeak at gmail.com  Tue Aug  2 12:08:51 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Tue, 2 Aug 2011 16:08:51 +0800
Subject: [Freeswitch-users] Mod_rad_auth issue for FS working with
 FreeRadius server
In-Reply-To: <CAOTmbMxL3pLOuw655fyT5VzAON_N0PrDN+-i-++=oZoYU6+Fmg@mail.gmail.com>
References: <CAOTmbMzvd5OHL8z=_8vu6Orp2_prGxaw1gmXYCM2=2+pgzs_WA@mail.gmail.com>
	<CAJoKKdX01Qc8ibs25PUELi8GrzMDNOrHb_bQvYdq3ZqA4XLy4w@mail.gmail.com>
	<CAOTmbMxL3pLOuw655fyT5VzAON_N0PrDN+-i-++=oZoYU6+Fmg@mail.gmail.com>
Message-ID: <CAOTmbMyvyDA-CRpaMC=WRGVkiyTG=Rs+n4hyeQWdyTZpqysSzw@mail.gmail.com>

i tried change to 'h323-conf-id' to 'h323-call-origin' in
02_unitest_rad-ANI-auth.xml, rad_auth.conf.xml, however, it still prompt
'[ERR] mod_rad_auth.c:428 Unknown attribute: key:h323-conf-id, not found in
dictionary', so where the mod_rad_auth read out the 'h323-conf-id'? very
very strange, which dictionary it was using...

Regards,
Charles

2011/8/2 fieldpeak <fieldpeak at gmail.com>

> Hi Tihomir,
>
> Finally the answer coming, i see the hope, thanks for your reply, :)
>
> As your advise, i only use one attribute(h323-conf-id) in my dialplan, and
> only one attribute(h323-conf-id) in rad_auth.conf.xml, and using the
> attached dictionary (from ciso) which contains this attribute, however, it
> still prompt 'unknown attribute', so i suspected if it was reading
> /usr/local/etc/radiusclient/dictionary, so i copy the same dictionary to
> /usr/local/freeswitch/radius/, it did not any help at all... very strange...
>
> Log:
> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set default_realm :=
> .
> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set radius_timeout :=
> 3.
> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set radius_retries :=
> 2.
> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set radius_deadtime
> := 0.
> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set bindaddr := *.
> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:371 ... radius:
> User-Name: 38516060333
> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:380 ... radius:
> User-Password: 003282
> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:396 ... radius:
> Called-station-Id: 16094191500
> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:413 Handle attribute:
> h323-conf-id
> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:428 Unknown attribute:
> key:h323-conf-id, not found in dictionary
> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:538 abort sending radius
> packet.
> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:546 An error occured during
> RADIUS Authentication(RC=-1)
> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:702 An error occured during
> radius authorization.
>
> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  AUTH_RESULT=)
>
>
>
>   <extension name="unitest_rad-ANI-auth">
>     <condition field="destination_number" expression="^601$">
>       <!-- <action application="log" data="INFO  Before Auth "/> -->
>
>       <action inline="true" application="set" data="CALLID=h323-conf-id
> =${uuid}"/>
>
>       <action inline="true" application="set" data="USERNAME=1001"/>
>       <action inline="true" application="set" data="PASSWD=1111"/>
>
>
>       <action application="sleep" data="2000"/>
>       <action application="auth_function" data="in ${DIALED_NUMBER}, in
> ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>
>
>     </condition>
>   </extension>
>
>
>
> <configuration name="rad_auth.conf" description="radius authentification
> module">
>   <settings>
>
>   </settings>
>
>   <client>
>     <param name="authserver" value="127.0.0.1:1812:gateway"/>
>     <param name="dictionary"
> value="/usr/local/etc/radiusclient/dictionary"/>
>     <param name="seqfile" value="/var/run/radius.seq"/>
>     <param name="mapfile" value="/usr/local/etc/radiusclient/port-id-map"/>
>     <param name="default_realm" value=""/>
>     <param name="radius_timeout" value="3"/>
>     <param name="radius_retries" value="2"/>
>     <param name="radius_deadtime" value="0"/>
>     <param name="bindaddr" value="*"/>
>   </client>
>
>   <vsas>
>
>
>     <param name="h323-conf-id" id="24" value="CALLID" pec="9" expr="1"
> direction="in"/>
>
>   </vsas>
>  </configuration>
>
>
>
> 2011/8/2 Tihomir Culjaga <tculjaga at gmail.com>
>
>> hi,
>>
>> dictionary.all is just the name of a file containing all attributes i
>> needed at that time.
>>
>> you can include other dictionaries by putting #INCLUDE <pathname> at the
>> end of the dictionary file you reference in rad_auth.conf.xml.
>> if the INCLUDE doesn't work, just append dictionary.cisco to your
>> dictionary file... and make your own file.
>>
>>
>> check inline comments down below...
>>
>>
>> T.
>>
>>
>> On Sun, Jul 31, 2011 at 10:46 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>>
>>> Hello Gurus,
>>>
>>> i met a issue when using
>>> mod_rad_auth(http://wiki.freeswitch.org/wiki/Mod_rad_auth) to works
>>> with freeradius server+mysql for AAA, the details is below, Could
>>> anyone give any hints, Thanks in advance.
>>>
>>> i setup a dial plan "unitest_rad-ANI-auth" as wiki above, however,
>>> when i dialed 601 to trigger the dial plan, the console show errors,
>>> it looks "h323-conf-id" is not in the directory, then i tried to add
>>> this attribute to the dictionary, however, it does not help, in the
>>> wiki, it mentioned the rad_auth.conf.xml contains <param
>>> name="dictionary"
>>> value="/usr/local/etc/radiusclient/dictionary.all"/>, however i did
>>> not find the file "dictionary.all" at that directory, so i use
>>> dictionary. BTW, the freeradius server + mysql works well.
>>>
>>
>> i just appended the information needed into dictionary.all file... (vendor
>> and attribute definition).
>>
>>
>>
>>>
>>> console errors:
>>>
>>> EXECUTE sofia/internal/1001 at 124.193.106.104 auth_function(in , in
>>> 38516060333, in 003282, out AUTH_RESULT)
>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:301 allocate initial
>>> structure.
>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:313 initialzed
>>> configuration.
>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set authserver
>>> := 127.0.0.1:1812:gateway.
>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set dictionary
>>> := /usr/local/etc/radiusclient/dictionary.
>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set seqfile :=
>>> /var/run/radius.seq.
>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set mapfile :=
>>> /usr/local/etc/radiusclient/port-id-map.
>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set default_realm
>>> := .
>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set radius_timeout
>>> := 3.
>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set radius_retries
>>> := 2.
>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set radius_deadtime
>>> := 0.
>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set bindaddr := *.
>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:371 ... radius:
>>> User-Name: 38516060333
>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:380 ... radius:
>>> User-Password: 003282
>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:391 ... radius:
>>> Called-station-Id is empty, ignoring...
>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:413 Handle
>>> attribute: h323-conf-id
>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:428 Unknown attribute:
>>> key:h323-conf-id, not found in dictionary
>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:538 abort sending
>>> radius packet.
>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:546 An error occured
>>> during RADIUS Authentication(RC=-1)
>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:702 An error occured
>>> during radius authorization.
>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  AUTH_RESULT=)
>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  AUTH_RESULT=
>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  billing_model=)
>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  billing_model=
>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  credit_amount=)
>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  credit_amount=
>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  currency=)
>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  currency=
>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  preffered_lang=)
>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  preffered_lang=
>>>
>>> added below in the dictionary(/usr/local/etc/radiusclient/dictionary):
>>>
>>> ATTRIBUTE       h323-conf-id            1008    string
>>>
>>
>> you need the vendor definition as well
>>
>>
>>>
>>>
>>> dial plan:
>>> <extension name="unitest_rad-ANI-auth">
>>>    <condition field="destination_number" expression="^601$">
>>>      <action application="log" data="INFO  Before Auth "/>
>>>
>>>      <action inline="true" application="set"
>>> data="CALLID=h323-conf-id=${uuid}"/>
>>>      <action inline="true" application="set"
>>> data="SERVICENUM=h323-prompt-id=${destination_number}"/>
>>>      <action inline="true" application="set"
>>> data="TRANSACTIONID=h323-ivr-out=transactionID:1234"/>
>>>  <!--      <action inline="true" application="set"
>>> data="CALLINGNUMBER=${caller_id_number}"/> -->
>>>      <action inline="true" application="set"
>>> data="CALLINGNUMBER=38516060333"/>
>>>      <action inline="true" application="set"
>>> data="USERNAME=38516060333"/>
>>>  <!--      <action inline="true" application="set"
>>> data="USERNAME=209354"/> -->
>>>      <action inline="true" application="set" data="PASSWD=003282"/>
>>>  <!--      <action inline="true" application="set"
>>> data="DIALED_NUMBER=16094191500"/>  -->
>>>
>>>      <action application="sleep" data="2000"/>
>>>      <action application="auth_function" data="in ${DIALED_NUMBER},
>>> in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>
>>>
>>>
>>>      <action application="log" data="INFO  AUTH_RESULT=${AUTH_RESULT}"/>
>>>      <action application="log" data="INFO
>>>  billing_model=${billing_model}"/>
>>>      <action application="log" data="INFO
>>>  credit_amount=${credit_amount}"/>
>>>      <action application="log" data="INFO  currency=${currency}"/>
>>>      <action application="log" data="INFO
>>>  preffered_lang=${preffered_lang}"/>
>>>      <action application="log" data="INFO  credit_time=${credit_time}"/>
>>>      <action application="log" data="INFO
>>> h323_ivr_duration=${h323_ivr_duration}"/>
>>>      <action application="log" data="INFO  return_code=${return_code}"/>
>>>      <!-- <action application="execute_extension" data="AUTH XML
>>> default"/> -->
>>>    </condition>
>>>  </extension>
>>>
>>>  radius_cdr.conf.xml:
>>>  <configuration name="radius_cdr.conf" description="RADIUS CDR
>>> Configuration">
>>>
>>>        <settings>
>>>
>>>                <!-- location of the radius dictionary files -->
>>>
>>>                <param name="dictionary"
>>> value="/usr/local/freeswitch/conf/radius/dictionary"/>
>>>
>>>
>> your dictionary file need to contain all the attributes you are trying to
>> use or to include other dictionaries (In this case dictionary.cisco) from
>> the dictionary file you are referencing here.
>>
>>
>>>                <!-- number of retries for each server -->
>>>
>>>                <param name="radius_retries" value="3"/>
>>>
>>>                <!-- number of seconds to wait between retries -->
>>>
>>>                <param name="radius_timeout" value="5"/>
>>>
>>>                <!-- accounting servers, up to 8 allowed -->
>>>
>>>                <!-- value is "host:port:secret", port is optional -->
>>>
>>>                <!-- use IP ADDRESSES, not hostnames -->
>>>
>>>                <param name="acct_server" value="127.0.0.1:1813
>>> :testing123"/>
>>>
>>>
>>>        </settings>
>>>
>>> </configuration>
>>>
>>>  the FS version:
>>>  FreeSWITCH Version 1.0.head (git-492bc6b 2011-07-23 12-53-04 -0400)
>>>
>>>  Regards,
>>>  Charles
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>>
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/2dffd6e7/attachment-0001.html 

From michael.knop at hcu-hamburg.de  Tue Aug  2 12:33:30 2011
From: michael.knop at hcu-hamburg.de (michael knop)
Date: Tue, 02 Aug 2011 10:33:30 +0200
Subject: [Freeswitch-users] ptime
In-Reply-To: <4E367769.9000602@hcu-hamburg.de>
References: <4E367769.9000602@hcu-hamburg.de>
Message-ID: <4E37B65A.20709@hcu-hamburg.de>

Hi,

the following line in $PREFIX/conf/sip_profiles/external.xml fixed my 
Sonus ptime mismatch problem which caused choppy voice:

<param name="inbound-codec-prefs" value="PCMU at 10i,PCMA at 10i"/>

/micha


From leon at scarlet-internet.nl  Tue Aug  2 13:19:02 2011
From: leon at scarlet-internet.nl (Leon de Rooij)
Date: Tue, 2 Aug 2011 11:19:02 +0200
Subject: [Freeswitch-users] Sip Headers advice (Not parsing properly)
In-Reply-To: <CAKbxfG8jYW2QQp1UwhFvsiVzJzs-HgQMchyfBG6QKgOwoKkWmA@mail.gmail.com>
References: <77BDF4BA-21C8-49AF-9C7A-9A02614FE2AB@iwoof.org>
	<CA5CC330.1A029%krice@freeswitch.org>
	<CAKbxfG8jYW2QQp1UwhFvsiVzJzs-HgQMchyfBG6QKgOwoKkWmA@mail.gmail.com>
Message-ID: <64EAB21B-D761-476E-B091-FCCB556CB2E8@scarlet-internet.nl>

Which in turn goes back to rfc822 - ARPA Internet Text Messages

http://www.w3.org/Protocols/rfc822/#z35

section 4.7.4

:-)

On Aug 2, 2011, at 8:10 AM, Anthony Minessale wrote:

> I think the standard for X- is inherited from it's daddy: HTTP and in
> turn from it's daddy: SMTP
> 
> 
> 
> On Mon, Aug 1, 2011 at 8:55 PM, Ken Rice <krice at freeswitch.org> wrote:
>> X- is not covered in any RFCs for SIP, however P- is covered...
>> 
>> X- is as you said kinda along for the ride
>> 
>> 
>> On 8/1/11 8:07 PM, "Andy Spitzer" <woof at iwoof.org> wrote:
>> 
>>> Woof!
>>> 
>>> On Aug 1, 2011, at 18:12 , Anthony Minessale wrote:
>>> 
>>>> maybe you should tell Cisco that they are supposed to start
>>>> nonstandard headers with X-
>>> 
>>> That's always been the convention, but there has been some noise about
>>> deprecating the use of the X- prefex in general  (see
>>> http://tools.ietf.org/html//draft-saintandre-xdash-03).
>>> 
>>> For SIP, I believe the convention was never codified into any standard, and it
>>> just came along for the ride with the other RFCS that it "inherited" from
>>> (such as HTTP).  So much as I like to rag on Cisco for non-standard SIP
>>> implementations, this is not one of them.
>>> 
>>> --Woof!
>>> 
>>> 
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> 
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/b26f5929/attachment.html 

From fieldpeak at gmail.com  Tue Aug  2 13:23:36 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Tue, 2 Aug 2011 17:23:36 +0800
Subject: [Freeswitch-users] Mod_rad_auth issue for FS working with
 FreeRadius server
In-Reply-To: <CAOTmbMyvyDA-CRpaMC=WRGVkiyTG=Rs+n4hyeQWdyTZpqysSzw@mail.gmail.com>
References: <CAOTmbMzvd5OHL8z=_8vu6Orp2_prGxaw1gmXYCM2=2+pgzs_WA@mail.gmail.com>
	<CAJoKKdX01Qc8ibs25PUELi8GrzMDNOrHb_bQvYdq3ZqA4XLy4w@mail.gmail.com>
	<CAOTmbMxL3pLOuw655fyT5VzAON_N0PrDN+-i-++=oZoYU6+Fmg@mail.gmail.com>
	<CAOTmbMyvyDA-CRpaMC=WRGVkiyTG=Rs+n4hyeQWdyTZpqysSzw@mail.gmail.com>
Message-ID: <CAOTmbMwFNMuMF=v27C3314RGmAkEmaK6h0svy-QK-sNDh7ZJgg@mail.gmail.com>

Also I added a log in the source code of mod_rad_auth.c to log the attribute
id,
the FS console log show attrid:589850 as below, however, the id for
h323-conf-id is 24.

2011-08-02 17:16:23.670990 [DEBUG] mod_rad_auth.c:413 Handle attribute:
h323-conf-id
2011-08-02 17:16:23.670990 [ERR] mod_rad_auth.c:423 attrid:589850
2011-08-02 17:16:23.670990 [ERR] mod_rad_auth.c:430 Unknown attribute:
key:h323-conf-id, not found in dictionary
2011-08-02 17:16:23.670990 [DEBUG] mod_rad_auth.c:540 abort sending radius
packet.
2011-08-02 17:16:23.670990 [ERR] mod_rad_auth.c:548 An error occured during
RADIUS Authentication(RC=-1)
2011-08-02 17:16:23.670990 [ERR] mod_rad_auth.c:704 An error occured during
radius authorization.


mod_rad_auth.c
...
                if (PCONFIGVSAS->pec != 0)
                    attrid = PCONFIGVSAS->id | (PCONFIGVSAS->pec << 16);
                else
                    attrid = PCONFIGVSAS->id ;

                switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR,
"attrid:%d\n", attrid);

                pda = rc_dict_getattr(rh, attrid);

                if (pda == NULL)
                {
                    result = ERROR_RC;
                    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR,
"Unknown attribute: key:%s, not found in dictionary\n", PCONFIGVSAS->name);
                    break;
                }

Regards,
Charles

2011/8/2 fieldpeak <fieldpeak at gmail.com>

> i tried change to 'h323-conf-id' to 'h323-call-origin' in
> 02_unitest_rad-ANI-auth.xml, rad_auth.conf.xml, however, it still prompt
> '[ERR] mod_rad_auth.c:428 Unknown attribute: key:h323-conf-id, not found
> in dictionary', so where the mod_rad_auth read out the 'h323-conf-id'? very
> very strange, which dictionary it was using...
>
> Regards,
> Charles
>
>
> 2011/8/2 fieldpeak <fieldpeak at gmail.com>
>
>> Hi Tihomir,
>>
>> Finally the answer coming, i see the hope, thanks for your reply, :)
>>
>> As your advise, i only use one attribute(h323-conf-id) in my dialplan, and
>> only one attribute(h323-conf-id) in rad_auth.conf.xml, and using the
>> attached dictionary (from ciso) which contains this attribute, however, it
>> still prompt 'unknown attribute', so i suspected if it was reading
>> /usr/local/etc/radiusclient/dictionary, so i copy the same dictionary to
>> /usr/local/freeswitch/radius/, it did not any help at all... very strange...
>>
>> Log:
>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set default_realm :=
>> .
>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set radius_timeout
>> := 3.
>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set radius_retries
>> := 2.
>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set radius_deadtime
>> := 0.
>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set bindaddr := *.
>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:371 ... radius:
>> User-Name: 38516060333
>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:380 ... radius:
>> User-Password: 003282
>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:396 ... radius:
>> Called-station-Id: 16094191500
>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:413 Handle attribute:
>> h323-conf-id
>> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:428 Unknown attribute:
>> key:h323-conf-id, not found in dictionary
>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:538 abort sending radius
>> packet.
>> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:546 An error occured
>> during RADIUS Authentication(RC=-1)
>> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:702 An error occured
>> during radius authorization.
>>
>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  AUTH_RESULT=)
>>
>>
>>
>>   <extension name="unitest_rad-ANI-auth">
>>     <condition field="destination_number" expression="^601$">
>>       <!-- <action application="log" data="INFO  Before Auth "/> -->
>>
>>       <action inline="true" application="set" data="CALLID=h323-conf-id
>> =${uuid}"/>
>>
>>       <action inline="true" application="set" data="USERNAME=1001"/>
>>       <action inline="true" application="set" data="PASSWD=1111"/>
>>
>>
>>       <action application="sleep" data="2000"/>
>>       <action application="auth_function" data="in ${DIALED_NUMBER}, in
>> ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>
>>
>>     </condition>
>>   </extension>
>>
>>
>>
>> <configuration name="rad_auth.conf" description="radius authentification
>> module">
>>   <settings>
>>
>>   </settings>
>>
>>   <client>
>>     <param name="authserver" value="127.0.0.1:1812:gateway"/>
>>     <param name="dictionary"
>> value="/usr/local/etc/radiusclient/dictionary"/>
>>     <param name="seqfile" value="/var/run/radius.seq"/>
>>     <param name="mapfile"
>> value="/usr/local/etc/radiusclient/port-id-map"/>
>>     <param name="default_realm" value=""/>
>>     <param name="radius_timeout" value="3"/>
>>     <param name="radius_retries" value="2"/>
>>     <param name="radius_deadtime" value="0"/>
>>     <param name="bindaddr" value="*"/>
>>   </client>
>>
>>   <vsas>
>>
>>
>>     <param name="h323-conf-id" id="24" value="CALLID" pec="9" expr="1"
>> direction="in"/>
>>
>>   </vsas>
>>  </configuration>
>>
>>
>>
>> 2011/8/2 Tihomir Culjaga <tculjaga at gmail.com>
>>
>>> hi,
>>>
>>> dictionary.all is just the name of a file containing all attributes i
>>> needed at that time.
>>>
>>> you can include other dictionaries by putting #INCLUDE <pathname> at the
>>> end of the dictionary file you reference in rad_auth.conf.xml.
>>> if the INCLUDE doesn't work, just append dictionary.cisco to your
>>> dictionary file... and make your own file.
>>>
>>>
>>> check inline comments down below...
>>>
>>>
>>> T.
>>>
>>>
>>> On Sun, Jul 31, 2011 at 10:46 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>>>
>>>> Hello Gurus,
>>>>
>>>> i met a issue when using
>>>> mod_rad_auth(http://wiki.freeswitch.org/wiki/Mod_rad_auth) to works
>>>> with freeradius server+mysql for AAA, the details is below, Could
>>>> anyone give any hints, Thanks in advance.
>>>>
>>>> i setup a dial plan "unitest_rad-ANI-auth" as wiki above, however,
>>>> when i dialed 601 to trigger the dial plan, the console show errors,
>>>> it looks "h323-conf-id" is not in the directory, then i tried to add
>>>> this attribute to the dictionary, however, it does not help, in the
>>>> wiki, it mentioned the rad_auth.conf.xml contains <param
>>>> name="dictionary"
>>>> value="/usr/local/etc/radiusclient/dictionary.all"/>, however i did
>>>> not find the file "dictionary.all" at that directory, so i use
>>>> dictionary. BTW, the freeradius server + mysql works well.
>>>>
>>>
>>> i just appended the information needed into dictionary.all file...
>>> (vendor and attribute definition).
>>>
>>>
>>>
>>>>
>>>> console errors:
>>>>
>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 auth_function(in , in
>>>> 38516060333, in 003282, out AUTH_RESULT)
>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:301 allocate initial
>>>> structure.
>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:313 initialzed
>>>> configuration.
>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set authserver
>>>> := 127.0.0.1:1812:gateway.
>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set dictionary
>>>> := /usr/local/etc/radiusclient/dictionary.
>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set seqfile :=
>>>> /var/run/radius.seq.
>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set mapfile :=
>>>> /usr/local/etc/radiusclient/port-id-map.
>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set default_realm
>>>> := .
>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set radius_timeout
>>>> := 3.
>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set radius_retries
>>>> := 2.
>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>> radius_deadtime := 0.
>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set bindaddr := *.
>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:371 ... radius:
>>>> User-Name: 38516060333
>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:380 ... radius:
>>>> User-Password: 003282
>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:391 ... radius:
>>>> Called-station-Id is empty, ignoring...
>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:413 Handle
>>>> attribute: h323-conf-id
>>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:428 Unknown attribute:
>>>> key:h323-conf-id, not found in dictionary
>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:538 abort sending
>>>> radius packet.
>>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:546 An error occured
>>>> during RADIUS Authentication(RC=-1)
>>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:702 An error occured
>>>> during radius authorization.
>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  AUTH_RESULT=)
>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  AUTH_RESULT=
>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  billing_model=)
>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  billing_model=
>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  credit_amount=)
>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  credit_amount=
>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  currency=)
>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  currency=
>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  preffered_lang=)
>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  preffered_lang=
>>>>
>>>> added below in the dictionary(/usr/local/etc/radiusclient/dictionary):
>>>>
>>>> ATTRIBUTE       h323-conf-id            1008    string
>>>>
>>>
>>> you need the vendor definition as well
>>>
>>>
>>>>
>>>>
>>>> dial plan:
>>>> <extension name="unitest_rad-ANI-auth">
>>>>    <condition field="destination_number" expression="^601$">
>>>>      <action application="log" data="INFO  Before Auth "/>
>>>>
>>>>      <action inline="true" application="set"
>>>> data="CALLID=h323-conf-id=${uuid}"/>
>>>>      <action inline="true" application="set"
>>>> data="SERVICENUM=h323-prompt-id=${destination_number}"/>
>>>>      <action inline="true" application="set"
>>>> data="TRANSACTIONID=h323-ivr-out=transactionID:1234"/>
>>>>  <!--      <action inline="true" application="set"
>>>> data="CALLINGNUMBER=${caller_id_number}"/> -->
>>>>      <action inline="true" application="set"
>>>> data="CALLINGNUMBER=38516060333"/>
>>>>      <action inline="true" application="set"
>>>> data="USERNAME=38516060333"/>
>>>>  <!--      <action inline="true" application="set"
>>>> data="USERNAME=209354"/> -->
>>>>      <action inline="true" application="set" data="PASSWD=003282"/>
>>>>  <!--      <action inline="true" application="set"
>>>> data="DIALED_NUMBER=16094191500"/>  -->
>>>>
>>>>      <action application="sleep" data="2000"/>
>>>>      <action application="auth_function" data="in ${DIALED_NUMBER},
>>>> in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>
>>>>
>>>>
>>>>      <action application="log" data="INFO  AUTH_RESULT=${AUTH_RESULT}"/>
>>>>      <action application="log" data="INFO
>>>>  billing_model=${billing_model}"/>
>>>>      <action application="log" data="INFO
>>>>  credit_amount=${credit_amount}"/>
>>>>      <action application="log" data="INFO  currency=${currency}"/>
>>>>      <action application="log" data="INFO
>>>>  preffered_lang=${preffered_lang}"/>
>>>>      <action application="log" data="INFO  credit_time=${credit_time}"/>
>>>>      <action application="log" data="INFO
>>>> h323_ivr_duration=${h323_ivr_duration}"/>
>>>>      <action application="log" data="INFO  return_code=${return_code}"/>
>>>>      <!-- <action application="execute_extension" data="AUTH XML
>>>> default"/> -->
>>>>    </condition>
>>>>  </extension>
>>>>
>>>>  radius_cdr.conf.xml:
>>>>  <configuration name="radius_cdr.conf" description="RADIUS CDR
>>>> Configuration">
>>>>
>>>>        <settings>
>>>>
>>>>                <!-- location of the radius dictionary files -->
>>>>
>>>>                <param name="dictionary"
>>>> value="/usr/local/freeswitch/conf/radius/dictionary"/>
>>>>
>>>>
>>> your dictionary file need to contain all the attributes you are trying to
>>> use or to include other dictionaries (In this case dictionary.cisco) from
>>> the dictionary file you are referencing here.
>>>
>>>
>>>>                <!-- number of retries for each server -->
>>>>
>>>>                <param name="radius_retries" value="3"/>
>>>>
>>>>                <!-- number of seconds to wait between retries -->
>>>>
>>>>                <param name="radius_timeout" value="5"/>
>>>>
>>>>                <!-- accounting servers, up to 8 allowed -->
>>>>
>>>>                <!-- value is "host:port:secret", port is optional -->
>>>>
>>>>                <!-- use IP ADDRESSES, not hostnames -->
>>>>
>>>>                <param name="acct_server" value="127.0.0.1:1813
>>>> :testing123"/>
>>>>
>>>>
>>>>        </settings>
>>>>
>>>> </configuration>
>>>>
>>>>  the FS version:
>>>>  FreeSWITCH Version 1.0.head (git-492bc6b 2011-07-23 12-53-04 -0400)
>>>>
>>>>  Regards,
>>>>  Charles
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>>
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/8d314741/attachment-0001.html 

From sascha.daniels at amooma.de  Tue Aug  2 13:59:44 2011
From: sascha.daniels at amooma.de (Sascha Daniels)
Date: Tue, 02 Aug 2011 11:59:44 +0200
Subject: [Freeswitch-users] Limit number of voicemails
Message-ID: <4E37CA90.4090806@amooma.de>

Hi together.

For a small appliance I need to limit the number of voicemails for each
user.

I know that I can limit the length. That is the first step.

Is there a way to disable the message recording, when the maximum number
of voicemails is reached?

Unfortunately I didn't find anything in the documentation.

Kind regards

                        Sascha

-- 
AMOOMA GmbH - Bachstr. 124 - 56566 Neuwied  -->  http://www.amooma.de
Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister Montabaur B14998

B?cher: http://das-asterisk-buch.de - http://ruby-auf-schienen.de



From lloydie.t at gmail.com  Tue Aug  2 14:41:45 2011
From: lloydie.t at gmail.com (lloyd thomas)
Date: Tue, 2 Aug 2011 11:41:45 +0100
Subject: [Freeswitch-users] Help setting up SIP reg
In-Reply-To: <CAKzWOxXYi-yO9DNDw1rHCduKq=h9q=BddFmn2dKuqALKb=t1fA@mail.gmail.com>
References: <CAE=3qOYPpTuRTvq7RTJeUgzsCtvrfCjMiKFGu1fg0PHyZRE+-g@mail.gmail.com>
	<CAFiqYu=c8mY5BMWPumy1CgZHZ81kMRPgoNiF-6mjFAkW1q30fA@mail.gmail.com>
	<CAE=3qOZxwERT3CQEFHetO_o2s=aQUwYZ54TmMRPx9eYD=A7NoQ@mail.gmail.com>
	<CAKzWOxUEsGUKojF-vvstL6FvcDMhePqYGdXEQXQ4x-EJ81Bk6w@mail.gmail.com>
	<CAE=3qOacZh50iepxJKJD5UvtesxZAsN7RnPKNe1zdLO9VmFsuA@mail.gmail.com>
	<CAE=3qOZsNuiimFBZJTf4SbPRA8bRq_hJfPUunEnT8rxZj3qTnA@mail.gmail.com>
	<CAE=3qObVuLOhp2GD29Da-GfzB_qkbkyDP4kPmRxjdbJi7XeirQ@mail.gmail.com>
	<CAKzWOxUyZoBS5Evp4b61vP8fm64Lef0n01ti3KmVraQO_BqYtA@mail.gmail.com>
	<CAE=3qObsDBDTsKY2ycK_NierbS_A+-f896yMW4OArTkgauYgRA@mail.gmail.com>
	<CAKzWOxXYi-yO9DNDw1rHCduKq=h9q=BddFmn2dKuqALKb=t1fA@mail.gmail.com>
Message-ID: <CAE=3qObVbDQi6g1m80WLtG5MhyosV2YYnqmOoausiz-yC+NmQQ@mail.gmail.com>

Just did a test, but no joy. I suspect I may have to dispense with the
gateway settings and just bridge straight from the dial plan, but it is just
a guess.


dialplan
----------------------------------------
<extension name="Local Dial">
   <condition field="destination_number" expression="^9([2-9][0-9]{5})$">
     <action application="bridge"
data="sofia/gateway/phisys-2circles/01869$1"/>
   </condition>
</extension>

gateway
---------------------------------------
<include>
  <gateway name="phisys-2circles">
    <param name="from-domain" value="80.40.150.150"/>
    <param name="proxy" value="80.40.150.150"/>
    <param name="sip-port" value="5060"/>
  </gateway>
</include>



errors
------------------------------------
2011-08-02 11:32:42.056302 [DEBUG] mod_dptools.c:1059 sofia/internal/
200 at phisys.tele.phi.co.uk SET [RFC2822_DATE]=[Tue, 02 Aug 2011 11:32:42
+0100]
EXECUTE sofia/internal/200 at phisys.tele.phi.co.ukbridge(sofia/gateway/phisys-2circles/01869321110)
2011-08-02 11:32:42.073667 [ERR] mod_sofia.c:3940 Invalid Gateway
2011-08-02 11:32:42.073667 [NOTICE] mod_sofia.c:4282 Close Channel N/A
[CS_NEW]
2011-08-02 11:32:42.076523 [DEBUG] switch_core_state_machine.c:452 ()
Running State Change CS_DESTROY
2011-08-02 11:32:42.079574 [DEBUG] switch_core_state_machine.c:462 (N/A)
State DESTROY
2011-08-02 11:32:42.079574 [DEBUG] mod_sofia.c:362 N/A SOFIA DESTROY
2011-08-02 11:32:42.081946 [DEBUG] switch_core_state_machine.c:462 (N/A)
State DESTROY going to sleep
2011-08-02 11:32:42.084166 [ERR] switch_ivr_originate.c:2640 Cannot create
outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]
2011-08-02 11:32:42.085649 [DEBUG] switch_ivr_originate.c:3506 Originate
Resulted in Error Cause: 28 [INVALID_NUMBER_FORMAT]
2011-08-02 11:32:42.087275 [INFO] mod_dptools.c:2623 Originate Failed.
Cause: INVALID_NUMBER_FORMAT
2011-08-02 11:32:42.088652 [DEBUG] switch_channel.c:2559 (sofia/internal/
200 at phisys.tele.phi.co.uk) Callstate Change RINGING -> HANGUP
2011-08-02 11:32:42.093248 [NOTICE] mod_dptools.c:2686 Hangup
sofia/internal/200 at phisys.tele.phi.co.uk [CS_EXECUTE]
[INVALID_NUMBER_FORMAT]
2011-08-02 11:32:42.096925 [DEBUG] switch_channel.c:2575 Send signal
sofia/internal/200 at phisys.tele.phi.co.uk [KILL]

On 1 August 2011 20:50, Michael Collins <msc at freeswitch.org> wrote:

> Okay, so what happens when you dial out? Sorry, it's been a few days and I
> don't recall where we left off. Be sure to include console log w/ siptrace
> on pastebin.freeswitch.org.
>
> -MC
>
>
> On Mon, Aug 1, 2011 at 12:35 PM, lloyd thomas <lloydie.t at gmail.com> wrote:
>
>> I think they have my IP on a white list.
>>
>>
>> On 1 August 2011 16:24, Michael Collins <msc at freeswitch.org> wrote:
>>
>>> Do they challenge you (digest auth) or do they have your IP address on a
>>> white list? That's a critical piece of information that only your provider
>>> can supply.
>>>
>>> -MC
>>>
>>>
>>> On Fri, Jul 29, 2011 at 9:31 PM, lloyd thomas <lloydie.t at gmail.com>wrote:
>>>
>>>> OK Inbound working with:
>>>>
>>>>   <!--2circles -->
>>>>     <list name="domains" default="allow">
>>>>       <node type="allow" cidr="80.40.150.150/32"/>
>>>>     </list>
>>>>
>>>> Just need to sort outbound.
>>>>
>>>>
>>>> On 30 July 2011 04:59, lloyd thomas <lloydie.t at gmail.com> wrote:
>>>>
>>>>> Hi, dialling in produces the following error.
>>>>>
>>>>> 2011-07-30 04:56:07.818936 [DEBUG] sofia.c:6517 IP 80.40.150.150
>>>>> Rejected by acl "domains". Falling back to Digest auth.
>>>>> 2011-07-30 04:56:07.826367 [WARNING] sofia_reg.c:1246 SIP auth
>>>>> challenge (INVITE) on sofia profile 'internal' for
>>>>> [01869******@172.16.XXX.XXX] from ip 80.40.150.150
>>>>>
>>>>>
>>>>>
>>>>> On 30 July 2011 04:34, lloyd thomas <lloydie.t at gmail.com> wrote:
>>>>>
>>>>>> I am registering with a them. I could not find suitable example in
>>>>>> http://wiki.freeswitch.org/wiki/SIP_Provider_Examples which
>>>>>>
>>>>>>
>>>>>> On 29 July 2011 21:57, Michael Collins <msc at freeswitch.org> wrote:
>>>>>>
>>>>>>> Are you registering with the provider or are they registering with
>>>>>>> you? If they register with you then a user example is appropriate. If you
>>>>>>> are registering with them then all you need is a gateway configured.
>>>>>>> -MC
>>>>>>>
>>>>>>>
>>>>>>> On Fri, Jul 29, 2011 at 1:40 PM, lloyd thomas <lloydie.t at gmail.com>wrote:
>>>>>>>
>>>>>>>> Sorry, example is not clear to me.
>>>>>>>> I don't understand why a user config is relevant to sip registration
>>>>>>>> for a provider.
>>>>>>>> An example will help me more. Maybe CIDR attribute in a sip_profile
>>>>>>>> gateway could help.
>>>>>>>>
>>>>>>>>
>>>>>>>> On 29 July 2011 19:55, Steven Ayre <steveayre at gmail.com> wrote:
>>>>>>>>
>>>>>>>>> Look at the cidr attribute in the user directory to authenticate by
>>>>>>>>> IP:
>>>>>>>>> http://wiki.freeswitch.org/wiki/Acl#Users
>>>>>>>>>
>>>>>>>>> -Steve
>>>>>>>>>
>>>>>>>>> On 29 July 2011 19:38, lloyd thomas <lloydie.t at gmail.com> wrote:
>>>>>>>>>
>>>>>>>>>> *Hi I need a little help setting up a SIP registration for a
>>>>>>>>>> provider that does not use auth.*
>>>>>>>>>>
>>>>>>>>>> *All I have is info below.*
>>>>>>>>>> **
>>>>>>>>>>
>>>>>>>>>> *
>>>>>>>>>> *
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> SBC/Proxy IP:        80.40.150.150:5060
>>>>>>>>>>
>>>>>>>>>> Authentication:      Trusted IP ? 88.221.85.33
>>>>>>>>>>
>>>>>>>>>> Assigned DDI:        01869******, 01869******
>>>>>>>>>>
>>>>>>>>>> DTMF Method:         RFC2833
>>>>>>>>>>
>>>>>>>>>> Status:              Live
>>>>>>>>>>
>>>>>>>>>> No. of trunks:       2x
>>>>>>>>>>
>>>>>>>>>> Session Timer:       1800
>>>>>>>>>>
>>>>>>>>>> Profile*:*             Generic (35060)
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> Apparently the following is used for *
>>>>>>>>>>
>>>>>>>>>> [vibe]
>>>>>>>>>>
>>>>>>>>>> type = friend
>>>>>>>>>>
>>>>>>>>>> host = 80.40.150.150
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/ce0609a1/attachment-0001.html 

From ovvenkatesan at gmail.com  Tue Aug  2 15:45:23 2011
From: ovvenkatesan at gmail.com (ovvenkat)
Date: Tue, 2 Aug 2011 17:15:23 +0530
Subject: [Freeswitch-users] freetdm error message
Message-ID: <CAABykDrOSPH4krH45EjxURkXOFJAGMV6q0T3CehGEQcgKzPfxQ@mail.gmail.com>

Hi to all,

In my "fs_cli"  I am getting bellow message very frequently.
Can you any one help me where is the error?

2011-08-02 17:11:33.028452 [INFO] ftmod_sangoma_isdn_stack_rcv.c:995
sng_isdn->s1: Invalid Q.921/Q.931 frame - ignoring len:1
2011-08-02 17:11:33.048455 [INFO] ftmod_sangoma_isdn_stack_rcv.c:995
sng_isdn->s1: Invalid Q.921/Q.931 frame - ignoring len:1
2011-08-02 17:11:33.048455 [INFO] ftmod_sangoma_isdn_stack_rcv.c:995
sng_isdn->s1: Invalid Q.921/Q.931 frame - ignoring len:1
2011-08-02 17:11:33.068455 [INFO] ftmod_sangoma_isdn_stack_rcv.c:995
sng_isdn->s1: Invalid Q.921/Q.931 frame - ignoring len:1
2011-08-02 17:11:33.068455 [INFO] ftmod_sangoma_isdn_stack_rcv.c:995
sng_isdn->s1: Invalid Q.921/Q.931 frame - ignoring len:1
2011-08-02 17:11:33.068455 [INFO] ftmod_sangoma_isdn_stack_rcv.c:995
sng_isdn->s1: Invalid Q.921/Q.931 frame - ignoring len:1



--
Regards
Venkatesan OV.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/e9794851/attachment.html 

From gmaruzz at gmail.com  Tue Aug  2 16:05:11 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 2 Aug 2011 14:05:11 +0200
Subject: [Freeswitch-users] freetdm error message
In-Reply-To: <CAABykDrOSPH4krH45EjxURkXOFJAGMV6q0T3CehGEQcgKzPfxQ@mail.gmail.com>
References: <CAABykDrOSPH4krH45EjxURkXOFJAGMV6q0T3CehGEQcgKzPfxQ@mail.gmail.com>
Message-ID: <CALXCt0rCV+FE+1i3S=0NpQVdURZsOwFL0ATkg0_cJFQ5dRwCzg@mail.gmail.com>

On 8/2/11, ovvenkat <ovvenkatesan at gmail.com> wrote:
> Hi to all,
>
> In my "fs_cli"  I am getting bellow message very frequently.
> Can you any one help me where is the error?
>
> 2011-08-02 17:11:33.028452 [INFO] ftmod_sangoma_isdn_stack_rcv.c:995
> sng_isdn->s1: Invalid Q.921/Q.931 frame - ignoring len:1
> 2011-08-02 17:11:33.048455 [INFO] ftmod_sangoma_isdn_stack_rcv.c:995
> sng_isdn->s1: Invalid Q.921/Q.931 frame - ignoring len:1
> 2011-08-02 17:11:33.048455 [INFO] ftmod_sangoma_isdn_stack_rcv.c:995
> sng_isdn->s1: Invalid Q.921/Q.931 frame - ignoring len:1
> 2011-08-02 17:11:33.068455 [INFO] ftmod_sangoma_isdn_stack_rcv.c:995
> sng_isdn->s1: Invalid Q.921/Q.931 frame - ignoring len:1
> 2011-08-02 17:11:33.068455 [INFO] ftmod_sangoma_isdn_stack_rcv.c:995
> sng_isdn->s1: Invalid Q.921/Q.931 frame - ignoring len:1
> 2011-08-02 17:11:33.068455 [INFO] ftmod_sangoma_isdn_stack_rcv.c:995
> sng_isdn->s1: Invalid Q.921/Q.931 frame - ignoring len:1
>
>
>
> --
> Regards
> Venkatesan OV.
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From steveayre at gmail.com  Tue Aug  2 16:48:07 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 2 Aug 2011 13:48:07 +0100
Subject: [Freeswitch-users] Help setting up SIP reg
In-Reply-To: <CAE=3qObVbDQi6g1m80WLtG5MhyosV2YYnqmOoausiz-yC+NmQQ@mail.gmail.com>
References: <CAE=3qOYPpTuRTvq7RTJeUgzsCtvrfCjMiKFGu1fg0PHyZRE+-g@mail.gmail.com>
	<CAFiqYu=c8mY5BMWPumy1CgZHZ81kMRPgoNiF-6mjFAkW1q30fA@mail.gmail.com>
	<CAE=3qOZxwERT3CQEFHetO_o2s=aQUwYZ54TmMRPx9eYD=A7NoQ@mail.gmail.com>
	<CAKzWOxUEsGUKojF-vvstL6FvcDMhePqYGdXEQXQ4x-EJ81Bk6w@mail.gmail.com>
	<CAE=3qOacZh50iepxJKJD5UvtesxZAsN7RnPKNe1zdLO9VmFsuA@mail.gmail.com>
	<CAE=3qOZsNuiimFBZJTf4SbPRA8bRq_hJfPUunEnT8rxZj3qTnA@mail.gmail.com>
	<CAE=3qObVuLOhp2GD29Da-GfzB_qkbkyDP4kPmRxjdbJi7XeirQ@mail.gmail.com>
	<CAKzWOxUyZoBS5Evp4b61vP8fm64Lef0n01ti3KmVraQO_BqYtA@mail.gmail.com>
	<CAE=3qObsDBDTsKY2ycK_NierbS_A+-f896yMW4OArTkgauYgRA@mail.gmail.com>
	<CAKzWOxXYi-yO9DNDw1rHCduKq=h9q=BddFmn2dKuqALKb=t1fA@mail.gmail.com>
	<CAE=3qObVbDQi6g1m80WLtG5MhyosV2YYnqmOoausiz-yC+NmQQ@mail.gmail.com>
Message-ID: <CAFiqYukcmMdBR44JNBK73wic9asQO04gi=jekf2BUo_qRdONWw@mail.gmail.com>

>
> EXECUTE sofia/internal/200 at phisys.tele.phi.co.ukbridge(sofia/gateway/phisys-2circles/01869321110)
> 2011-08-02 11:32:42.073667 [ERR] mod_sofia.c:3940 Invalid Gateway
>

This error message means no gateway named 'phisys-2circles' has been loaded.
It's a configuration problem.

Connect via fs_cli and run the command 'sofia status'. Do you see the
gateway listed?

Also try posting on pastebin the configuration file
/usr/local/freeswitch/log/freeswitch.fsxml - it's created by FreeSWITCH when
it starts and is the flat XML file generated after all includes have been
done. If it's a syntax problem or file not being included that should let us
see where the config error lies.

Using a gateway will work, once the config problem is fixed.

-Steve




On 2 August 2011 11:41, lloyd thomas <lloydie.t at gmail.com> wrote:

> Just did a test, but no joy. I suspect I may have to dispense with the
> gateway settings and just bridge straight from the dial plan, but it is just
> a guess.
>
>
> dialplan
> ----------------------------------------
> <extension name="Local Dial">
>    <condition field="destination_number" expression="^9([2-9][0-9]{5})$">
>      <action application="bridge"
> data="sofia/gateway/phisys-2circles/01869$1"/>
>    </condition>
> </extension>
>
> gateway
> ---------------------------------------
> <include>
>   <gateway name="phisys-2circles">
>     <param name="from-domain" value="80.40.150.150"/>
>     <param name="proxy" value="80.40.150.150"/>
>     <param name="sip-port" value="5060"/>
>   </gateway>
> </include>
>
>
>
> errors
> ------------------------------------
> 2011-08-02 11:32:42.056302 [DEBUG] mod_dptools.c:1059 sofia/internal/
> 200 at phisys.tele.phi.co.uk SET [RFC2822_DATE]=[Tue, 02 Aug 2011 11:32:42
> +0100]
> EXECUTE sofia/internal/200 at phisys.tele.phi.co.ukbridge(sofia/gateway/phisys-2circles/01869321110)
> 2011-08-02 11:32:42.073667 [ERR] mod_sofia.c:3940 Invalid Gateway
> 2011-08-02 11:32:42.073667 [NOTICE] mod_sofia.c:4282 Close Channel N/A
> [CS_NEW]
> 2011-08-02 11:32:42.076523 [DEBUG] switch_core_state_machine.c:452 ()
> Running State Change CS_DESTROY
> 2011-08-02 11:32:42.079574 [DEBUG] switch_core_state_machine.c:462 (N/A)
> State DESTROY
> 2011-08-02 11:32:42.079574 [DEBUG] mod_sofia.c:362 N/A SOFIA DESTROY
> 2011-08-02 11:32:42.081946 [DEBUG] switch_core_state_machine.c:462 (N/A)
> State DESTROY going to sleep
> 2011-08-02 11:32:42.084166 [ERR] switch_ivr_originate.c:2640 Cannot create
> outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]
> 2011-08-02 11:32:42.085649 [DEBUG] switch_ivr_originate.c:3506 Originate
> Resulted in Error Cause: 28 [INVALID_NUMBER_FORMAT]
> 2011-08-02 11:32:42.087275 [INFO] mod_dptools.c:2623 Originate Failed.
> Cause: INVALID_NUMBER_FORMAT
> 2011-08-02 11:32:42.088652 [DEBUG] switch_channel.c:2559 (sofia/internal/
> 200 at phisys.tele.phi.co.uk) Callstate Change RINGING -> HANGUP
> 2011-08-02 11:32:42.093248 [NOTICE] mod_dptools.c:2686 Hangup
> sofia/internal/200 at phisys.tele.phi.co.uk [CS_EXECUTE]
> [INVALID_NUMBER_FORMAT]
> 2011-08-02 11:32:42.096925 [DEBUG] switch_channel.c:2575 Send signal
> sofia/internal/200 at phisys.tele.phi.co.uk [KILL]
>
>
> On 1 August 2011 20:50, Michael Collins <msc at freeswitch.org> wrote:
>
>> Okay, so what happens when you dial out? Sorry, it's been a few days and I
>> don't recall where we left off. Be sure to include console log w/ siptrace
>> on pastebin.freeswitch.org.
>>
>> -MC
>>
>>
>> On Mon, Aug 1, 2011 at 12:35 PM, lloyd thomas <lloydie.t at gmail.com>wrote:
>>
>>> I think they have my IP on a white list.
>>>
>>>
>>> On 1 August 2011 16:24, Michael Collins <msc at freeswitch.org> wrote:
>>>
>>>> Do they challenge you (digest auth) or do they have your IP address on a
>>>> white list? That's a critical piece of information that only your provider
>>>> can supply.
>>>>
>>>> -MC
>>>>
>>>>
>>>> On Fri, Jul 29, 2011 at 9:31 PM, lloyd thomas <lloydie.t at gmail.com>wrote:
>>>>
>>>>> OK Inbound working with:
>>>>>
>>>>>   <!--2circles -->
>>>>>     <list name="domains" default="allow">
>>>>>       <node type="allow" cidr="80.40.150.150/32"/>
>>>>>     </list>
>>>>>
>>>>> Just need to sort outbound.
>>>>>
>>>>>
>>>>> On 30 July 2011 04:59, lloyd thomas <lloydie.t at gmail.com> wrote:
>>>>>
>>>>>> Hi, dialling in produces the following error.
>>>>>>
>>>>>> 2011-07-30 04:56:07.818936 [DEBUG] sofia.c:6517 IP 80.40.150.150
>>>>>> Rejected by acl "domains". Falling back to Digest auth.
>>>>>> 2011-07-30 04:56:07.826367 [WARNING] sofia_reg.c:1246 SIP auth
>>>>>> challenge (INVITE) on sofia profile 'internal' for
>>>>>> [01869******@172.16.XXX.XXX] from ip 80.40.150.150
>>>>>>
>>>>>>
>>>>>>
>>>>>> On 30 July 2011 04:34, lloyd thomas <lloydie.t at gmail.com> wrote:
>>>>>>
>>>>>>> I am registering with a them. I could not find suitable example in
>>>>>>> http://wiki.freeswitch.org/wiki/SIP_Provider_Examples which
>>>>>>>
>>>>>>>
>>>>>>> On 29 July 2011 21:57, Michael Collins <msc at freeswitch.org> wrote:
>>>>>>>
>>>>>>>> Are you registering with the provider or are they registering with
>>>>>>>> you? If they register with you then a user example is appropriate. If you
>>>>>>>> are registering with them then all you need is a gateway configured.
>>>>>>>> -MC
>>>>>>>>
>>>>>>>>
>>>>>>>> On Fri, Jul 29, 2011 at 1:40 PM, lloyd thomas <lloydie.t at gmail.com>wrote:
>>>>>>>>
>>>>>>>>> Sorry, example is not clear to me.
>>>>>>>>> I don't understand why a user config is relevant to sip
>>>>>>>>> registration for a provider.
>>>>>>>>> An example will help me more. Maybe CIDR attribute in a sip_profile
>>>>>>>>> gateway could help.
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On 29 July 2011 19:55, Steven Ayre <steveayre at gmail.com> wrote:
>>>>>>>>>
>>>>>>>>>> Look at the cidr attribute in the user directory to authenticate
>>>>>>>>>> by IP:
>>>>>>>>>> http://wiki.freeswitch.org/wiki/Acl#Users
>>>>>>>>>>
>>>>>>>>>> -Steve
>>>>>>>>>>
>>>>>>>>>> On 29 July 2011 19:38, lloyd thomas <lloydie.t at gmail.com> wrote:
>>>>>>>>>>
>>>>>>>>>>> *Hi I need a little help setting up a SIP registration for a
>>>>>>>>>>> provider that does not use auth.*
>>>>>>>>>>>
>>>>>>>>>>> *All I have is info below.*
>>>>>>>>>>> **
>>>>>>>>>>>
>>>>>>>>>>> *
>>>>>>>>>>> *
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> SBC/Proxy IP:        80.40.150.150:5060
>>>>>>>>>>>
>>>>>>>>>>> Authentication:      Trusted IP ? 88.221.85.33
>>>>>>>>>>>
>>>>>>>>>>> Assigned DDI:        01869******, 01869******
>>>>>>>>>>>
>>>>>>>>>>> DTMF Method:         RFC2833
>>>>>>>>>>>
>>>>>>>>>>> Status:              Live
>>>>>>>>>>>
>>>>>>>>>>> No. of trunks:       2x
>>>>>>>>>>>
>>>>>>>>>>> Session Timer:       1800
>>>>>>>>>>>
>>>>>>>>>>> Profile*:*             Generic (35060)
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> Apparently the following is used for *
>>>>>>>>>>>
>>>>>>>>>>> [vibe]
>>>>>>>>>>>
>>>>>>>>>>> type = friend
>>>>>>>>>>>
>>>>>>>>>>> host = 80.40.150.150
>>>>>>>>>>>
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>>
>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/aa26c01f/attachment-0001.html 

From dgarcia at anew.com.ve  Tue Aug  2 17:42:43 2011
From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar)
Date: Tue, 02 Aug 2011 09:12:43 -0430
Subject: [Freeswitch-users] Fidelio
In-Reply-To: <CABH8QX0cgqbPcCtS=2Z0ron_vqX=5-YMrnkGJgqsOYXx81+x9Q@mail.gmail.com>
References: <CAHKqMb5dnE1mgdFaV65R=tk-zxvfz30=hi+3AyxezENQXVsrYw@mail.gmail.com>
	<CAB2KLyfDvLjmqUU2coOJzG6ss6xzKvsGD5OOsS+bVwKaXx+P0g@mail.gmail.com>
	<CAFiqYukzYyKZ_Tj8w8PFrM4FFpXzZiguRYNfqP8H=ah0iZLkhA@mail.gmail.com>
	<CAB2KLycjzFtLc8jTBM9377CDi488K9nDmVD=Zz-13MZ7_w-cxA@mail.gmail.com>
	<CABH8QX2y1mu1L=9-KZk2HH-Xh9MFOS98p4ctvvQrF7AW_RJV_A@mail.gmail.com>
	<CAHKqMb5uAypVLHgqkt+GFNC_oj5wDPnFbzKHxjwr0QhcEhYJPw@mail.gmail.com>
	<CABH8QX1HHvqL_YK0rbs-m1Ky1GtbZ6OQKyKMKkAne8XV5WBSig@mail.gmail.com>
	<CAHKqMb51RbLBo+pmKwhJ82=ZWaGcyM6bd6y881Vh=4cAjztt5A@mail.gmail.com>
	<CABH8QX0cgqbPcCtS=2Z0ron_vqX=5-YMrnkGJgqsOYXx81+x9Q@mail.gmail.com>
Message-ID: <4E37FED3.7080306@anew.com.ve>

Hello all
If you have money to afford the certification process go ahead, at the 
end you will get back the money if you see that you can put a hotel 
solution based on FS in your local market.

I am new in FS world. I barely

I used to work with Alcatel-Lucent OXE (IP-PBX). First the OXE, by 
itself, offer hotel funcionality: voicemail for guest, guest room admin 
(ready/free, cleaning, occupied,etc), alarm clock (at the hour 
programmed by the guest from their phone or for "hotel phone console" or 
admin interfase; the phone ring), fax service, room billing (quite 
different fron a normal cdr), etc, etc, etc. Second, OXE offer a 
propetary protocol to make possible PMS like fidelio to interface with, 
that means, Fidelio become the administrative interface, so from fidelio 
control room status, services, etc.

So the first, do you have to adapt FS to offer a minimum of hotel 
funcionality. Second about make a PMS-FS integration could be two 
options or more: 1) adapt FS to PMS (using the integration doc); 2) many 
pbx vendors offer info about their hotel link protocol for free (not 
Alcatel), and are supported by fidelio like, OXE does, so you could 
develop a interface that emulate an already supported hotel link towards 
Fidelio and FS.

I have a question:  has someone work with FS to offer an hotel solution?



On 8/1/2011 9:19 PM, Jo?o Mesquita wrote:
> Like I said, if you can raise enough money to be able to get the 
> certification going, I can talk to the ppl I know on Micros and see if 
> we can get FS certified (no guarantees), but it will cost at least 5k 
> USD, so I really don't know if there is enough interest.
>
> Regards,
> Jo?o Mesquita
>
>
>
> On Mon, Aug 1, 2011 at 10:36 PM, Nandy Dagondon <gcd at i.ph 
> <mailto:gcd at i.ph>> wrote:
>
>     yes, certification is big obstacle. too much for our intended
>     project.  perhaps FS can apply for certification as a platform not
>     on a per-company/model basis. i just hope Fidelio would be open to
>     the idea.
>
>     another option would be to ask Asterisk-FIAS connectors like
>     PBillX the possibility to include FreeSwitch.
>
>     2011/8/2 Jo?o Mesquita <jmesquita at freeswitch.org
>     <mailto:jmesquita at freeswitch.org>>
>
>         I think there are none doing this at this moment. Here's what
>         I know tho:
>
>         1. I haven't seen the contracts or legal papers involved in
>         the certification
>         2. I've never done a connector even tho I worked with
>         companies that did in the past
>         3. I know that the homologation/certification costs money and
>         not little money. As far as I know, there are differences
>         between countries when it comes to the fee to be paid. Here is
>         Argentina it is something close to USD$5.000,00.
>         4. For you to be officially certified by them, you need to
>         have your system installed on at least 3 hotels with Fidelio,
>         otherwise you just lost money.
>
>         That should pretty much give you a hint of what you are facing
>         when it comes to the certification process. Does your project
>         accomodate these imposed barriers? If they do, I might be able
>         to help.
>
>         Regards,
>         Jo?o Mesquita
>
>
>
>
>         On Mon, Aug 1, 2011 at 9:38 PM, Nandy Dagondon <gcd at i.ph
>         <mailto:gcd at i.ph>> wrote:
>
>             hi Jo?o,
>
>             since you had FIAS experience, would a commercial
>              mod_fias license/certification be possible just like g.729?
>
>             re companies selling FIAS connectors, i can't find one
>             that connects FS to Fidelio. otherwise, they would sound
>             off in this thread.
>
>             -nandy
>
>             2011/7/6 Jo?o Mesquita <jmesquita at freeswitch.org
>             <mailto:jmesquita at freeswitch.org>>
>
>                 Guys, be careful because I think this document as well
>                 as the protocol are confidential. I had to sign an NDA
>                 with Fidelio to get my hands on it and pay a fee for
>                 it as well. You might as well confirm it since you all
>                 seem to be in the US where this type of information
>                 might be easier to get.
>
>                 There are LOTS of companies selling their connectors
>                 to Fidelio...
>
>                 One other point is that you need to have the
>                 certification with them to be considered compatible,
>                 otherwise, no consultant will install the connector on
>                 the fidelio side.
>
>                 Regards,
>                 Jo?o Mesquita
>
>
>
>
>                 On Tue, Jul 5, 2011 at 1:52 PM, Luis F Urrea
>                 <lfurrea at gmail.com <mailto:lfurrea at gmail.com>> wrote:
>
>                     Awesome! great suggestions to get started,
>
>                     There is also a FIAS simulator floating around.
>
>                     That one may be a little harder to find? :)
>
>                     On Tue, Jul 5, 2011 at 10:44 AM, Steven Ayre
>                     <steveayre at gmail.com <mailto:steveayre at gmail.com>>
>                     wrote:
>
>                         I'm assuming it's this document:
>                         ftp://ftp.veracomp.com.pl/net/nomadix/Nomadix%20-%20PMS%20info/FIAS150.pdf
>                         Quite easy to google once I had the version
>                         number.
>
>                         You may find the nicest approach is to write a
>                         FOSS libfias, then write an endpoint module to
>                         tie FS and libfias together. Plenty of
>                         existing endpoint modules (mod_sofia
>                         mod_skinny mod_opal mod_h323 etc) can show you
>                         examples to get you started. Don't forget to
>                         read the FS API documentation too:
>                         http://docs.freeswitch.org/
>
>                         I'm assuming there are no license/patent
>                         restrictions to using FIAS?
>
>                         Good luck!
>
>                         -Steve
>
>
>
>
>                         On 5 July 2011 17:30, Luis F Urrea
>                         <lfurrea at gmail.com <mailto:lfurrea at gmail.com>>
>                         wrote:
>
>                             Hello Nandy,
>
>                             A couple of months ago I started some
>                             research on the subject and concluded I
>                             had to write my own interface to FS,
>                             however I haven't had the time to get the
>                             project off ground yet.
>
>                             I do have a copy of FIAS specification
>                             version 1.5 from 2001 which is publicly
>                             available I am sure it's not the latest
>                             but it should cover the basics.
>
>                             Please contact me off list if you have a
>                             hard time getting it online.
>
>                             Regards
>
>                             On Mon, Jul 4, 2011 at 4:07 PM, Nandy
>                             Dagondon <gcd at i.ph <mailto:gcd at i.ph>> wrote:
>
>                                 hi everybody,
>
>                                 anyone working on interfacing FS with
>                                 Fidelio Hotel PMS?  i can't find the
>                                 FIAS protocol/specs online.  is this
>                                 freely available?
>
>                                 tks,
>                                 nandy
>
>
>                                 _______________________________________________
>                                 Join us at ClueCon 2011, Aug 9-11, Chicago
>                                 http://www.cluecon.com 877-7-4ACLUE
>
>                                 FreeSWITCH-users mailing list
>                                 FreeSWITCH-users at lists.freeswitch.org
>                                 <mailto:FreeSWITCH-users at lists.freeswitch.org>
>                                 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>                                 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>                                 http://www.freeswitch.org
>
>
>
>                             _______________________________________________
>                             Join us at ClueCon 2011, Aug 9-11, Chicago
>                             http://www.cluecon.com 877-7-4ACLUE
>
>                             FreeSWITCH-users mailing list
>                             FreeSWITCH-users at lists.freeswitch.org
>                             <mailto:FreeSWITCH-users at lists.freeswitch.org>
>                             http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>                             UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>                             http://www.freeswitch.org
>
>
>
>                         _______________________________________________
>                         Join us at ClueCon 2011, Aug 9-11, Chicago
>                         http://www.cluecon.com 877-7-4ACLUE
>
>                         FreeSWITCH-users mailing list
>                         FreeSWITCH-users at lists.freeswitch.org
>                         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>                         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>                         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>                         http://www.freeswitch.org
>
>
>
>                     _______________________________________________
>                     Join us at ClueCon 2011, Aug 9-11, Chicago
>                     http://www.cluecon.com 877-7-4ACLUE
>
>                     FreeSWITCH-users mailing list
>                     FreeSWITCH-users at lists.freeswitch.org
>                     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>                     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>                     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>                     http://www.freeswitch.org
>
>
>
>                 _______________________________________________
>                 Join us at ClueCon 2011, Aug 9-11, Chicago
>                 http://www.cluecon.com 877-7-4ACLUE
>
>                 FreeSWITCH-users mailing list
>                 FreeSWITCH-users at lists.freeswitch.org
>                 <mailto:FreeSWITCH-users at lists.freeswitch.org>
>                 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>                 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>                 http://www.freeswitch.org
>
>
>
>             _______________________________________________
>             Join us at ClueCon 2011, Aug 9-11, Chicago
>             http://www.cluecon.com 877-7-4ACLUE
>
>             FreeSWITCH-users mailing list
>             FreeSWITCH-users at lists.freeswitch.org
>             <mailto:FreeSWITCH-users at lists.freeswitch.org>
>             http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>             UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>             http://www.freeswitch.org
>
>
>
>         _______________________________________________
>         Join us at ClueCon 2011, Aug 9-11, Chicago
>         http://www.cluecon.com 877-7-4ACLUE
>
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>
>
>
>     _______________________________________________
>     Join us at ClueCon 2011, Aug 9-11, Chicago
>     http://www.cluecon.com 877-7-4ACLUE
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com <http://www.avg.com>
> Version: 10.0.1390 / Virus Database: 1518/3802 - Release Date: 08/01/11
>


-- 
Atentamente,
*Dario Garc?a*
Consultor.

CCCT, Nivel C2, Sector Yarey, Mz,
Ofc. MZ03a.
Caracas-Venezuela.
Tel?fono: +58 212 9081842
Cel: +58 412 2221515
dgarcia at anew.com.ve
http://www.anew.com.ve
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/49d835a9/attachment-0001.html 

From jmesquita at freeswitch.org  Tue Aug  2 19:32:15 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Tue, 2 Aug 2011 12:32:15 -0300
Subject: [Freeswitch-users] Fidelio
In-Reply-To: <4E37FED3.7080306@anew.com.ve>
References: <CAHKqMb5dnE1mgdFaV65R=tk-zxvfz30=hi+3AyxezENQXVsrYw@mail.gmail.com>
	<CAB2KLyfDvLjmqUU2coOJzG6ss6xzKvsGD5OOsS+bVwKaXx+P0g@mail.gmail.com>
	<CAFiqYukzYyKZ_Tj8w8PFrM4FFpXzZiguRYNfqP8H=ah0iZLkhA@mail.gmail.com>
	<CAB2KLycjzFtLc8jTBM9377CDi488K9nDmVD=Zz-13MZ7_w-cxA@mail.gmail.com>
	<CABH8QX2y1mu1L=9-KZk2HH-Xh9MFOS98p4ctvvQrF7AW_RJV_A@mail.gmail.com>
	<CAHKqMb5uAypVLHgqkt+GFNC_oj5wDPnFbzKHxjwr0QhcEhYJPw@mail.gmail.com>
	<CABH8QX1HHvqL_YK0rbs-m1Ky1GtbZ6OQKyKMKkAne8XV5WBSig@mail.gmail.com>
	<CAHKqMb51RbLBo+pmKwhJ82=ZWaGcyM6bd6y881Vh=4cAjztt5A@mail.gmail.com>
	<CABH8QX0cgqbPcCtS=2Z0ron_vqX=5-YMrnkGJgqsOYXx81+x9Q@mail.gmail.com>
	<4E37FED3.7080306@anew.com.ve>
Message-ID: <CABH8QX3+gukdj_wLvefbKd_YVEq2Rv2Y8FcDpQ+Xc_cGD+7UWA@mail.gmail.com>

Dear Dario,

Ante todo, un saludo de uno m?s que habla espa?ol!

I have worked with an Argentinian company called BitSense for them to
develop their hospitality solution called Coral. They have CSTA integrated
with Siemens amongst other things and their base is in fact FreeSWITCH. I am
unsure of how many hotels they managed to install and their integration with
Fidelio. I have worked in the past for a brazilian company that has a
similar product (I think they are the current Siemens solution) and I do
know ppl inside of Micros, hence my last email.

Yo conozco a ANEW por otros motivos ajenos a FS y de hecho cuando recibi tu
mail me puse en contacto con Jhonny Lagos, pero me enter? que ya no esta mas
en ANEW. Est?n buscando una soluci?n hotelera? Te puedo presentar a esta
gente, son clientes mios. Ser? un gusto ayudarlos en lo que pueda.

Saludos,
Jo?o Mesquita



2011/8/2 Saugort Dario Garcia Tovar <dgarcia at anew.com.ve>

>  Hello all
> If you have money to afford the certification process go ahead, at the end
> you will get back the money if you see that you can put a hotel solution
> based on FS in your local market.
>
> I am new in FS world. I barely
>
> I used to work with Alcatel-Lucent OXE (IP-PBX). First the OXE, by itself,
> offer hotel funcionality: voicemail for guest, guest room admin (ready/free,
> cleaning, occupied,etc), alarm clock (at the hour programmed by the guest
> from their phone or for "hotel phone console" or admin interfase; the phone
> ring), fax service, room billing (quite different fron a normal cdr), etc,
> etc, etc. Second, OXE offer a propetary protocol to make possible PMS like
> fidelio to interface with, that means, Fidelio become the administrative
> interface, so from fidelio control room status, services, etc.
>
> So the first, do you have to adapt FS to offer a minimum of hotel
> funcionality. Second about make a PMS-FS integration could be two options or
> more: 1) adapt FS to PMS (using the integration doc); 2) many pbx vendors
> offer info about their hotel link protocol for free (not Alcatel), and are
> supported by fidelio like, OXE does, so you could develop a interface that
> emulate an already supported hotel link towards Fidelio and FS.
>
> I have a question:  has someone work with FS to offer an hotel solution?
>
>
>
>
> On 8/1/2011 9:19 PM, Jo?o Mesquita wrote:
>
> Like I said, if you can raise enough money to be able to get the
> certification going, I can talk to the ppl I know on Micros and see if we
> can get FS certified (no guarantees), but it will cost at least 5k USD, so I
> really don't know if there is enough interest.
>
> Regards,
> Jo?o Mesquita
>
>
>
> On Mon, Aug 1, 2011 at 10:36 PM, Nandy Dagondon <gcd at i.ph> wrote:
>
>> yes, certification is big obstacle. too much for our intended project.
>>  perhaps FS can apply for certification as a platform not on a
>> per-company/model basis. i just hope Fidelio would be open to the idea.
>>
>>  another option would be to ask Asterisk-FIAS connectors like PBillX the
>> possibility to include FreeSwitch.
>>
>> 2011/8/2 Jo?o Mesquita <jmesquita at freeswitch.org>
>>
>>> I think there are none doing this at this moment. Here's what I know tho:
>>>
>>> 1. I haven't seen the contracts or legal papers involved in the
>>> certification
>>> 2. I've never done a connector even tho I worked with companies that did
>>> in the past
>>> 3. I know that the homologation/certification costs money and not little
>>> money. As far as I know, there are differences between countries when it
>>> comes to the fee to be paid. Here is Argentina it is something close to
>>> USD$5.000,00.
>>> 4. For you to be officially certified by them, you need to have your
>>> system installed on at least 3 hotels with Fidelio, otherwise you just lost
>>> money.
>>>
>>> That should pretty much give you a hint of what you are facing when it
>>> comes to the certification process. Does your project accomodate these
>>> imposed barriers? If they do, I might be able to help.
>>>
>>> Regards,
>>> Jo?o Mesquita
>>>
>>>
>>>
>>>
>>> On Mon, Aug 1, 2011 at 9:38 PM, Nandy Dagondon <gcd at i.ph> wrote:
>>>
>>>> hi Jo?o,
>>>>
>>>>  since you had FIAS experience, would a commercial  mod_fias
>>>> license/certification be possible just like g.729?
>>>>
>>>>  re companies selling FIAS connectors, i can't find one that connects
>>>> FS to Fidelio. otherwise, they would sound off in this thread.
>>>>
>>>>  -nandy
>>>>
>>>>  2011/7/6 Jo?o Mesquita <jmesquita at freeswitch.org>
>>>>
>>>>> Guys, be careful because I think this document as well as the protocol
>>>>> are confidential. I had to sign an NDA with Fidelio to get my hands on it
>>>>> and pay a fee for it as well. You might as well confirm it since you all
>>>>> seem to be in the US where this type of information might be easier to get.
>>>>>
>>>>>  There are LOTS of companies selling their connectors to Fidelio...
>>>>>
>>>>>  One other point is that you need to have the certification with them
>>>>> to be considered compatible, otherwise, no consultant will install the
>>>>> connector on the fidelio side.
>>>>>
>>>>>  Regards,
>>>>> Jo?o Mesquita
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Tue, Jul 5, 2011 at 1:52 PM, Luis F Urrea <lfurrea at gmail.com>wrote:
>>>>>
>>>>>> Awesome! great suggestions to get started,
>>>>>>
>>>>>>  There is also a FIAS simulator floating around.
>>>>>>
>>>>>>  That one may be a little harder to find? :)
>>>>>>
>>>>>> On Tue, Jul 5, 2011 at 10:44 AM, Steven Ayre <steveayre at gmail.com>wrote:
>>>>>>
>>>>>>> I'm assuming it's this document:
>>>>>>>
>>>>>>> ftp://ftp.veracomp.com.pl/net/nomadix/Nomadix%20-%20PMS%20info/FIAS150.pdf
>>>>>>> Quite easy to google once I had the version number.
>>>>>>>
>>>>>>> You may find the nicest approach is to write a FOSS libfias, then
>>>>>>> write an endpoint module to tie FS and libfias together. Plenty of
>>>>>>> existing endpoint modules (mod_sofia mod_skinny mod_opal mod_h323 etc) can show
>>>>>>> you examples to get you started. Don't forget to read the FS API
>>>>>>> documentation too: http://docs.freeswitch.org/
>>>>>>>
>>>>>>> I'm assuming there are no license/patent restrictions to using FIAS?
>>>>>>>
>>>>>>> Good luck!
>>>>>>>
>>>>>>> -Steve
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On 5 July 2011 17:30, Luis F Urrea <lfurrea at gmail.com> wrote:
>>>>>>>
>>>>>>>> Hello Nandy,
>>>>>>>>
>>>>>>>>  A couple of months ago I started some research on the subject and
>>>>>>>> concluded I had to write my own interface to FS, however I haven't had the
>>>>>>>> time to get the project off ground yet.
>>>>>>>>
>>>>>>>>  I do have a copy of FIAS specification version 1.5 from 2001 which
>>>>>>>> is publicly available I am sure it's not the latest but it should cover the
>>>>>>>> basics.
>>>>>>>>
>>>>>>>>  Please contact me off list if you have a hard time getting it
>>>>>>>> online.
>>>>>>>>
>>>>>>>>  Regards
>>>>>>>>
>>>>>>>>  On Mon, Jul 4, 2011 at 4:07 PM, Nandy Dagondon <gcd at i.ph> wrote:
>>>>>>>>
>>>>>>>>>  hi everybody,
>>>>>>>>>
>>>>>>>>>  anyone working on interfacing FS with Fidelio Hotel PMS?  i can't
>>>>>>>>> find the FIAS protocol/specs online.  is this freely available?
>>>>>>>>>
>>>>>>>>>  tks,
>>>>>>>>> nandy
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>  _______________________________________________
>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicagohttp://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 10.0.1390 / Virus Database: 1518/3802 - Release Date: 08/01/11
>
>
>
> --
> Atentamente,
> *Dario Garc?a*
> Consultor.
>
> CCCT, Nivel C2, Sector Yarey, Mz,
> Ofc. MZ03a.
> Caracas-Venezuela.
> Tel?fono: +58 212 9081842
> Cel: +58 412 2221515
> dgarcia at anew.com.ve
> http://www.anew.com.ve
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/1858b96c/attachment-0001.html 

From jmoran at secureachsystems.com  Tue Aug  2 19:57:03 2011
From: jmoran at secureachsystems.com (Jason Moran)
Date: Tue, 2 Aug 2011 11:57:03 -0400
Subject: [Freeswitch-users] TTS/unimrcp engines
Message-ID: <361E98F99D3CC3439EED59BC1924ED69508754@SERVER2003.SecuReachSystems.local>

I'm trying to wrap up which TTS engine we will use over unimrcp. We'll
use between 10 and 40 simultaneous TTS lines.

So far I have tested the new IVONA mrcpv1 plugin but I am waiting on
their dev's to work out a few issues I uncovered.

 

Which TTS engines have been successfully used by others?

 

I've heard good things about Loquendo, but I have had problems getting
in contact with them.

Acapela sounds nice on their website - does anybody know how their
unimrcp plugin sounds?

Cepstral is cheap, but it seems as though they do not have a current
release for Open Suse 11.6 (32bit). 

 

Any others? Recommendations?

-Jason

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/196b4a0a/attachment.html 

From vermeulen.deon at gmail.com  Tue Aug  2 11:22:14 2011
From: vermeulen.deon at gmail.com (Deon Vermeulen)
Date: Tue, 2 Aug 2011 08:22:14 +0100
Subject: [Freeswitch-users] Audio issues when FS as PSTN/SBC Gateway for
	OpenSIPS
Message-ID: <94C7CF0B-F433-40B3-A911-85D4BD0809D3@gmail.com>

Hi List


My Current setup is as follows.

PSTN -- Sangoma A500 bri -- FS -- OpenSIPS

DID: bridges called number to ext at opensips

My Phones are located on our office private LAN, while FS and OpenSIPS are both on Public Network with Public IPs.
There is NAT between the IP Phones and OpenSIPS/FS.
I am using non standard SIP ports as well as not defined any rules for RTP on the firewall.
I have a many to one NAT scenario on the Firewall.


OpenSIPS is setup to not Proxy any media between the phones and FS.
Media flows directly between FS and phones.

When I make calls to PSTN, audio is Crystal clear on both sides of the call.

When I receive calls from the PSTN I sometimes have audio, sometimes none and most of the time one way (to PSTN).

I have done quite a lot of sip trace debugging in fs_cli and I really can't see any problems in the traces.
I have also enabled inbound-proxy-media in all sip profiles, just to make sure I don't miss it somewhere in the call flow.


I have been digging around in the wiki, forums as well as some archived mails wrt FS and NAT, but I don't 100% understand how FS handles NAT by default in the External.xml profile.
I'm also not 100% sure how FS will handle this type of scenario as no Phones are registered to it and it also has to pass SIP and SDP info with OpenSIPS and not directly with the Phones.


Any advise and or guidance will really be appreciated.


Thank you very much.

Kind Regards
Deon Vermeulen

From Stefan.Weigel at allianz-warranty.com  Tue Aug  2 11:43:11 2011
From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan)
Date: Tue, 2 Aug 2011 09:43:11 +0200
Subject: [Freeswitch-users] Various question - CallerID / Call transfer via
	REFER and CDR
Message-ID: <5003D7D3E06F514E8C682F18D223265C04D3B36D1E@AZWSMS03.azwarranty.int>

Hi all,



I'm currently stuck with some issues I can't figure out.

I'm using FreeSWITCH Version 1.0.head (git-db5f504 2011-07-17 17-00-38 -0400), phone devices are Polycom Soundpoint IP 560.



1.       When receiving a call I change some of the incoming Caller-ID names. It's working like a charm.

But if I do a call forward (it's a 'Consultative transfer'-Party A dials party C's number and talks

privately with party C after the call is answered, and then completes the transfer or hangs up) I only

see the party A's number/name on the display C. Is there a possibility to get the original caller-ID/-name displayed on the phone of Party C ?



2.       When doing a 'Consultative transfer', I only get one CDR entry using mod_cdr_csv.

For example calling from my cellphone 0179 12345678 to extension 615, then doing a call transfer from 615 to 803 I get

an entry in CDR with calling_id = 615, destination_number = 803.

Is it possible to trigger the creation of a CDR entry ? I tried using process_cdr but only one entry get's logged.

In cdr_csv.conf I'm using logging of a-leg, for billing / call statistic.





Thanks in advance and best regards



Stefan

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/bf5fced5/attachment-0001.html 

From Stefan.Weigel at allianz-warranty.com  Tue Aug  2 16:19:58 2011
From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan)
Date: Tue, 2 Aug 2011 14:19:58 +0200
Subject: [Freeswitch-users] Various question - CallerID / Call transfer via
	REFER and CDR
Message-ID: <5003D7D3E06F514E8C682F18D223265C04D3B36D22@AZWSMS03.azwarranty.int>

Hi all,



I'm currently stuck with some issues I can't figure out.

I'm using FreeSWITCH Version 1.0.head (git-db5f504 2011-07-17 17-00-38 -0400), phone devices are Polycom Soundpoint IP 560.



1.       When receiving a call I change some of the incoming Caller-ID names. It's working like a charm.

But if I do a call forward (it's a 'Consultative transfer'-Party A dials party C's number and talks

privately with party C after the call is answered, and then completes the transfer or hangs up) I only

see the party A's number/name on the display C. Is there a possibility to get the original caller-ID/-name displayed on the phone of Party C ?



2.       When doing a 'Consultative transfer', I only get one CDR entry using mod_cdr_csv.

For example calling from my cellphone 0179 12345678 to extension 615, then doing a call transfer from 615 to 803 I get

an entry in CDR with calling_id = 615, destination_number = 803.

Is it possible to trigger the creation of a CDR entry ? I tried using process_cdr but only one entry get's logged.

In cdr_csv.conf I'm using logging of a-leg, for billing / call statistic.





Thanks in advance and best regards



Stefan



-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/710f1a31/attachment-0001.html 

From rhuddleston at gmail.com  Tue Aug  2 17:41:56 2011
From: rhuddleston at gmail.com (Robert Huddleston)
Date: Tue, 2 Aug 2011 09:41:56 -0400
Subject: [Freeswitch-users] Limit number of voicemails
In-Reply-To: <4E37CA90.4090806@amooma.de>
References: <4E37CA90.4090806@amooma.de>
Message-ID: <05a801cc5119$f3142e00$d93c8a00$@com>

Guten Tag

Have you considered writing your own patch for this?

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sascha
Daniels
Sent: Tuesday, August 02, 2011 6:00 AM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Limit number of voicemails

Hi together.

For a small appliance I need to limit the number of voicemails for each
user.

I know that I can limit the length. That is the first step.

Is there a way to disable the message recording, when the maximum number
of voicemails is reached?

Unfortunately I didn't find anything in the documentation.

Kind regards

                        Sascha

-- 
AMOOMA GmbH - Bachstr. 124 - 56566 Neuwied  -->  http://www.amooma.de
Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister Montabaur B14998

B?cher: http://das-asterisk-buch.de - http://ruby-auf-schienen.de


_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

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



From rhuddleston at gmail.com  Tue Aug  2 17:45:38 2011
From: rhuddleston at gmail.com (Robert Huddleston)
Date: Tue, 2 Aug 2011 09:45:38 -0400
Subject: [Freeswitch-users] Limit number of voicemails
In-Reply-To: <4E37CA90.4090806@amooma.de>
References: <4E37CA90.4090806@amooma.de>
Message-ID: <05a901cc511a$775ee8d0$661cba70$@com>

Actually http://wiki.freeswitch.org/wiki/Mod_voicemail#vm-disk-quota

In the manual...

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sascha
Daniels
Sent: Tuesday, August 02, 2011 6:00 AM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Limit number of voicemails

Hi together.

For a small appliance I need to limit the number of voicemails for each
user.

I know that I can limit the length. That is the first step.

Is there a way to disable the message recording, when the maximum number
of voicemails is reached?

Unfortunately I didn't find anything in the documentation.

Kind regards

                        Sascha

-- 
AMOOMA GmbH - Bachstr. 124 - 56566 Neuwied  -->  http://www.amooma.de
Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister Montabaur B14998

B?cher: http://das-asterisk-buch.de - http://ruby-auf-schienen.de


_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

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



From rhuddleston at gmail.com  Tue Aug  2 19:50:47 2011
From: rhuddleston at gmail.com (Robert Huddleston)
Date: Tue, 2 Aug 2011 11:50:47 -0400
Subject: [Freeswitch-users] Fidelio
In-Reply-To: <CABH8QX3+gukdj_wLvefbKd_YVEq2Rv2Y8FcDpQ+Xc_cGD+7UWA@mail.gmail.com>
References: <CAHKqMb5dnE1mgdFaV65R=tk-zxvfz30=hi+3AyxezENQXVsrYw@mail.gmail.com>
	<CAB2KLyfDvLjmqUU2coOJzG6ss6xzKvsGD5OOsS+bVwKaXx+P0g@mail.gmail.com>
	<CAFiqYukzYyKZ_Tj8w8PFrM4FFpXzZiguRYNfqP8H=ah0iZLkhA@mail.gmail.com>
	<CAB2KLycjzFtLc8jTBM9377CDi488K9nDmVD=Zz-13MZ7_w-cxA@mail.gmail.com>
	<CABH8QX2y1mu1L=9-KZk2HH-Xh9MFOS98p4ctvvQrF7AW_RJV_A@mail.gmail.com>
	<CAHKqMb5uAypVLHgqkt+GFNC_oj5wDPnFbzKHxjwr0QhcEhYJPw@mail.gmail.com>
	<CABH8QX1HHvqL_YK0rbs-m1Ky1GtbZ6OQKyKMKkAne8XV5WBSig@mail.gmail.com>
	<CAHKqMb51RbLBo+pmKwhJ82=ZWaGcyM6bd6y881Vh=4cAjztt5A@mail.gmail.com>
	<CABH8QX0cgqbPcCtS=2Z0ron_vqX=5-YMrnkGJgqsOYXx81+x9Q@mail.gmail.com>
	<4E37FED3.7080306@anew.com.ve>
	<CABH8QX3+gukdj_wLvefbKd_YVEq2Rv2Y8FcDpQ+Xc_cGD+7UWA@mail.gmail.com>
Message-ID: <062201cc512b$f308b720$d91a2560$@com>

Maybe we should start a new mailing list ? as this Fidelio conversation is
very small audienced

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jo?o
Mesquita
Sent: Tuesday, August 02, 2011 11:32 AM
To: Saugort Dario Garcia Tovar
Cc: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Fidelio

 

Dear Dario,

Ante todo, un saludo de uno m?s que habla espa?ol!

I have worked with an Argentinian company called BitSense for them to
develop their hospitality solution called Coral. They have CSTA integrated
with Siemens amongst other things and their base is in fact FreeSWITCH. I am
unsure of how many hotels they managed to install and their integration with
Fidelio. I have worked in the past for a brazilian company that has a
similar product (I think they are the current Siemens solution) and I do
know ppl inside of Micros, hence my last email.

Yo conozco a ANEW por otros motivos ajenos a FS y de hecho cuando recibi tu
mail me puse en contacto con Jhonny Lagos, pero me enter? que ya no esta mas
en ANEW. Est?n buscando una soluci?n hotelera? Te puedo presentar a esta
gente, son clientes mios. Ser? un gusto ayudarlos en lo que pueda.

Saludos,
Jo?o Mesquita




2011/8/2 Saugort Dario Garcia Tovar <dgarcia at anew.com.ve>

Hello all
If you have money to afford the certification process go ahead, at the end
you will get back the money if you see that you can put a hotel solution
based on FS in your local market.

I am new in FS world. I barely

I used to work with Alcatel-Lucent OXE (IP-PBX). First the OXE, by itself,
offer hotel funcionality: voicemail for guest, guest room admin (ready/free,
cleaning, occupied,etc), alarm clock (at the hour programmed by the guest
from their phone or for "hotel phone console" or admin interfase; the phone
ring), fax service, room billing (quite different fron a normal cdr), etc,
etc, etc. Second, OXE offer a propetary protocol to make possible PMS like
fidelio to interface with, that means, Fidelio become the administrative
interface, so from fidelio control room status, services, etc.

So the first, do you have to adapt FS to offer a minimum of hotel
funcionality. Second about make a PMS-FS integration could be two options or
more: 1) adapt FS to PMS (using the integration doc); 2) many pbx vendors
offer info about their hotel link protocol for free (not Alcatel), and are
supported by fidelio like, OXE does, so you could develop a interface that
emulate an already supported hotel link towards Fidelio and FS.

I have a question:  has someone work with FS to offer an hotel solution?





On 8/1/2011 9:19 PM, Jo?o Mesquita wrote: 

Like I said, if you can raise enough money to be able to get the
certification going, I can talk to the ppl I know on Micros and see if we
can get FS certified (no guarantees), but it will cost at least 5k USD, so I
really don't know if there is enough interest.

Regards,
Jo?o Mesquita




On Mon, Aug 1, 2011 at 10:36 PM, Nandy Dagondon <gcd at i.ph> wrote:

yes, certification is big obstacle. too much for our intended project.
perhaps FS can apply for certification as a platform not on a
per-company/model basis. i just hope Fidelio would be open to the idea.  

 

another option would be to ask Asterisk-FIAS connectors like PBillX the
possibility to include FreeSwitch.

 

2011/8/2 Jo?o Mesquita <jmesquita at freeswitch.org>

I think there are none doing this at this moment. Here's what I know tho:

1. I haven't seen the contracts or legal papers involved in the
certification
2. I've never done a connector even tho I worked with companies that did in
the past
3. I know that the homologation/certification costs money and not little
money. As far as I know, there are differences between countries when it
comes to the fee to be paid. Here is Argentina it is something close to
USD$5.000,00.
4. For you to be officially certified by them, you need to have your system
installed on at least 3 hotels with Fidelio, otherwise you just lost money.

That should pretty much give you a hint of what you are facing when it comes
to the certification process. Does your project accomodate these imposed
barriers? If they do, I might be able to help.

Regards,
Jo?o Mesquita 






On Mon, Aug 1, 2011 at 9:38 PM, Nandy Dagondon <gcd at i.ph> wrote:

hi Jo?o, 

 

since you had FIAS experience, would a commercial  mod_fias
license/certification be possible just like g.729?  

 

re companies selling FIAS connectors, i can't find one that connects FS to
Fidelio. otherwise, they would sound off in this thread.

-nandy

2011/7/6 Jo?o Mesquita <jmesquita at freeswitch.org>

Guys, be careful because I think this document as well as the protocol are
confidential. I had to sign an NDA with Fidelio to get my hands on it and
pay a fee for it as well. You might as well confirm it since you all seem to
be in the US where this type of information might be easier to get. 

 

There are LOTS of companies selling their connectors to Fidelio...

 

One other point is that you need to have the certification with them to be
considered compatible, otherwise, no consultant will install the connector
on the fidelio side. 


Regards,

Jo?o Mesquita 






On Tue, Jul 5, 2011 at 1:52 PM, Luis F Urrea <lfurrea at gmail.com> wrote:

Awesome! great suggestions to get started, 

 

There is also a FIAS simulator floating around.

 

That one may be a little harder to find? :)

 

On Tue, Jul 5, 2011 at 10:44 AM, Steven Ayre <steveayre at gmail.com> wrote:

I'm assuming it's this document:
ftp://ftp.veracomp.com.pl/net/nomadix/Nomadix%20-%20PMS%20info/FIAS150.pdf
Quite easy to google once I had the version number.

You may find the nicest approach is to write a FOSS libfias, then write an
endpoint module to tie FS and libfias together. Plenty of existing endpoint
modules (mod_sofia mod_skinny mod_opal mod_h323 etc) can show you examples
to get you started. Don't forget to read the FS API documentation too:
http://docs.freeswitch.org/

I'm assuming there are no license/patent restrictions to using FIAS?

Good luck!

-Steve 






On 5 July 2011 17:30, Luis F Urrea <lfurrea at gmail.com> wrote:

Hello Nandy, 

 

A couple of months ago I started some research on the subject and concluded
I had to write my own interface to FS, however I haven't had the time to get
the project off ground yet.

 

I do have a copy of FIAS specification version 1.5 from 2001 which is
publicly available I am sure it's not the latest but it should cover the
basics.

 

Please contact me off list if you have a hard time getting it online.

 

Regards

On Mon, Jul 4, 2011 at 4:07 PM, Nandy Dagondon <gcd at i.ph> wrote:

hi everybody,


 

anyone working on interfacing FS with Fidelio Hotel PMS?  i can't find the
FIAS protocol/specs online.  is this freely available?

 

tks,

nandy

 

_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

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

 


_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

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

 


_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

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

 


_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

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

 


_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

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

 


_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

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

 


_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

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

 


_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

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





_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE
 
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

No virus found in this message.
Checked by AVG - www.avg.com
Version: 10.0.1390 / Virus Database: 1518/3802 - Release Date: 08/01/11

 

-- 
Atentamente, 
Dario Garc?a 
Consultor. 

CCCT, Nivel C2, Sector Yarey, Mz, 
Ofc. MZ03a. 
Caracas-Venezuela. 
Tel?fono: +58 212 9081842 
Cel: +58 412 2221515 
dgarcia at anew.com.ve 
http://www.anew.com.ve

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/7e71318c/attachment-0001.html 

From freeswitch at aastral.net  Tue Aug  2 20:20:36 2011
From: freeswitch at aastral.net (Bill W.)
Date: Tue, 02 Aug 2011 12:20:36 -0400
Subject: [Freeswitch-users] ESL: not bein able to determine when SIP
 gateway is down when originating a call
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C59EBABB87C@cooper>
References: <BANLkTik9R9xqEEQic9miZbvN7T8MWCF-JA@mail.gmail.com>
	<1Qo6kg-0001Wa-LC@mail.aastral.net>,
	<CAKbxfG_j7fRYa3yCvQAcOk=tKDr1ynWOVZqoi2geumDw80qZ0Q@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59EBABB87C@cooper>
Message-ID: <1QoHhx-0002y0-IY@mail.aastral.net>

Thanks for your input guys.

Since I'm not a programmer, patching sofia is out of the question.  I'd
do more harm than good.

I haven't had a chance yet to look at all the events after I originate
to a dead gateway to find an elegant solution.   My hack is to look for
a CHANNEL_CALLSTATE event with Channel-Call-State = RINGING.   If that
doesn't occur, then I assume the call has failed.

I did see INVALID_NUMBER_FORMAT in the cli but I didn't notice it in the
events.  I'll grep for it though to make sure.

Thanks,
Bill



On 8/2/11 2:02 AM, Peter Olsson wrote:
> In this case there should at least be a BACKGROUND_JOB event with the result from originate (INVALID_NUMBER_FORMAT) - are you monitoring these events? If you want a specific event fired for "gateway down" you will have to create a patch for this.
> 
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Anthony Minessale [anthony.minessale at gmail.com]
> Skickat: den 2 augusti 2011 06:59
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] ESL: not bein able to determine when SIP gateway is down when originating a call
> 
> make a patch to add a new custom sofia event and fire it
> 
> 
> On Mon, Aug 1, 2011 at 11:38 PM, Bill W. <freeswitch at aastral.net> wrote:
>> Hey Anton,
>>
>> I'm running in to this same issue.  Did you ever find a result?
>>
>> Thanks,
>> Bill
>>
>> On 5/14/11 1:17 PM, Anton VG wrote:
>>> I'm trying to catch an error, in case I would dial wrong (non
>>> existent) gateway (intentionally!)
>>>
>>> I'm running ESL outbound listener, subscribing to all events,
>>> if I do bgapi 'originate' to a live gateway - there are normal events
>>> flow, and I
>>> can track what is happening.
>>> But if I issue originate to a gateway, which is not configured or
>>> simply down - there
>>> are no any events fired.
>>>
>>> I only have an error on FS console
>>>
>>> 2011-05-14 20:58:13.072927 [ERR] mod_sofia.c:4044 Invalid Gateway
>>> following by
>>> 2011-05-14 20:58:13.072927 [ERR] switch_ivr_originate.c:2447 Cannot
>>> create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]
>>>
>>> But HOW to catch the given in ESL?
>>>
>>> sofia.c seems just does not have event code for that cases
>>>
>>> if (profile_name && !profile_found) {
>>> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "No Such
>>> Profile '%s'\n", profile_name);
>>>> status = SWITCH_STATUS_FALSE;
>>> }
>>>
>>> logically there should be a proper way to determine that gateway is bad in my
>>> ESL dialplan, by catching the proper event/reply/whatever,
>>> For the moment i did trick: esl.api('sofia status gateway
>>> GatewayWhichIsDown')
>>>
>>> When in production, and there is more than a single route, there will
>>> be plenty of cases, when you dial a bad gateway, so there should be a
>>> way for ESL dialplan to determine that a gateway is not callable for a
>>> moment, the reason WHY and  to retry with another one.
>>>
>>> The trick above is bad, since:
>>> 1. blocking api query, before evey single gateway call attempt.
>>> 2. Gateway maybe known in UP state, but the state is stale, in dial in
>>> fact will go to DOWN gateway. So, ESL dialplan will screw in that case
>>>
>>> Any clue?
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> 
> 
> 
> --
> Anthony Minessale II
> 
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> !DSPAM:4e3784ca32763825510562!
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From vkozak at abisoft.spb.ru  Tue Aug  2 20:26:52 2011
From: vkozak at abisoft.spb.ru (Kozak Vladimir)
Date: Tue, 2 Aug 2011 20:26:52 +0400
Subject: [Freeswitch-users] EXCHANGE_ROUTING_ERROR after ~70 transfer to
	dialplan extebsion
Message-ID: <576190BDB2B847A180F79DF6B311A49C@abisoft.biz>

Hi all.
FreeSWITCH Version 1.0.head (git-9ff8f53 2011-05-03 12-13-52 -0400)

I have problem with transfer channel to FS extension.
I send from external system to FS command as
   "SendMsg " + uniqueId + "\n" +
   "call-command: execute\n" +
   "execute-app-name: transfer\n" +
   "execute-app-arg: " + destination + "\n\n";

after ~70 correct transfer actions FS send BYE to my phone and send HANGUP_EVENT with cause EXCHANGE_ROUTING_ERROR to my system. (FS log attached)

from FS loggs:
2011-08-09 04:48:38.525293 [DEBUG] mod_sofia.c:457 Channel sofia/internal/1000 at vkozak.starpoundtech.net hanging up, cause: EXCHANGE_ROUTING_ERROR
2011-08-09 04:48:38.525293 [DEBUG] mod_sofia.c:500 Sending BYE to sofia/internal/1000 at vkozak.starpoundtech.net

what for does limitation of transfer operations exist?
how can I avoid this issue?
is't possible to increase number of hops for transfer command?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/65dfe969/attachment-0001.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: FS.log
Type: application/octet-stream
Size: 56084 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/65dfe969/attachment-0001.obj 

From mays.david at gmail.com  Tue Aug  2 20:32:18 2011
From: mays.david at gmail.com (David Ma)
Date: Wed, 3 Aug 2011 00:32:18 +0800
Subject: [Freeswitch-users] How to use ESL event
Message-ID: <CAMLBmRJMsShGoDw0ZHKRmVi8_uza_GxDviqr05KrTW7BGztL+A@mail.gmail.com>

Hello All,

I have some problem with receiving and handling ESL events.

My code is like:

void handle_eslswitch_event_plain(esl_handle_t *handle, esl_event_t *event)
{
        ... ...
        TRACE_DEBUG("ESL event [%d] %s: UUID: %s",
                event->event_id, fs_get_event_header(event, "Event-Name"),
uniqueid);
         print_event(event);
        ... ...
}

void handle_event(esl_handle_t *handle, esl_event_t *last_event)
{
        ... ...
        if (!strcasecmp(type, "text/event-plain")) {
            handle_eslswitch_event_plain(handle, handle->last_ievent);
        }
        ... ...
}

int main()
{
        ... ...
        //receive from event socket
        if (handle.last_event) handle_event(&handle, handle.last_event);
        ... ...

Usually I receive correct events. But I occasionally receive incorrect
event. See below the 1st is the normal event and the 2nd is the wrong event:

<2011-07-29 14:25:29> [DEBUG] ESL event [8] CHANNEL_HANGUP_COMPLETE: UUID:
f320e4a6-db23-46d0-8d89-9957eccbd4c9
<2011-07-29 14:25:29> [DEBUG] RECV EVENT
Event-Name: CHANNEL_HANGUP_COMPLETE
Core-UUID: 26b77cba-8fdd-486d-90ec-6844bca58c72
FreeSWITCH-Hostname: fs01
FreeSWITCH-IPv4: 10.1.1.46
FreeSWITCH-IPv6: ::1
Event-Date-Local: 2011-07-29 14:25:29

However, usually after failure in executing "hangup" and getting "-ERR ..."
in the event->last_sr_reply, I have wrong event, but not always (an -ERR
returned from executing "hangup" doesn't always result in a wrong event).
Here it is:

<2011-07-29 14:25:29> [NOTICE] -ERR invalid session id
[f320e4a6-db23-46d0-8d89-9957eccbd4c9]
<2011-07-28 15:55:23> [DEBUG] ESL event [0] : UUID:
<2011-07-28 15:55:23> [DEBUG] RECV EVENT
Content-Length: 6485
Content-Type: text/event-plain

Event-Name: CHANNEL_HANGUP
Core-UUID: 26b77cba-8fdd-486d-90ec-6844bca58c72
FreeSWITCH-Hostname: fs01
FreeSWITCH-IPv4: 10.1.1.46
FreeSWITCH-IPv6: %3A%3A1
Event-Date-Local: 2011-07-28%2015%3A55%3A23

In the above case, both Event ID and UUID are invalid.

I think i need more information regarding how to use the data elements in
ESL event, especially how to use the following:

    char last_reply[1024];
    /*! Las command reply when called with esl_send_recv */
    char last_sr_reply[1024];
    /*! Last event received. Only populated when **save_event is NULL */
    esl_event_t *last_event;
    /*! Last event received when called by esl_send_recv */
    esl_event_t *last_sr_event;
    /*! This will hold already processed events queued by esl_recv_event */
    esl_event_t *race_event;
    /*! Events that have content-type == text/plain and a body */
    esl_event_t *last_ievent;

My questions, what event should I check if I am interested in only
channel/call related events, and how? Do I check handle.last_event, or
handle.last_ievent, or what?

Please kindly advice.

Thanks,
D.Ma
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110803/ff14b149/attachment.html 

From lakersman2006 at yahoo.com  Tue Aug  2 21:43:31 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Tue, 2 Aug 2011 10:43:31 -0700 (PDT)
Subject: [Freeswitch-users] (no subject)
Message-ID: <1312307011.32126.YahooMailNeo@web161004.mail.bf1.yahoo.com>

I am writing a perl script to bridge a call and need to be able to pass back any early media to the caller so I have set "ignore_early_media=false". My problem is that when I am getting early media and then the call finishes, the app seems to be calling the bridge app again to make a new call, even though I have no loops programmed into the script. I have set "hangup_after_bridge=true" but that does not help. Any help would be greatly appreciated.



###################################My code#############################################

#!/usr/bin/perl -w

our $session;

my $ringback_tone = "%(2000,4000,440,480)";??? #US RINGBACK TONE

if ($session->ready ()) 
{????? 
??? #set bridge settings
??? $session->execute("set", "ringback=$ringback_tone");??? #set the ringback tone type
??? $session->execute("set", "instant_ringback=true");??? #set to ring instantly
??? $session->execute("set", "ignore_early_media=false");??? #set to NOT ignore early media
??? $session->execute("set", "call_timeout=20");??? ??? #only works if "ignore_early_media=true"
??? $session->execute("set", "bridge_answer_timeout=20"); ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? 
??? $session->execute("set", "progress_timeout=15");
??? $session->execute("set", "continue_on_fail=false");
??? $session->execute("set", "hangup_after_bridge=true"); 
??? $session->execute("set", "bridge_pre_execute_bleg_app=info"); 
??? $session->execute("bridge", "sofia/gateway/carrier1/5214498052059");
??? $session->hangup();
}
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/474ce490/attachment.html 

From infos at madovsky.org  Tue Aug  2 22:01:22 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 2 Aug 2011 14:01:22 -0400
Subject: [Freeswitch-users] mod_rtmp question
References: <794224500D9C43C3860909EE801A3DC8@e1705>
	<CAFiqYu=zOqWe6Gxecg74WCr_LXq2Ca0JoE8dbGJpDrk-5U=6nw@mail.gmail.com>
Message-ID: <E89B2D97D7DF46EDB81DF8DB3955291E@e1705>

Ok Steve,

it means that for the moment receive calls from 
odbc cluster with mod_rtmp as endpoint can't work neither (?)

Thanks

Franck
  ----- Original Message ----- 
  From: Steven Ayre 
  To: FreeSWITCH Users Help 
  Sent: Monday, August 01, 2011 3:48 AM
  Subject: Re: [Freeswitch-users] mod_rtmp question


  That's because mod_sofia stores registrations in a the database (which in your case must be a shared ODBC one) while mod_rtmp currently has everything in memory. Yes it's expected, and there's no workaround at the moment other than running the ESL command on all nodes.

  Don't assume all modules work in the same way...

  -Steve




  On 31 July 2011 21:13, Madovsky <infos at madovsky.org> wrote:

    I use freeswitch in ODBC mode.
    in sofia endpoint if I type
    "sofia status profile internal"
    I get all registrations from all nodes

    but in rtmp endpoint
    "rtmp status profile internal reg"

    I get registrations of the local node.

    is it a normal behaviour ?
    if yes which rtmp command can give the same
    result of sofia with ODBC ?

    Thanks

    Franck







    _______________________________________________
    Join us at ClueCon 2011, Aug 9-11, Chicago
    http://www.cluecon.com 877-7-4ACLUE

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





------------------------------------------------------------------------------


  _______________________________________________
  Join us at ClueCon 2011, Aug 9-11, Chicago
  http://www.cluecon.com 877-7-4ACLUE

  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/8f0ae6ef/attachment.html 

From msc at freeswitch.org  Tue Aug  2 22:16:58 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 2 Aug 2011 11:16:58 -0700
Subject: [Freeswitch-users] Help setting up SIP reg
In-Reply-To: <CAE=3qObVbDQi6g1m80WLtG5MhyosV2YYnqmOoausiz-yC+NmQQ@mail.gmail.com>
References: <CAE=3qOYPpTuRTvq7RTJeUgzsCtvrfCjMiKFGu1fg0PHyZRE+-g@mail.gmail.com>
	<CAFiqYu=c8mY5BMWPumy1CgZHZ81kMRPgoNiF-6mjFAkW1q30fA@mail.gmail.com>
	<CAE=3qOZxwERT3CQEFHetO_o2s=aQUwYZ54TmMRPx9eYD=A7NoQ@mail.gmail.com>
	<CAKzWOxUEsGUKojF-vvstL6FvcDMhePqYGdXEQXQ4x-EJ81Bk6w@mail.gmail.com>
	<CAE=3qOacZh50iepxJKJD5UvtesxZAsN7RnPKNe1zdLO9VmFsuA@mail.gmail.com>
	<CAE=3qOZsNuiimFBZJTf4SbPRA8bRq_hJfPUunEnT8rxZj3qTnA@mail.gmail.com>
	<CAE=3qObVuLOhp2GD29Da-GfzB_qkbkyDP4kPmRxjdbJi7XeirQ@mail.gmail.com>
	<CAKzWOxUyZoBS5Evp4b61vP8fm64Lef0n01ti3KmVraQO_BqYtA@mail.gmail.com>
	<CAE=3qObsDBDTsKY2ycK_NierbS_A+-f896yMW4OArTkgauYgRA@mail.gmail.com>
	<CAKzWOxXYi-yO9DNDw1rHCduKq=h9q=BddFmn2dKuqALKb=t1fA@mail.gmail.com>
	<CAE=3qObVbDQi6g1m80WLtG5MhyosV2YYnqmOoausiz-yC+NmQQ@mail.gmail.com>
Message-ID: <CAKzWOxVC51tge2o1xEAW2NTvA2ewUD-XnanLRgRjsymtrRT-Og@mail.gmail.com>

Lloyd,

The gateway you created I think has a few issues. Let's have you bypass it
altogether with a single change to your dialplan. Change your bridge line to
this:

<action application="bridge" data="sofia/external/01869$1 at 80.40.150.150"/>

This bypasses the gateway and sends the call straight out the external
profile. (You could also send it out the internal profile.)

Try that and see what happens. If you have issues then do the usual console
log and siptrace and put it into pastebin.freeswitch.org. Be sure to choose
"FreeSWITCH Log" for the syntax highlighting.

-MC

On Tue, Aug 2, 2011 at 3:41 AM, lloyd thomas <lloydie.t at gmail.com> wrote:

> Just did a test, but no joy. I suspect I may have to dispense with the
> gateway settings and just bridge straight from the dial plan, but it is just
> a guess.
>
>
> dialplan
> ----------------------------------------
> <extension name="Local Dial">
>    <condition field="destination_number" expression="^9([2-9][0-9]{5})$">
>      <action application="bridge"
> data="sofia/gateway/phisys-2circles/01869$1"/>
>    </condition>
> </extension>
>
> gateway
> ---------------------------------------
> <include>
>   <gateway name="phisys-2circles">
>     <param name="from-domain" value="80.40.150.150"/>
>     <param name="proxy" value="80.40.150.150"/>
>     <param name="sip-port" value="5060"/>
>   </gateway>
> </include>
>
>
>
> errors
> ------------------------------------
> 2011-08-02 11:32:42.056302 [DEBUG] mod_dptools.c:1059 sofia/internal/
> 200 at phisys.tele.phi.co.uk SET [RFC2822_DATE]=[Tue, 02 Aug 2011 11:32:42
> +0100]
> EXECUTE sofia/internal/200 at phisys.tele.phi.co.ukbridge(sofia/gateway/phisys-2circles/01869321110)
> 2011-08-02 11:32:42.073667 [ERR] mod_sofia.c:3940 Invalid Gateway
> 2011-08-02 11:32:42.073667 [NOTICE] mod_sofia.c:4282 Close Channel N/A
> [CS_NEW]
> 2011-08-02 11:32:42.076523 [DEBUG] switch_core_state_machine.c:452 ()
> Running State Change CS_DESTROY
> 2011-08-02 11:32:42.079574 [DEBUG] switch_core_state_machine.c:462 (N/A)
> State DESTROY
> 2011-08-02 11:32:42.079574 [DEBUG] mod_sofia.c:362 N/A SOFIA DESTROY
> 2011-08-02 11:32:42.081946 [DEBUG] switch_core_state_machine.c:462 (N/A)
> State DESTROY going to sleep
> 2011-08-02 11:32:42.084166 [ERR] switch_ivr_originate.c:2640 Cannot create
> outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]
> 2011-08-02 11:32:42.085649 [DEBUG] switch_ivr_originate.c:3506 Originate
> Resulted in Error Cause: 28 [INVALID_NUMBER_FORMAT]
> 2011-08-02 11:32:42.087275 [INFO] mod_dptools.c:2623 Originate Failed.
> Cause: INVALID_NUMBER_FORMAT
> 2011-08-02 11:32:42.088652 [DEBUG] switch_channel.c:2559 (sofia/internal/
> 200 at phisys.tele.phi.co.uk) Callstate Change RINGING -> HANGUP
> 2011-08-02 11:32:42.093248 [NOTICE] mod_dptools.c:2686 Hangup
> sofia/internal/200 at phisys.tele.phi.co.uk [CS_EXECUTE]
> [INVALID_NUMBER_FORMAT]
> 2011-08-02 11:32:42.096925 [DEBUG] switch_channel.c:2575 Send signal
> sofia/internal/200 at phisys.tele.phi.co.uk [KILL]
>
>
> On 1 August 2011 20:50, Michael Collins <msc at freeswitch.org> wrote:
>
>> Okay, so what happens when you dial out? Sorry, it's been a few days and I
>> don't recall where we left off. Be sure to include console log w/ siptrace
>> on pastebin.freeswitch.org.
>>
>> -MC
>>
>>
>> On Mon, Aug 1, 2011 at 12:35 PM, lloyd thomas <lloydie.t at gmail.com>wrote:
>>
>>> I think they have my IP on a white list.
>>>
>>>
>>> On 1 August 2011 16:24, Michael Collins <msc at freeswitch.org> wrote:
>>>
>>>> Do they challenge you (digest auth) or do they have your IP address on a
>>>> white list? That's a critical piece of information that only your provider
>>>> can supply.
>>>>
>>>> -MC
>>>>
>>>>
>>>> On Fri, Jul 29, 2011 at 9:31 PM, lloyd thomas <lloydie.t at gmail.com>wrote:
>>>>
>>>>> OK Inbound working with:
>>>>>
>>>>>   <!--2circles -->
>>>>>     <list name="domains" default="allow">
>>>>>       <node type="allow" cidr="80.40.150.150/32"/>
>>>>>     </list>
>>>>>
>>>>> Just need to sort outbound.
>>>>>
>>>>>
>>>>> On 30 July 2011 04:59, lloyd thomas <lloydie.t at gmail.com> wrote:
>>>>>
>>>>>> Hi, dialling in produces the following error.
>>>>>>
>>>>>> 2011-07-30 04:56:07.818936 [DEBUG] sofia.c:6517 IP 80.40.150.150
>>>>>> Rejected by acl "domains". Falling back to Digest auth.
>>>>>> 2011-07-30 04:56:07.826367 [WARNING] sofia_reg.c:1246 SIP auth
>>>>>> challenge (INVITE) on sofia profile 'internal' for
>>>>>> [01869******@172.16.XXX.XXX] from ip 80.40.150.150
>>>>>>
>>>>>>
>>>>>>
>>>>>> On 30 July 2011 04:34, lloyd thomas <lloydie.t at gmail.com> wrote:
>>>>>>
>>>>>>> I am registering with a them. I could not find suitable example in
>>>>>>> http://wiki.freeswitch.org/wiki/SIP_Provider_Examples which
>>>>>>>
>>>>>>>
>>>>>>> On 29 July 2011 21:57, Michael Collins <msc at freeswitch.org> wrote:
>>>>>>>
>>>>>>>> Are you registering with the provider or are they registering with
>>>>>>>> you? If they register with you then a user example is appropriate. If you
>>>>>>>> are registering with them then all you need is a gateway configured.
>>>>>>>> -MC
>>>>>>>>
>>>>>>>>
>>>>>>>> On Fri, Jul 29, 2011 at 1:40 PM, lloyd thomas <lloydie.t at gmail.com>wrote:
>>>>>>>>
>>>>>>>>> Sorry, example is not clear to me.
>>>>>>>>> I don't understand why a user config is relevant to sip
>>>>>>>>> registration for a provider.
>>>>>>>>> An example will help me more. Maybe CIDR attribute in a sip_profile
>>>>>>>>> gateway could help.
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On 29 July 2011 19:55, Steven Ayre <steveayre at gmail.com> wrote:
>>>>>>>>>
>>>>>>>>>> Look at the cidr attribute in the user directory to authenticate
>>>>>>>>>> by IP:
>>>>>>>>>> http://wiki.freeswitch.org/wiki/Acl#Users
>>>>>>>>>>
>>>>>>>>>> -Steve
>>>>>>>>>>
>>>>>>>>>> On 29 July 2011 19:38, lloyd thomas <lloydie.t at gmail.com> wrote:
>>>>>>>>>>
>>>>>>>>>>> *Hi I need a little help setting up a SIP registration for a
>>>>>>>>>>> provider that does not use auth.*
>>>>>>>>>>>
>>>>>>>>>>> *All I have is info below.*
>>>>>>>>>>> **
>>>>>>>>>>>
>>>>>>>>>>> *
>>>>>>>>>>> *
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> SBC/Proxy IP:        80.40.150.150:5060
>>>>>>>>>>>
>>>>>>>>>>> Authentication:      Trusted IP ? 88.221.85.33
>>>>>>>>>>>
>>>>>>>>>>> Assigned DDI:        01869******, 01869******
>>>>>>>>>>>
>>>>>>>>>>> DTMF Method:         RFC2833
>>>>>>>>>>>
>>>>>>>>>>> Status:              Live
>>>>>>>>>>>
>>>>>>>>>>> No. of trunks:       2x
>>>>>>>>>>>
>>>>>>>>>>> Session Timer:       1800
>>>>>>>>>>>
>>>>>>>>>>> Profile*:*             Generic (35060)
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> Apparently the following is used for *
>>>>>>>>>>>
>>>>>>>>>>> [vibe]
>>>>>>>>>>>
>>>>>>>>>>> type = friend
>>>>>>>>>>>
>>>>>>>>>>> host = 80.40.150.150
>>>>>>>>>>>
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>>
>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/eb17330a/attachment-0001.html 

From msc at freeswitch.org  Tue Aug  2 23:07:08 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 2 Aug 2011 12:07:08 -0700
Subject: [Freeswitch-users] Limit number of voicemails
In-Reply-To: <05a901cc511a$775ee8d0$661cba70$@com>
References: <4E37CA90.4090806@amooma.de> <05a901cc511a$775ee8d0$661cba70$@com>
Message-ID: <CAKzWOxXj1Cn1yL9d4CwxCZxHRKXdTcmVfKw00xh2VuH8CaxJbw@mail.gmail.com>

This is actually better than limiting the number of messages since you can
control actual disk space. (One REALLY long message will take more disk
space than five really short ones.)

However, in the interests of actually answering the question, and
demonstrating that with FreeSWITCH anything is possible, here is a simple
dialplan trick that will allow you to limit the number messages:

<extension name="check vm box limit" continue="true">
  <condition field="destination_number" expression="^(10[01][0-9])$"
break="never">
    <action application="set" data="dialed_extension=$1"/>
    <action application="set" data="dest_user=$1@${domain}" inline="true"/>
    <action application="set" data="box_count=${vm_boxcount ${dest_user}}"
inline="true">
    <action application="log" data="INFO User ${dest_user} has ${box_count}
message(s)"/>
  </condition>

  <condition field="${expr below(${boxcount},{$max_count})"
expression="^1$">
    <action application="voicemail" data="${domain} ${dialed_extension}"/>
    <anti-action application="playback"
data="voicemail/vm-mailbox_full.wav"/>
  </condition>
</extension>

The key is to use the vm_boxcount API to get the current message count for
that user, then use the expr API and the "below" function to make sure that
the box_count is less than whatever the max number of messages (max_count in
this example). The action and anti-action tags then handle the rest.

I just threw this together off the top of my head so please actually test it
out and make sure it works. Once you confirm that, please add it to the
voicemail wiki page.

-MC

On Tue, Aug 2, 2011 at 6:45 AM, Robert Huddleston <rhuddleston at gmail.com>wrote:

> Actually http://wiki.freeswitch.org/wiki/Mod_voicemail#vm-disk-quota
>
> In the manual...
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sascha
> Daniels
> Sent: Tuesday, August 02, 2011 6:00 AM
> To: FreeSWITCH Users Help
> Subject: [Freeswitch-users] Limit number of voicemails
>
> Hi together.
>
> For a small appliance I need to limit the number of voicemails for each
> user.
>
> I know that I can limit the length. That is the first step.
>
> Is there a way to disable the message recording, when the maximum number
> of voicemails is reached?
>
> Unfortunately I didn't find anything in the documentation.
>
> Kind regards
>
>                        Sascha
>
> --
> AMOOMA GmbH - Bachstr. 124 - 56566 Neuwied  -->  http://www.amooma.de
> Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister Montabaur B14998
>
> B?cher: http://das-asterisk-buch.de - http://ruby-auf-schienen.de
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/1a2739f3/attachment.html 

From msc at freeswitch.org  Tue Aug  2 23:13:34 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 2 Aug 2011 12:13:34 -0700
Subject: [Freeswitch-users] EXCHANGE_ROUTING_ERROR after ~70 transfer to
 dialplan extebsion
In-Reply-To: <576190BDB2B847A180F79DF6B311A49C@abisoft.biz>
References: <576190BDB2B847A180F79DF6B311A49C@abisoft.biz>
Message-ID: <CAKzWOxVeG7HTNmUsH0e6iRxgAd45u8WX=arDaq6B9B6gfgOoAA@mail.gmail.com>

If you have 70 hops through the dialplan then it's probably because you have
a routing loop. The 70 hops acts as a "circuit breaker" to keep the call
from looping forever.

The next step for you is to determine why your call is looping through the
dialplan so many times. is that truly needed?

-MC

2011/8/2 Kozak Vladimir <vkozak at abisoft.spb.ru>

> **
> Hi all.
> FreeSWITCH Version 1.0.head (git-9ff8f53 2011-05-03 12-13-52 -0400)
>
> I have problem with transfer channel to FS extension.
> I send from external system to FS command as
>    "SendMsg " + uniqueId + "\n" +
>    "call-command: execute\n" +
>    "execute-app-name: transfer\n" +
>    "execute-app-arg: " + destination + "\n\n";
>
> after ~70 correct transfer actions FS send BYE to my phone and send
> HANGUP_EVENT with cause EXCHANGE_ROUTING_ERROR to my system. (FS log
> attached)
>
> from FS loggs:
> 2011-08-09 04:48:38.525293 [DEBUG] mod_sofia.c:457 Channel
> sofia/internal/1000 at vkozak.starpoundtech.net hanging up, cause:
> EXCHANGE_ROUTING_ERROR
> 2011-08-09 04:48:38.525293 [DEBUG] mod_sofia.c:500 Sending BYE to
> sofia/internal/1000 at vkozak.starpoundtech.net
>
> what for does limitation of transfer operations exist?
> how can I avoid this issue?
> is't possible to increase number of hops for transfer command?
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/afbb6d15/attachment.html 

From msc at freeswitch.org  Tue Aug  2 23:16:00 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 2 Aug 2011 12:16:00 -0700
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <1312307011.32126.YahooMailNeo@web161004.mail.bf1.yahoo.com>
References: <1312307011.32126.YahooMailNeo@web161004.mail.bf1.yahoo.com>
Message-ID: <CAKzWOxVoX9U6gfbikbj0npSZj5JKzzj94vjT8hrNDtZjxD-Uxw@mail.gmail.com>

Get a console log w/ siptrace and drop it into pastebin.freeswitch.org. Be
sure to use "FreeSWITCH Log" as the syntax highlighting. Usually the logs
will have ample information to help determine what is happening.

-MC

On Tue, Aug 2, 2011 at 10:43 AM, Sam <lakersman2006 at yahoo.com> wrote:

> I am writing a perl script to bridge a call and need to be able to pass
> back any early media to the caller so I have set "ignore_early_media=false".
> My problem is that when I am getting early media and then the call finishes,
> the app seems to be calling the bridge app again to make a new call, even
> though I have no loops programmed into the script. I have set
> "hangup_after_bridge=true" but that does not help. Any help would be greatly
> appreciated.
>
>
> ###################################My
> code#############################################
> #!/usr/bin/perl -w
>
> our $session;
>
> my $ringback_tone = "%(2000,4000,440,480)";    #US RINGBACK TONE
>
> if ($session->ready ())
> {
>     #set bridge settings
>     $session->execute("set", "ringback=$ringback_tone");    #set the
> ringback tone type
>     $session->execute("set", "instant_ringback=true");    #set to ring
> instantly
>     $session->execute("set", "ignore_early_media=false");    #set to NOT
> ignore early media
>     $session->execute("set", "call_timeout=20");        #only works if
> "ignore_early_media=true"
>     $session->execute("set", "bridge_answer_timeout=20");
>
>     $session->execute("set", "progress_timeout=15");
>     $session->execute("set", "continue_on_fail=false");
>     $session->execute("set", "hangup_after_bridge=true");
>     $session->execute("set", "bridge_pre_execute_bleg_app=info");
>     $session->execute("bridge", "sofia/gateway/carrier1/5214498052059");
>     $session->hangup();
> }
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/67e6cea4/attachment.html 

From lakersman2006 at yahoo.com  Tue Aug  2 23:30:37 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Tue, 2 Aug 2011 12:30:37 -0700 (PDT)
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <CAKzWOxVoX9U6gfbikbj0npSZj5JKzzj94vjT8hrNDtZjxD-Uxw@mail.gmail.com>
References: <1312307011.32126.YahooMailNeo@web161004.mail.bf1.yahoo.com>
	<CAKzWOxVoX9U6gfbikbj0npSZj5JKzzj94vjT8hrNDtZjxD-Uxw@mail.gmail.com>
Message-ID: <1312313437.77623.YahooMailNeo@web161003.mail.bf1.yahoo.com>

MC,

Here is the link to the console/ sip trace log http://pastebin.freeswitch.org/16945



________________________________
From: Michael Collins <msc at freeswitch.org>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, August 2, 2011 12:16 PM
Subject: Re: [Freeswitch-users] (no subject)


Get a console log w/ siptrace and drop it into pastebin.freeswitch.org. Be sure to use "FreeSWITCH Log" as the syntax highlighting. Usually the logs will have ample information to help determine what is happening.

-MC


On Tue, Aug 2, 2011 at 10:43 AM, Sam <lakersman2006 at yahoo.com> wrote:

I am writing a perl script to bridge a call and need to be able to pass back any early media to the caller so I have set "ignore_early_media=false". My problem is that when I am getting early media and then the call finishes, the app seems to be calling the bridge app again to make a new call, even though I have no loops programmed into the script. I have set "hangup_after_bridge=true" but that does not help. Any help would be greatly appreciated.
>
>
>
>
>
>###################################My code#############################################
>
>#!/usr/bin/perl -w
>
>our $session;
>
>my $ringback_tone = "%(2000,4000,440,480)";??? #US RINGBACK TONE
>
>if ($session->ready ()) 
>{????? 
>??? #set
 bridge settings
>??? $session->execute("set", "ringback=$ringback_tone");??? #set the ringback tone type
>??? $session->execute("set", "instant_ringback=true");??? #set to ring instantly
>??? $session->execute("set", "ignore_early_media=false");??? #set to NOT ignore early media
>??? $session->execute("set", "call_timeout=20");??? ??? #only works if "ignore_early_media=true"
>??? $session->execute("set", "bridge_answer_timeout=20"); ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? 
>??? $session->execute("set",
 "progress_timeout=15");
>??? $session->execute("set", "continue_on_fail=false");
>??? $session->execute("set", "hangup_after_bridge=true"); 
>??? $session->execute("set", "bridge_pre_execute_bleg_app=info"); 
>??? $session->execute("bridge", "sofia/gateway/carrier1/5214498052059");
>??? $session->hangup();
>}
>_______________________________________________
>Join us at ClueCon 2011, Aug 9-11, Chicago
>http://www.cluecon.com 877-7-4ACLUE
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>

_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/8172c411/attachment-0001.html 

From anthony.minessale at gmail.com  Tue Aug  2 23:36:51 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 2 Aug 2011 14:36:51 -0500
Subject: [Freeswitch-users] EXCHANGE_ROUTING_ERROR after ~70 transfer to
 dialplan extebsion
In-Reply-To: <CAKzWOxVeG7HTNmUsH0e6iRxgAd45u8WX=arDaq6B9B6gfgOoAA@mail.gmail.com>
References: <576190BDB2B847A180F79DF6B311A49C@abisoft.biz>
	<CAKzWOxVeG7HTNmUsH0e6iRxgAd45u8WX=arDaq6B9B6gfgOoAA@mail.gmail.com>
Message-ID: <CAKbxfG8LhjsoFzMQE+BRgBk2HRGFiiHxTJUjvvXKb+F5Oo8ffw@mail.gmail.com>

increase the max_forwards header

On Tue, Aug 2, 2011 at 2:13 PM, Michael Collins <msc at freeswitch.org> wrote:
> If you have 70 hops through the dialplan then it's probably because you have
> a routing loop. The 70 hops acts as a "circuit breaker" to keep the call
> from looping forever.
> The next step for you is to determine why your call is looping through the
> dialplan so many times. is that truly needed?
> -MC
>
> 2011/8/2 Kozak Vladimir <vkozak at abisoft.spb.ru>
>>
>> Hi all.
>> FreeSWITCH Version 1.0.head (git-9ff8f53 2011-05-03 12-13-52 -0400)
>>
>> I have problem with transfer channel to FS extension.
>> I send from external system?to FS command as
>> ?? "SendMsg " + uniqueId + "\n" +
>> ???"call-command: execute\n" +
>> ???"execute-app-name: transfer\n" +
>> ???"execute-app-arg: " + destination + "\n\n";
>>
>> after ~70 correct transfer actions FS send BYE to my phone and send
>> HANGUP_EVENT with cause EXCHANGE_ROUTING_ERROR to my system. (FS log
>> attached)
>>
>> from FS loggs:
>> 2011-08-09 04:48:38.525293 [DEBUG] mod_sofia.c:457 Channel
>> sofia/internal/1000 at vkozak.starpoundtech.net hanging up, cause:
>> EXCHANGE_ROUTING_ERROR
>> 2011-08-09 04:48:38.525293 [DEBUG] mod_sofia.c:500 Sending BYE to
>> sofia/internal/1000 at vkozak.starpoundtech.net
>>
>> what for does limitation of transfer operations exist?
>> how can I avoid this issue?
>> is't possible to increase number of hops for transfer command?
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From Hector.Geraldino at ip-soft.net  Tue Aug  2 23:53:08 2011
From: Hector.Geraldino at ip-soft.net (Hector Geraldino)
Date: Tue, 2 Aug 2011 15:53:08 -0400
Subject: [Freeswitch-users] TTS/unimrcp engines
In-Reply-To: <361E98F99D3CC3439EED59BC1924ED69508754@SERVER2003.SecuReachSystems.local>
References: <361E98F99D3CC3439EED59BC1924ED69508754@SERVER2003.SecuReachSystems.local>
Message-ID: <6A6B4C284AD15042B429EB9D904544AD021FD8A24D@NY1-EXMB-01.ip-soft.net>

Hi Jason,

I've been playing with ASR/TTS and FS for the last 3 months. The best TTS engine I've tried so far is from Nuance, unfortunately their product is not available to the public (even the demo) and is quite expensive.

I have tested Cepstral and Loquendo voices, but none of them are close to Nuance's product. So it will depend on how professional you want your voices to sound, and of course your budget.


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jason Moran
Sent: Tuesday, August 02, 2011 11:57 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] TTS/unimrcp engines

I'm trying to wrap up which TTS engine we will use over unimrcp. We'll use between 10 and 40 simultaneous TTS lines.
So far I have tested the new IVONA mrcpv1 plugin but I am waiting on their dev's to work out a few issues I uncovered.

Which TTS engines have been successfully used by others?

I've heard good things about Loquendo, but I have had problems getting in contact with them.
Acapela sounds nice on their website - does anybody know how their unimrcp plugin sounds?
Cepstral is cheap, but it seems as though they do not have a current release for Open Suse 11.6 (32bit).

Any others? Recommendations?
-Jason
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/ce6d7e76/attachment.html 

From msc at freeswitch.org  Wed Aug  3 00:08:41 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 2 Aug 2011 13:08:41 -0700
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <1312313437.77623.YahooMailNeo@web161003.mail.bf1.yahoo.com>
References: <1312307011.32126.YahooMailNeo@web161004.mail.bf1.yahoo.com>
	<CAKzWOxVoX9U6gfbikbj0npSZj5JKzzj94vjT8hrNDtZjxD-Uxw@mail.gmail.com>
	<1312313437.77623.YahooMailNeo@web161003.mail.bf1.yahoo.com>
Message-ID: <CAKzWOxW40E4TSJ0p865iUiVLMd4PuROOx+UdN2KunBwrbsdW1Q@mail.gmail.com>

On Tue, Aug 2, 2011 at 12:30 PM, Sam <lakersman2006 at yahoo.com> wrote:

> MC,
>
> Here is the link to the console/ sip trace log
> http://pastebin.freeswitch.org/16945
>
>
Not sure what is happening. Pastebin the dialplan extension(s) that handle
this call. Something interesting must be going on.

-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/07b9d722/attachment.html 

From krice at freeswitch.org  Wed Aug  3 00:16:29 2011
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 02 Aug 2011 15:16:29 -0500
Subject: [Freeswitch-users] Infolink, ServerPronto, ColoPronto,
	Etc  <YES ITS IS OFF TOPIC>
Message-ID: <CA5DC54D.1A07C%krice@freeswitch.org>

Just a word of warning...

There is a company in Florida that operates under many company names
including but not limited to InfoLink, ServerPronto, ColoPronto, and
others...

This Company makes it IMPOSSIBLE to close an account... I had colocation
(rented a dedicated server) from them for several months, during the entire
time, the bandwidth was never right, altho they promised several TB of
transfer, bandwidth was such that promised transfer amounts were never
achievable... I secured separate Colo, issued an order to cancel the account
by opening a ticket through their secured web portal... These idiots
continue to attempt to charge my credit card to the point where I had to
cancel my card and order a new one... And til this day they continue to try
and collect and refuse to refund any money they collected after the cancel
order... 


Avoid these guys like the plague unless you want to jump through hoops

K

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/a0e426d4/attachment.html 

From lakersman2006 at yahoo.com  Wed Aug  3 00:19:57 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Tue, 2 Aug 2011 13:19:57 -0700 (PDT)
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <CAKzWOxW40E4TSJ0p865iUiVLMd4PuROOx+UdN2KunBwrbsdW1Q@mail.gmail.com>
References: <1312307011.32126.YahooMailNeo@web161004.mail.bf1.yahoo.com>
	<CAKzWOxVoX9U6gfbikbj0npSZj5JKzzj94vjT8hrNDtZjxD-Uxw@mail.gmail.com>
	<1312313437.77623.YahooMailNeo@web161003.mail.bf1.yahoo.com>
	<CAKzWOxW40E4TSJ0p865iUiVLMd4PuROOx+UdN2KunBwrbsdW1Q@mail.gmail.com>
Message-ID: <1312316397.65960.YahooMailNeo@web161007.mail.bf1.yahoo.com>

Here is my dialplan, pretty straight forward.

<extension name="public_did">
??? <condition field="destination_number" expression="^(\d{11})|(\d{10})$">
????? <action application="perl" data="/usr/local/freeswitch/scripts/test2.pl"/>
??? </condition>
? </extension>



________________________________
From: Michael Collins <msc at freeswitch.org>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, August 2, 2011 1:08 PM
Subject: Re: [Freeswitch-users] (no subject)





On Tue, Aug 2, 2011 at 12:30 PM, Sam <lakersman2006 at yahoo.com> wrote:

MC,
>
>
>Here is the link to the console/ sip trace log http://pastebin.freeswitch.org/16945
>
>

Not sure what is happening. Pastebin the dialplan extension(s) that handle this call. Something interesting must be going on.

-MC?
_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/987f340e/attachment.html 

From msc at freeswitch.org  Wed Aug  3 00:29:20 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 2 Aug 2011 13:29:20 -0700
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <1312316397.65960.YahooMailNeo@web161007.mail.bf1.yahoo.com>
References: <1312307011.32126.YahooMailNeo@web161004.mail.bf1.yahoo.com>
	<CAKzWOxVoX9U6gfbikbj0npSZj5JKzzj94vjT8hrNDtZjxD-Uxw@mail.gmail.com>
	<1312313437.77623.YahooMailNeo@web161003.mail.bf1.yahoo.com>
	<CAKzWOxW40E4TSJ0p865iUiVLMd4PuROOx+UdN2KunBwrbsdW1Q@mail.gmail.com>
	<1312316397.65960.YahooMailNeo@web161007.mail.bf1.yahoo.com>
Message-ID: <CAKzWOxUc=4rfTgnAxpRd7EFUA3F9Q8iBZMneNoTzyjFNeLCv3Q@mail.gmail.com>

In your pastebin of the call log, how many different calls was that? I saw
like 4 incoming calls, however I couldn't specifically see the issue you
were experiencing. Can you look at your log and see if you can isolate the
approximate log lines where the issue is occurring?

-MC

On Tue, Aug 2, 2011 at 1:19 PM, Sam <lakersman2006 at yahoo.com> wrote:

> Here is my dialplan, pretty straight forward.
>
> <extension name="public_did">
>     <condition field="destination_number" expression="^(\d{11})|(\d{10})$">
>       <action application="perl" data="/usr/local/freeswitch/scripts/
> test2.pl"/>
>     </condition>
>   </extension>
>
> ------------------------------
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Tuesday, August 2, 2011 1:08 PM
> *Subject:* Re: [Freeswitch-users] (no subject)
>
>
>
> On Tue, Aug 2, 2011 at 12:30 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> MC,
>
> Here is the link to the console/ sip trace log
> http://pastebin.freeswitch.org/16945
>
>
> Not sure what is happening. Pastebin the dialplan extension(s) that handle
> this call. Something interesting must be going on.
>
> -MC
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/11d59ffa/attachment-0001.html 

From lakersman2006 at yahoo.com  Wed Aug  3 00:38:17 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Tue, 2 Aug 2011 13:38:17 -0700 (PDT)
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <CAKzWOxUc=4rfTgnAxpRd7EFUA3F9Q8iBZMneNoTzyjFNeLCv3Q@mail.gmail.com>
References: <1312307011.32126.YahooMailNeo@web161004.mail.bf1.yahoo.com>
	<CAKzWOxVoX9U6gfbikbj0npSZj5JKzzj94vjT8hrNDtZjxD-Uxw@mail.gmail.com>
	<1312313437.77623.YahooMailNeo@web161003.mail.bf1.yahoo.com>
	<CAKzWOxW40E4TSJ0p865iUiVLMd4PuROOx+UdN2KunBwrbsdW1Q@mail.gmail.com>
	<1312316397.65960.YahooMailNeo@web161007.mail.bf1.yahoo.com>
	<CAKzWOxUc=4rfTgnAxpRd7EFUA3F9Q8iBZMneNoTzyjFNeLCv3Q@mail.gmail.com>
Message-ID: <1312317497.83886.YahooMailNeo@web161017.mail.bf1.yahoo.com>

In the pastebin, that was only 1 single call that was executed with the bridge app. The issue is that once the call channel is shutdown, a new channel gets created even though the call should be hung up. And I have noticed that this occurs when I leave out $session->answer(); in the beginning of the script. I was told by someone else on the mailing list to leave out $session->answer(); since the call has already been answered by the dialplan.



________________________________
From: Michael Collins <msc at freeswitch.org>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, August 2, 2011 1:29 PM
Subject: Re: [Freeswitch-users] (no subject)


In your pastebin of the call log, how many different calls was that? I saw like 4 incoming calls, however I couldn't specifically see the issue you were experiencing. Can you look at your log and see if you can isolate the approximate log lines where the issue is occurring?

-MC


On Tue, Aug 2, 2011 at 1:19 PM, Sam <lakersman2006 at yahoo.com> wrote:

Here is my dialplan, pretty straight forward.
>
>
><extension name="public_did">
>??? <condition field="destination_number" expression="^(\d{11})|(\d{10})$">
>????? <action application="perl" data="/usr/local/freeswitch/scripts/test2.pl"/>
>??? </condition>
>? </extension>
>
>
>
>________________________________
>From: Michael Collins <msc at freeswitch.org>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>Sent: Tuesday, August 2, 2011 1:08 PM
>Subject: Re: [Freeswitch-users] (no subject)
>
>
>
>
>
>
>On Tue, Aug 2, 2011 at 12:30 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
>MC,
>>
>>
>>Here is the link to the console/ sip trace log http://pastebin.freeswitch.org/16945
>>
>>
>
>
>Not sure what is happening. Pastebin the dialplan extension(s) that handle this call. Something interesting must be going on.
>
>
>-MC?
>_______________________________________________
>Join us at ClueCon 2011, Aug 9-11, Chicago
>http://www.cluecon.com 877-7-4ACLUE
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>
>
>_______________________________________________
>Join us at ClueCon 2011, Aug 9-11, Chicago
>http://www.cluecon.com 877-7-4ACLUE
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>

_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/e6c93061/attachment.html 

From tculjaga at gmail.com  Wed Aug  3 01:23:14 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 2 Aug 2011 23:23:14 +0200
Subject: [Freeswitch-users] Mod_rad_auth issue for FS working with
 FreeRadius server
In-Reply-To: <CAOTmbMyvyDA-CRpaMC=WRGVkiyTG=Rs+n4hyeQWdyTZpqysSzw@mail.gmail.com>
References: <CAOTmbMzvd5OHL8z=_8vu6Orp2_prGxaw1gmXYCM2=2+pgzs_WA@mail.gmail.com>
	<CAJoKKdX01Qc8ibs25PUELi8GrzMDNOrHb_bQvYdq3ZqA4XLy4w@mail.gmail.com>
	<CAOTmbMxL3pLOuw655fyT5VzAON_N0PrDN+-i-++=oZoYU6+Fmg@mail.gmail.com>
	<CAOTmbMyvyDA-CRpaMC=WRGVkiyTG=Rs+n4hyeQWdyTZpqysSzw@mail.gmail.com>
Message-ID: <CAJoKKdXSwF02S9tBkorGwOjj57AtkRpU0c177qXASEpodKVk=w@mail.gmail.com>

did u use the dictionary i have attached ?

On Tue, Aug 2, 2011 at 10:08 AM, fieldpeak <fieldpeak at gmail.com> wrote:

> i tried change to 'h323-conf-id' to 'h323-call-origin' in
> 02_unitest_rad-ANI-auth.xml, rad_auth.conf.xml, however, it still prompt
> '[ERR] mod_rad_auth.c:428 Unknown attribute: key:h323-conf-id, not found
> in dictionary', so where the mod_rad_auth read out the 'h323-conf-id'? very
> very strange, which dictionary it was using...
>
> Regards,
> Charles
>
>
> 2011/8/2 fieldpeak <fieldpeak at gmail.com>
>
>> Hi Tihomir,
>>
>> Finally the answer coming, i see the hope, thanks for your reply, :)
>>
>> As your advise, i only use one attribute(h323-conf-id) in my dialplan, and
>> only one attribute(h323-conf-id) in rad_auth.conf.xml, and using the
>> attached dictionary (from ciso) which contains this attribute, however, it
>> still prompt 'unknown attribute', so i suspected if it was reading
>> /usr/local/etc/radiusclient/dictionary, so i copy the same dictionary to
>> /usr/local/freeswitch/radius/, it did not any help at all... very strange...
>>
>> Log:
>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set default_realm :=
>> .
>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set radius_timeout
>> := 3.
>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set radius_retries
>> := 2.
>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set radius_deadtime
>> := 0.
>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set bindaddr := *.
>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:371 ... radius:
>> User-Name: 38516060333
>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:380 ... radius:
>> User-Password: 003282
>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:396 ... radius:
>> Called-station-Id: 16094191500
>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:413 Handle attribute:
>> h323-conf-id
>> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:428 Unknown attribute:
>> key:h323-conf-id, not found in dictionary
>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:538 abort sending radius
>> packet.
>> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:546 An error occured
>> during RADIUS Authentication(RC=-1)
>> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:702 An error occured
>> during radius authorization.
>>
>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  AUTH_RESULT=)
>>
>>
>>
>>   <extension name="unitest_rad-ANI-auth">
>>     <condition field="destination_number" expression="^601$">
>>       <!-- <action application="log" data="INFO  Before Auth "/> -->
>>
>>       <action inline="true" application="set" data="CALLID=h323-conf-id
>> =${uuid}"/>
>>
>>       <action inline="true" application="set" data="USERNAME=1001"/>
>>       <action inline="true" application="set" data="PASSWD=1111"/>
>>
>>
>>       <action application="sleep" data="2000"/>
>>       <action application="auth_function" data="in ${DIALED_NUMBER}, in
>> ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>
>>
>>     </condition>
>>   </extension>
>>
>>
>>
>> <configuration name="rad_auth.conf" description="radius authentification
>> module">
>>   <settings>
>>
>>   </settings>
>>
>>   <client>
>>     <param name="authserver" value="127.0.0.1:1812:gateway"/>
>>     <param name="dictionary"
>> value="/usr/local/etc/radiusclient/dictionary"/>
>>     <param name="seqfile" value="/var/run/radius.seq"/>
>>     <param name="mapfile"
>> value="/usr/local/etc/radiusclient/port-id-map"/>
>>     <param name="default_realm" value=""/>
>>     <param name="radius_timeout" value="3"/>
>>     <param name="radius_retries" value="2"/>
>>     <param name="radius_deadtime" value="0"/>
>>     <param name="bindaddr" value="*"/>
>>   </client>
>>
>>   <vsas>
>>
>>
>>     <param name="h323-conf-id" id="24" value="CALLID" pec="9" expr="1"
>> direction="in"/>
>>
>>   </vsas>
>>  </configuration>
>>
>>
>>
>> 2011/8/2 Tihomir Culjaga <tculjaga at gmail.com>
>>
>>> hi,
>>>
>>> dictionary.all is just the name of a file containing all attributes i
>>> needed at that time.
>>>
>>> you can include other dictionaries by putting #INCLUDE <pathname> at the
>>> end of the dictionary file you reference in rad_auth.conf.xml.
>>> if the INCLUDE doesn't work, just append dictionary.cisco to your
>>> dictionary file... and make your own file.
>>>
>>>
>>> check inline comments down below...
>>>
>>>
>>> T.
>>>
>>>
>>> On Sun, Jul 31, 2011 at 10:46 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>>>
>>>> Hello Gurus,
>>>>
>>>> i met a issue when using
>>>> mod_rad_auth(http://wiki.freeswitch.org/wiki/Mod_rad_auth) to works
>>>> with freeradius server+mysql for AAA, the details is below, Could
>>>> anyone give any hints, Thanks in advance.
>>>>
>>>> i setup a dial plan "unitest_rad-ANI-auth" as wiki above, however,
>>>> when i dialed 601 to trigger the dial plan, the console show errors,
>>>> it looks "h323-conf-id" is not in the directory, then i tried to add
>>>> this attribute to the dictionary, however, it does not help, in the
>>>> wiki, it mentioned the rad_auth.conf.xml contains <param
>>>> name="dictionary"
>>>> value="/usr/local/etc/radiusclient/dictionary.all"/>, however i did
>>>> not find the file "dictionary.all" at that directory, so i use
>>>> dictionary. BTW, the freeradius server + mysql works well.
>>>>
>>>
>>> i just appended the information needed into dictionary.all file...
>>> (vendor and attribute definition).
>>>
>>>
>>>
>>>>
>>>> console errors:
>>>>
>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 auth_function(in , in
>>>> 38516060333, in 003282, out AUTH_RESULT)
>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:301 allocate initial
>>>> structure.
>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:313 initialzed
>>>> configuration.
>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set authserver
>>>> := 127.0.0.1:1812:gateway.
>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set dictionary
>>>> := /usr/local/etc/radiusclient/dictionary.
>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set seqfile :=
>>>> /var/run/radius.seq.
>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set mapfile :=
>>>> /usr/local/etc/radiusclient/port-id-map.
>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set default_realm
>>>> := .
>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set radius_timeout
>>>> := 3.
>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set radius_retries
>>>> := 2.
>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>> radius_deadtime := 0.
>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set bindaddr := *.
>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:371 ... radius:
>>>> User-Name: 38516060333
>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:380 ... radius:
>>>> User-Password: 003282
>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:391 ... radius:
>>>> Called-station-Id is empty, ignoring...
>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:413 Handle
>>>> attribute: h323-conf-id
>>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:428 Unknown attribute:
>>>> key:h323-conf-id, not found in dictionary
>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:538 abort sending
>>>> radius packet.
>>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:546 An error occured
>>>> during RADIUS Authentication(RC=-1)
>>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:702 An error occured
>>>> during radius authorization.
>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  AUTH_RESULT=)
>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  AUTH_RESULT=
>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  billing_model=)
>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  billing_model=
>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  credit_amount=)
>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  credit_amount=
>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  currency=)
>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  currency=
>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  preffered_lang=)
>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  preffered_lang=
>>>>
>>>> added below in the dictionary(/usr/local/etc/radiusclient/dictionary):
>>>>
>>>> ATTRIBUTE       h323-conf-id            1008    string
>>>>
>>>
>>> you need the vendor definition as well
>>>
>>>
>>>>
>>>>
>>>> dial plan:
>>>> <extension name="unitest_rad-ANI-auth">
>>>>    <condition field="destination_number" expression="^601$">
>>>>      <action application="log" data="INFO  Before Auth "/>
>>>>
>>>>      <action inline="true" application="set"
>>>> data="CALLID=h323-conf-id=${uuid}"/>
>>>>      <action inline="true" application="set"
>>>> data="SERVICENUM=h323-prompt-id=${destination_number}"/>
>>>>      <action inline="true" application="set"
>>>> data="TRANSACTIONID=h323-ivr-out=transactionID:1234"/>
>>>>  <!--      <action inline="true" application="set"
>>>> data="CALLINGNUMBER=${caller_id_number}"/> -->
>>>>      <action inline="true" application="set"
>>>> data="CALLINGNUMBER=38516060333"/>
>>>>      <action inline="true" application="set"
>>>> data="USERNAME=38516060333"/>
>>>>  <!--      <action inline="true" application="set"
>>>> data="USERNAME=209354"/> -->
>>>>      <action inline="true" application="set" data="PASSWD=003282"/>
>>>>  <!--      <action inline="true" application="set"
>>>> data="DIALED_NUMBER=16094191500"/>  -->
>>>>
>>>>      <action application="sleep" data="2000"/>
>>>>      <action application="auth_function" data="in ${DIALED_NUMBER},
>>>> in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>
>>>>
>>>>
>>>>      <action application="log" data="INFO  AUTH_RESULT=${AUTH_RESULT}"/>
>>>>      <action application="log" data="INFO
>>>>  billing_model=${billing_model}"/>
>>>>      <action application="log" data="INFO
>>>>  credit_amount=${credit_amount}"/>
>>>>      <action application="log" data="INFO  currency=${currency}"/>
>>>>      <action application="log" data="INFO
>>>>  preffered_lang=${preffered_lang}"/>
>>>>      <action application="log" data="INFO  credit_time=${credit_time}"/>
>>>>      <action application="log" data="INFO
>>>> h323_ivr_duration=${h323_ivr_duration}"/>
>>>>      <action application="log" data="INFO  return_code=${return_code}"/>
>>>>      <!-- <action application="execute_extension" data="AUTH XML
>>>> default"/> -->
>>>>    </condition>
>>>>  </extension>
>>>>
>>>>  radius_cdr.conf.xml:
>>>>  <configuration name="radius_cdr.conf" description="RADIUS CDR
>>>> Configuration">
>>>>
>>>>        <settings>
>>>>
>>>>                <!-- location of the radius dictionary files -->
>>>>
>>>>                <param name="dictionary"
>>>> value="/usr/local/freeswitch/conf/radius/dictionary"/>
>>>>
>>>>
>>> your dictionary file need to contain all the attributes you are trying to
>>> use or to include other dictionaries (In this case dictionary.cisco) from
>>> the dictionary file you are referencing here.
>>>
>>>
>>>>                <!-- number of retries for each server -->
>>>>
>>>>                <param name="radius_retries" value="3"/>
>>>>
>>>>                <!-- number of seconds to wait between retries -->
>>>>
>>>>                <param name="radius_timeout" value="5"/>
>>>>
>>>>                <!-- accounting servers, up to 8 allowed -->
>>>>
>>>>                <!-- value is "host:port:secret", port is optional -->
>>>>
>>>>                <!-- use IP ADDRESSES, not hostnames -->
>>>>
>>>>                <param name="acct_server" value="127.0.0.1:1813
>>>> :testing123"/>
>>>>
>>>>
>>>>        </settings>
>>>>
>>>> </configuration>
>>>>
>>>>  the FS version:
>>>>  FreeSWITCH Version 1.0.head (git-492bc6b 2011-07-23 12-53-04 -0400)
>>>>
>>>>  Regards,
>>>>  Charles
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>>
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/0bb9369f/attachment-0001.html 

From jmoran at secureachsystems.com  Wed Aug  3 01:27:49 2011
From: jmoran at secureachsystems.com (Jason Moran)
Date: Tue, 2 Aug 2011 17:27:49 -0400
Subject: [Freeswitch-users] TTS/unimrcp engines
References: <361E98F99D3CC3439EED59BC1924ED69508754@SERVER2003.SecuReachSystems.local>
	<6A6B4C284AD15042B429EB9D904544AD021FD8A24D@NY1-EXMB-01.ip-soft.net>
Message-ID: <361E98F99D3CC3439EED59BC1924ED695466DC@SERVER2003.SecuReachSystems.local>

Thanks Hector, Nuance is likely a bit out of our price range. Hopefully
we can find a professional voice for a reasonable cost.  Did you test
these out on windows, linux? Which versions?

 

Do you think I'd need a separate server to run unimrcp+TTS or could I
get away with it being on the same server as FS for a little while?

-Jason

 

 

From: Hector Geraldino [mailto:Hector.Geraldino at ip-soft.net] 
Sent: Tuesday, August 02, 2011 3:53 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] TTS/unimrcp engines

 

Hi Jason,

 

I've been playing with ASR/TTS and FS for the last 3 months. The best
TTS engine I've tried so far is from Nuance, unfortunately their product
is not available to the public (even the demo) and is quite expensive.

 

I have tested Cepstral and Loquendo voices, but none of them are close
to Nuance's product. So it will depend on how professional you want your
voices to sound, and of course your budget.

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Jason Moran
Sent: Tuesday, August 02, 2011 11:57 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] TTS/unimrcp engines

 

I'm trying to wrap up which TTS engine we will use over unimrcp. We'll
use between 10 and 40 simultaneous TTS lines.

So far I have tested the new IVONA mrcpv1 plugin but I am waiting on
their dev's to work out a few issues I uncovered.

 

Which TTS engines have been successfully used by others?

 

I've heard good things about Loquendo, but I have had problems getting
in contact with them.

Acapela sounds nice on their website - does anybody know how their
unimrcp plugin sounds?

Cepstral is cheap, but it seems as though they do not have a current
release for Open Suse 11.6 (32bit). 

 

Any others? Recommendations?

-Jason

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/3e43d732/attachment.html 

From Hector.Geraldino at ip-soft.net  Wed Aug  3 02:04:39 2011
From: Hector.Geraldino at ip-soft.net (Hector Geraldino)
Date: Tue, 2 Aug 2011 18:04:39 -0400
Subject: [Freeswitch-users] TTS/unimrcp engines
In-Reply-To: <361E98F99D3CC3439EED59BC1924ED695466DC@SERVER2003.SecuReachSystems.local>
References: <361E98F99D3CC3439EED59BC1924ED69508754@SERVER2003.SecuReachSystems.local>
	<6A6B4C284AD15042B429EB9D904544AD021FD8A24D@NY1-EXMB-01.ip-soft.net>
	<361E98F99D3CC3439EED59BC1924ED695466DC@SERVER2003.SecuReachSystems.local>
Message-ID: <6A6B4C284AD15042B429EB9D904544AD021FD8A262@NY1-EXMB-01.ip-soft.net>

On RHEL 5.

I did use a separate box to run the speech server, but obviously it's not a must-do requirement. In fact in my devel environment both FS and Nuance Speech_Server + Vocalizer for Network + Recognizer (ASR) are running on the same machine.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jason Moran
Sent: Tuesday, August 02, 2011 5:28 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] TTS/unimrcp engines

Thanks Hector, Nuance is likely a bit out of our price range. Hopefully we can find a professional voice for a reasonable cost.  Did you test these out on windows, linux? Which versions?

Do you think I'd need a separate server to run unimrcp+TTS or could I get away with it being on the same server as FS for a little while?
-Jason


From: Hector Geraldino [mailto:Hector.Geraldino at ip-soft.net]
Sent: Tuesday, August 02, 2011 3:53 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] TTS/unimrcp engines

Hi Jason,

I've been playing with ASR/TTS and FS for the last 3 months. The best TTS engine I've tried so far is from Nuance, unfortunately their product is not available to the public (even the demo) and is quite expensive.

I have tested Cepstral and Loquendo voices, but none of them are close to Nuance's product. So it will depend on how professional you want your voices to sound, and of course your budget.


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jason Moran
Sent: Tuesday, August 02, 2011 11:57 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] TTS/unimrcp engines

I'm trying to wrap up which TTS engine we will use over unimrcp. We'll use between 10 and 40 simultaneous TTS lines.
So far I have tested the new IVONA mrcpv1 plugin but I am waiting on their dev's to work out a few issues I uncovered.

Which TTS engines have been successfully used by others?

I've heard good things about Loquendo, but I have had problems getting in contact with them.
Acapela sounds nice on their website - does anybody know how their unimrcp plugin sounds?
Cepstral is cheap, but it seems as though they do not have a current release for Open Suse 11.6 (32bit).

Any others? Recommendations?
-Jason
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/cab915f0/attachment.html 

From msc at freeswitch.org  Wed Aug  3 02:14:23 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 2 Aug 2011 15:14:23 -0700
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <1312317497.83886.YahooMailNeo@web161017.mail.bf1.yahoo.com>
References: <1312307011.32126.YahooMailNeo@web161004.mail.bf1.yahoo.com>
	<CAKzWOxVoX9U6gfbikbj0npSZj5JKzzj94vjT8hrNDtZjxD-Uxw@mail.gmail.com>
	<1312313437.77623.YahooMailNeo@web161003.mail.bf1.yahoo.com>
	<CAKzWOxW40E4TSJ0p865iUiVLMd4PuROOx+UdN2KunBwrbsdW1Q@mail.gmail.com>
	<1312316397.65960.YahooMailNeo@web161007.mail.bf1.yahoo.com>
	<CAKzWOxUc=4rfTgnAxpRd7EFUA3F9Q8iBZMneNoTzyjFNeLCv3Q@mail.gmail.com>
	<1312317497.83886.YahooMailNeo@web161017.mail.bf1.yahoo.com>
Message-ID: <CAKzWOxVPE13SgeOwPZE3mqOVfoxqzwkQN-8Fc3z_Pc8neYnV9A@mail.gmail.com>

Just a thought... try adding another "breakout" for your loop...

#!/usr/bin/perl -w

our $session;

my $ringback_tone = "%(2000,4000,440,480)";    #US RINGBACK TONE
my $end_call = 0;
if ( $session->ready() && !$end_call )
{
    #set bridge settings
    $session->execute("set", "ringback=$ringback_tone");    #set the
ringback tone type
    $session->execute("set", "instant_ringback=true");    #set to ring
instantly
    $session->execute("set", "ignore_early_media=false");    #set to NOT
ignore early media
    $session->execute("set", "call_timeout=20");        #only works if
"ignore_early_media=true"
    $session->execute("set", "bridge_answer_timeout=20");

    $session->execute("set", "progress_timeout=15");
    $session->execute("set", "continue_on_fail=false");
    $session->execute("set", "hangup_after_bridge=true");
    $session->execute("set", "bridge_pre_execute_bleg_app=info");
    $session->execute("bridge", "sofia/gateway/carrier1/5214498052059");
    $session->hangup();
    $end_call = 1;
}

Also, I don't know if it was a typo or not, but you had this:

if ($session->ready ())

as opposed to

if ($session->ready())

Make sure that you fix that before testing further. :)

-MC

On Tue, Aug 2, 2011 at 1:38 PM, Sam <lakersman2006 at yahoo.com> wrote:

> In the pastebin, that was only 1 single call that was executed with the
> bridge app. The issue is that once the call channel is shutdown, a new
> channel gets created even though the call should be hung up. And I have
> noticed that this occurs when I leave out $session->answer(); in the
> beginning of the script. I was told by someone else on the mailing list to
> leave out $session->answer(); since the call has already been answered by
> the dialplan.
>
> ------------------------------
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Tuesday, August 2, 2011 1:29 PM
>
> *Subject:* Re: [Freeswitch-users] (no subject)
>
> In your pastebin of the call log, how many different calls was that? I saw
> like 4 incoming calls, however I couldn't specifically see the issue you
> were experiencing. Can you look at your log and see if you can isolate the
> approximate log lines where the issue is occurring?
>
> -MC
>
> On Tue, Aug 2, 2011 at 1:19 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> Here is my dialplan, pretty straight forward.
>
> <extension name="public_did">
>     <condition field="destination_number" expression="^(\d{11})|(\d{10})$">
>       <action application="perl" data="/usr/local/freeswitch/scripts/
> test2.pl"/>
>     </condition>
>   </extension>
>
> ------------------------------
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Tuesday, August 2, 2011 1:08 PM
> *Subject:* Re: [Freeswitch-users] (no subject)
>
>
>
> On Tue, Aug 2, 2011 at 12:30 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> MC,
>
> Here is the link to the console/ sip trace log
> http://pastebin.freeswitch.org/16945
>
>
> Not sure what is happening. Pastebin the dialplan extension(s) that handle
> this call. Something interesting must be going on.
>
> -MC
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/d6bcc417/attachment-0001.html 

From lakersman2006 at yahoo.com  Wed Aug  3 02:48:24 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Tue, 2 Aug 2011 15:48:24 -0700 (PDT)
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <CAKzWOxVPE13SgeOwPZE3mqOVfoxqzwkQN-8Fc3z_Pc8neYnV9A@mail.gmail.com>
References: <1312307011.32126.YahooMailNeo@web161004.mail.bf1.yahoo.com>
	<CAKzWOxVoX9U6gfbikbj0npSZj5JKzzj94vjT8hrNDtZjxD-Uxw@mail.gmail.com>
	<1312313437.77623.YahooMailNeo@web161003.mail.bf1.yahoo.com>
	<CAKzWOxW40E4TSJ0p865iUiVLMd4PuROOx+UdN2KunBwrbsdW1Q@mail.gmail.com>
	<1312316397.65960.YahooMailNeo@web161007.mail.bf1.yahoo.com>
	<CAKzWOxUc=4rfTgnAxpRd7EFUA3F9Q8iBZMneNoTzyjFNeLCv3Q@mail.gmail.com>
	<1312317497.83886.YahooMailNeo@web161017.mail.bf1.yahoo.com>
	<CAKzWOxVPE13SgeOwPZE3mqOVfoxqzwkQN-8Fc3z_Pc8neYnV9A@mail.gmail.com>
Message-ID: <1312325304.39236.YahooMailNeo@web161019.mail.bf1.yahoo.com>

I made the corrections you suggested in my code, but still the same behavior. The bridge app seems to be in some type of internal loop and it does not end.



________________________________
From: Michael Collins <msc at freeswitch.org>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, August 2, 2011 3:14 PM
Subject: Re: [Freeswitch-users] (no subject)


Just a thought... try adding another "breakout" for your loop...

#!/usr/bin/perl -w

our $session;

my $ringback_tone = "%(2000,4000,440,480)";??? #US RINGBACK TONE
my $end_call = 0;
if ( $session->ready() && !$end_call )?
{??????
??? #set bridge settings
??? $session->execute("set", "ringback=$ringback_tone");??? #set the ringback tone type
??? $session->execute("set", "instant_ringback=true");??? #set to ring instantly
??? $session->execute("set", "ignore_early_media=false");??? #set to NOT ignore early media
??? $session->execute("set", "call_timeout=20");??? ??? #only works if "ignore_early_media=true"
??? $session->execute("set", "bridge_answer_timeout=20"); ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ????
??? $session->execute("set", "progress_timeout=15");
??? $session->execute("set", "continue_on_fail=false");
??? $session->execute("set", "hangup_after_bridge=true");?
??? $session->execute("set", "bridge_pre_execute_bleg_app=info");?
??? $session->execute("bridge", "sofia/gateway/carrier1/5214498052059");
??? $session->hangup();
? ? $end_call = 1;?
}

Also, I don't know if it was a typo or not, but you had this:

if ($session->ready ())

as opposed to?

if ($session->ready())

Make sure that you fix that before testing further. :)

-MC

On Tue, Aug 2, 2011 at 1:38 PM, Sam <lakersman2006 at yahoo.com> wrote:

In the pastebin, that was only 1 single call that was executed with the bridge app. The issue is that once the call channel is shutdown, a new channel gets created even though the call should be hung up. And I have noticed that this occurs when I leave out $session->answer(); in the beginning of the script. I was told by someone else on the mailing list to leave out $session->answer(); since the call has already been answered by the dialplan.
>
>
>
>
>________________________________
>From: Michael Collins <msc at freeswitch.org>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>Sent: Tuesday, August 2, 2011 1:29 PM
>
>Subject: Re: [Freeswitch-users] (no subject)
>
>
>
>In your pastebin of the call log, how many different calls was that? I saw like 4 incoming calls, however I couldn't specifically see the issue you were experiencing. Can you look at your log and see if you can isolate the approximate log lines where the issue is occurring?
>
>
>-MC
>
>
>On Tue, Aug 2, 2011 at 1:19 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
>Here is my dialplan, pretty straight forward.
>>
>>
>><extension name="public_did">
>>??? <condition field="destination_number" expression="^(\d{11})|(\d{10})$">
>>????? <action application="perl" data="/usr/local/freeswitch/scripts/test2.pl"/>
>>??? </condition>
>>? </extension>
>>
>>
>>
>>________________________________
>>From: Michael Collins <msc at freeswitch.org>
>>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>Sent: Tuesday, August 2, 2011 1:08 PM
>>Subject: Re: [Freeswitch-users] (no subject)
>>
>>
>>
>>
>>
>>
>>On Tue, Aug 2, 2011 at 12:30 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>
>>MC,
>>>
>>>
>>>Here is the link to the console/ sip trace log http://pastebin.freeswitch.org/16945
>>>
>>>
>>
>>
>>Not sure what is happening. Pastebin the dialplan extension(s) that handle this call. Something interesting must be going on.
>>
>>
>>-MC?
>>_______________________________________________
>>Join us at ClueCon 2011, Aug 9-11, Chicago
>>http://www.cluecon.com 877-7-4ACLUE
>>
>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>>
>>
>>_______________________________________________
>>Join us at ClueCon 2011, Aug 9-11, Chicago
>>http://www.cluecon.com 877-7-4ACLUE
>>
>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>>
>
>_______________________________________________
>Join us at ClueCon 2011, Aug 9-11, Chicago
>http://www.cluecon.com 877-7-4ACLUE
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>
>
>_______________________________________________
>Join us at ClueCon 2011, Aug 9-11, Chicago
>http://www.cluecon.com 877-7-4ACLUE
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>

_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/a1023023/attachment.html 

From msc at freeswitch.org  Wed Aug  3 02:56:57 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 2 Aug 2011 15:56:57 -0700
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <1312325304.39236.YahooMailNeo@web161019.mail.bf1.yahoo.com>
References: <1312307011.32126.YahooMailNeo@web161004.mail.bf1.yahoo.com>
	<CAKzWOxVoX9U6gfbikbj0npSZj5JKzzj94vjT8hrNDtZjxD-Uxw@mail.gmail.com>
	<1312313437.77623.YahooMailNeo@web161003.mail.bf1.yahoo.com>
	<CAKzWOxW40E4TSJ0p865iUiVLMd4PuROOx+UdN2KunBwrbsdW1Q@mail.gmail.com>
	<1312316397.65960.YahooMailNeo@web161007.mail.bf1.yahoo.com>
	<CAKzWOxUc=4rfTgnAxpRd7EFUA3F9Q8iBZMneNoTzyjFNeLCv3Q@mail.gmail.com>
	<1312317497.83886.YahooMailNeo@web161017.mail.bf1.yahoo.com>
	<CAKzWOxVPE13SgeOwPZE3mqOVfoxqzwkQN-8Fc3z_Pc8neYnV9A@mail.gmail.com>
	<1312325304.39236.YahooMailNeo@web161019.mail.bf1.yahoo.com>
Message-ID: <CAKzWOxVjWde-1isxbrbjOhFMrSwQUzsx7H6yHoXLedbe2b0N3Q@mail.gmail.com>

One last thing... put this as the last line of your script:

1;

In other words, end it with w "true" value. I saw that on the wiki, and as
you know, wikis are NEVER wrong. :P

-MC

On Tue, Aug 2, 2011 at 3:48 PM, Sam <lakersman2006 at yahoo.com> wrote:

> I made the corrections you suggested in my code, but still the same
> behavior. The bridge app seems to be in some type of internal loop and it
> does not end.
>
> ------------------------------
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Tuesday, August 2, 2011 3:14 PM
>
> *Subject:* Re: [Freeswitch-users] (no subject)
>
> Just a thought... try adding another "breakout" for your loop...
>
> #!/usr/bin/perl -w
>
> our $session;
>
> my $ringback_tone = "%(2000,4000,440,480)";    #US RINGBACK TONE
> my $end_call = 0;
> if ( $session->ready() && !$end_call )
> {
>     #set bridge settings
>     $session->execute("set", "ringback=$ringback_tone");    #set the
> ringback tone type
>     $session->execute("set", "instant_ringback=true");    #set to ring
> instantly
>     $session->execute("set", "ignore_early_media=false");    #set to NOT
> ignore early media
>     $session->execute("set", "call_timeout=20");        #only works if
> "ignore_early_media=true"
>     $session->execute("set", "bridge_answer_timeout=20");
>
>     $session->execute("set", "progress_timeout=15");
>     $session->execute("set", "continue_on_fail=false");
>     $session->execute("set", "hangup_after_bridge=true");
>     $session->execute("set", "bridge_pre_execute_bleg_app=info");
>     $session->execute("bridge", "sofia/gateway/carrier1/5214498052059");
>     $session->hangup();
>     $end_call = 1;
> }
>
> Also, I don't know if it was a typo or not, but you had this:
>
> if ($session->ready ())
>
> as opposed to
>
> if ($session->ready())
>
> Make sure that you fix that before testing further. :)
>
> -MC
>
> On Tue, Aug 2, 2011 at 1:38 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> In the pastebin, that was only 1 single call that was executed with the
> bridge app. The issue is that once the call channel is shutdown, a new
> channel gets created even though the call should be hung up. And I have
> noticed that this occurs when I leave out $session->answer(); in the
> beginning of the script. I was told by someone else on the mailing list to
> leave out $session->answer(); since the call has already been answered by
> the dialplan.
>
> ------------------------------
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Tuesday, August 2, 2011 1:29 PM
>
> *Subject:* Re: [Freeswitch-users] (no subject)
>
> In your pastebin of the call log, how many different calls was that? I saw
> like 4 incoming calls, however I couldn't specifically see the issue you
> were experiencing. Can you look at your log and see if you can isolate the
> approximate log lines where the issue is occurring?
>
> -MC
>
> On Tue, Aug 2, 2011 at 1:19 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> Here is my dialplan, pretty straight forward.
>
> <extension name="public_did">
>     <condition field="destination_number" expression="^(\d{11})|(\d{10})$">
>       <action application="perl" data="/usr/local/freeswitch/scripts/
> test2.pl"/>
>     </condition>
>   </extension>
>
> ------------------------------
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Tuesday, August 2, 2011 1:08 PM
> *Subject:* Re: [Freeswitch-users] (no subject)
>
>
>
> On Tue, Aug 2, 2011 at 12:30 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> MC,
>
> Here is the link to the console/ sip trace log
> http://pastebin.freeswitch.org/16945
>
>
> Not sure what is happening. Pastebin the dialplan extension(s) that handle
> this call. Something interesting must be going on.
>
> -MC
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/5b0fdf72/attachment-0001.html 

From lakersman2006 at yahoo.com  Wed Aug  3 03:26:23 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Tue, 2 Aug 2011 16:26:23 -0700 (PDT)
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <CAKzWOxVjWde-1isxbrbjOhFMrSwQUzsx7H6yHoXLedbe2b0N3Q@mail.gmail.com>
References: <1312307011.32126.YahooMailNeo@web161004.mail.bf1.yahoo.com>
	<CAKzWOxVoX9U6gfbikbj0npSZj5JKzzj94vjT8hrNDtZjxD-Uxw@mail.gmail.com>
	<1312313437.77623.YahooMailNeo@web161003.mail.bf1.yahoo.com>
	<CAKzWOxW40E4TSJ0p865iUiVLMd4PuROOx+UdN2KunBwrbsdW1Q@mail.gmail.com>
	<1312316397.65960.YahooMailNeo@web161007.mail.bf1.yahoo.com>
	<CAKzWOxUc=4rfTgnAxpRd7EFUA3F9Q8iBZMneNoTzyjFNeLCv3Q@mail.gmail.com>
	<1312317497.83886.YahooMailNeo@web161017.mail.bf1.yahoo.com>
	<CAKzWOxVPE13SgeOwPZE3mqOVfoxqzwkQN-8Fc3z_Pc8neYnV9A@mail.gmail.com>
	<1312325304.39236.YahooMailNeo@web161019.mail.bf1.yahoo.com>
	<CAKzWOxVjWde-1isxbrbjOhFMrSwQUzsx7H6yHoXLedbe2b0N3Q@mail.gmail.com>
Message-ID: <1312327583.59336.YahooMailNeo@web161019.mail.bf1.yahoo.com>

No, still same behavior.



________________________________
From: Michael Collins <msc at freeswitch.org>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, August 2, 2011 3:56 PM
Subject: Re: [Freeswitch-users] (no subject)


One last thing... put this as the last line of your script:

1;

In other words, end it with w "true" value. I saw that on the wiki, and as you know, wikis are NEVER wrong. :P

-MC



On Tue, Aug 2, 2011 at 3:48 PM, Sam <lakersman2006 at yahoo.com> wrote:

I made the corrections you suggested in my code, but still the same behavior. The bridge app seems to be in some type of internal loop and it does not end.
>
>
>
>
>________________________________
>From: Michael Collins <msc at freeswitch.org>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>Sent: Tuesday, August 2, 2011 3:14 PM
>
>Subject: Re: [Freeswitch-users] (no subject)
>
>
>
>Just a thought... try adding another "breakout" for your loop...
>
>
>#!/usr/bin/perl -w
>
>our $session;
>
>my $ringback_tone = "%(2000,4000,440,480)";??? #US RINGBACK TONE
>my $end_call = 0;
>if ( $session->ready() && !$end_call )?
>{??????
>??? #set bridge settings
>??? $session->execute("set", "ringback=$ringback_tone");??? #set the ringback tone type
>??? $session->execute("set", "instant_ringback=true");??? #set to ring instantly
>??? $session->execute("set", "ignore_early_media=false");??? #set to NOT ignore early media
>??? $session->execute("set", "call_timeout=20");??? ??? #only works if "ignore_early_media=true"
>??? $session->execute("set", "bridge_answer_timeout=20"); ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ????
>??? $session->execute("set", "progress_timeout=15");
>??? $session->execute("set", "continue_on_fail=false");
>??? $session->execute("set", "hangup_after_bridge=true");?
>??? $session->execute("set", "bridge_pre_execute_bleg_app=info");?
>??? $session->execute("bridge", "sofia/gateway/carrier1/5214498052059");
>??? $session->hangup();
>? ? $end_call = 1;?
>}
>
>
>Also, I don't know if it was a typo or not, but you had this:
>
>
>if ($session->ready ())
>
>
>as opposed to?
>
>
>if ($session->ready())
>
>
>Make sure that you fix that before testing further. :)
>
>
>-MC
>
>On Tue, Aug 2, 2011 at 1:38 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
>In the pastebin, that was only 1 single call that was executed with the bridge app. The issue is that once the call channel is shutdown, a new channel gets created even though the call should be hung up. And I have noticed that this occurs when I leave out $session->answer(); in the beginning of the script. I was told by someone else on the mailing list to leave out $session->answer(); since the call has already been answered by the dialplan.
>>
>>
>>
>>
>>________________________________
>>From: Michael Collins <msc at freeswitch.org>
>>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>Sent: Tuesday, August 2, 2011 1:29 PM
>>
>>Subject: Re: [Freeswitch-users] (no subject)
>>
>>
>>
>>In your pastebin of the call log, how many different calls was that? I saw like 4 incoming calls, however I couldn't specifically see the issue you were experiencing. Can you look at your log and see if you can isolate the approximate log lines where the issue is occurring?
>>
>>
>>-MC
>>
>>
>>On Tue, Aug 2, 2011 at 1:19 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>
>>Here is my dialplan, pretty straight forward.
>>>
>>>
>>><extension name="public_did">
>>>??? <condition field="destination_number" expression="^(\d{11})|(\d{10})$">
>>>????? <action application="perl" data="/usr/local/freeswitch/scripts/test2.pl"/>
>>>??? </condition>
>>>? </extension>
>>>
>>>
>>>
>>>________________________________
>>>From: Michael Collins <msc at freeswitch.org>
>>>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>Sent: Tuesday, August 2, 2011 1:08 PM
>>>Subject: Re: [Freeswitch-users] (no subject)
>>>
>>>
>>>
>>>
>>>
>>>
>>>On Tue, Aug 2, 2011 at 12:30 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>>
>>>MC,
>>>>
>>>>
>>>>Here is the link to the console/ sip trace log http://pastebin.freeswitch.org/16945
>>>>
>>>>
>>>
>>>
>>>Not sure what is happening. Pastebin the dialplan extension(s) that handle this call. Something interesting must be going on.
>>>
>>>
>>>-MC?
>>>_______________________________________________
>>>Join us at ClueCon 2011, Aug 9-11, Chicago
>>>http://www.cluecon.com 877-7-4ACLUE
>>>
>>>FreeSWITCH-users mailing list
>>>FreeSWITCH-users at lists.freeswitch.org
>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>http://www.freeswitch.org
>>>
>>>
>>>
>>>_______________________________________________
>>>Join us at ClueCon 2011, Aug 9-11, Chicago
>>>http://www.cluecon.com 877-7-4ACLUE
>>>
>>>FreeSWITCH-users mailing list
>>>FreeSWITCH-users at lists.freeswitch.org
>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>http://www.freeswitch.org
>>>
>>>
>>
>>_______________________________________________
>>Join us at ClueCon 2011, Aug 9-11, Chicago
>>http://www.cluecon.com 877-7-4ACLUE
>>
>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>>
>>
>>_______________________________________________
>>Join us at ClueCon 2011, Aug 9-11, Chicago
>>http://www.cluecon.com 877-7-4ACLUE
>>
>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>>
>
>_______________________________________________
>Join us at ClueCon 2011, Aug 9-11, Chicago
>http://www.cluecon.com 877-7-4ACLUE
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>
>
>_______________________________________________
>Join us at ClueCon 2011, Aug 9-11, Chicago
>http://www.cluecon.com 877-7-4ACLUE
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>

_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/7867b268/attachment.html 

From cmcureau at gmail.com  Wed Aug  3 03:28:22 2011
From: cmcureau at gmail.com (Chris Cureau)
Date: Tue, 2 Aug 2011 18:28:22 -0500
Subject: [Freeswitch-users] Help with choppy audio after attended
	transfer
In-Reply-To: <CAKbxfG_XUjkLJTcMFP4sHbp+KYy6fM8cTOuzyhyMRHh-DbVp=A@mail.gmail.com>
References: <CAL0fNgWOuhAeMgLS3UsD4tSqdDd2FGSqXw4T-H9AnfVLBGOr5Q@mail.gmail.com>
	<CAKbxfG-UOWoisXbbZaEH0Ruzm7=e2oobyvwqM1vJ-pQFtDuOow@mail.gmail.com>
	<CAL0fNgVu9EP4jFO9XzYw+SbZKiz8W7rN03A_6TWhOi-9cJf9bA@mail.gmail.com>
	<CAKbxfG_XUjkLJTcMFP4sHbp+KYy6fM8cTOuzyhyMRHh-DbVp=A@mail.gmail.com>
Message-ID: <CAL0fNgW730m1cKFrCRsA=ACeq=6nOWYWWaubJAeGUaXGk36rZg@mail.gmail.com>

Latest git doesn't help unfortunately.

More research is beginning to confirm its a codec issue.  When I limit the
phone to codes that use 20ms packetization, I hear no incoming audio on
outgoing calls.  Incoming calls are okay.  When using All for codes (and the
30ms packetization) I get the choppy voice when leaving MOH.



On Mon, Aug 1, 2011 at 8:34 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> can you do make current and try latest git?
>
>
> On Mon, Aug 1, 2011 at 10:46 AM, Chris Cureau <cmcureau at gmail.com> wrote:
> > Anthony,
> >
> > Thanks for answering...and sorry for the delay.  I've already checked all
> of
> > the ptime settings I can, and all phones plus freeswitch are set to use
> 20ms
> > packetization.  I've even set "scrooge" in the codec negotiation, but I
> keep
> > running into this issue.  I've updated my post with "sofia global
> siptrace
> > on".
> >
> > I am assuming that the ptime issue happens around line 2462
> > (http://pastebin.freeswitch.org/16935)
> >
> > 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:4711 Audio Codec Compare
> > [PCMU:0:8000:20:64000]/[PCMU:0:8000:30:64000]
> > 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:2753 Already using PCMU
> > 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:4819 Set 2833 dtmf send
> > payload to 101
> > 2011-08-01 09:12:37.332892 [DEBUG] sofia.c:5599 Processing updated SDP
> > 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:3042 Audio params are
> > unchanged for sofia/internal/sip:1003 at 10.0.1.205:5060.
> > 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:3052
> > sofia/internal/sip:1003 at 10.0.1.205:5060 Setting audio receive payload in
> > Re-INVITE to 0
> >
> > Could this be an issue with the Aastra'a firmware?  Or maybe the MOH is
> > being processed at 30ms instead of 20ms, and the negotiation is not
> updated
> > somehow?
> >
> > I don't mean to sound ignorant, but I'm really at a loss here...and
> thanks
> > again for any help!
> >
> > Cheers,
> > Chris
> >
> > On Fri, Jul 29, 2011 at 10:44 AM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> probably ptime related thing.
> >> you should have included the sip trace "sofia global siptrace on"
> >>
> >>
> >> On Fri, Jul 29, 2011 at 12:28 AM, Chris Cureau <cmcureau at gmail.com>
> wrote:
> >> > I'm having some issues with extremely choppy audio after a call has
> been
> >> > sent to another extension via an automated transfer.  The audio is
> great
> >> > when the call is answered.  Shortly after, the transfer button is
> >> > pressed
> >> > and the incoming call hears music on hold.  The music on hold is sent
> to
> >> > the
> >> > caller sounds fine as does the conversation between extensions.  When
> >> > the
> >> > transfer is completed, the caller hears what sounds like someone
> >> > speaking
> >> > through a fan (though slower) but incoming audio sounds fine.
> >> >
> >> > Since it's such a large log, I posted it to the FreeSWITCH pastebin:
> >> > http://pastebin.freeswitch.org/16911
> >> >
> >> > I'm thinking that it has something to do with the transition from MOH
> to
> >> > the
> >> > internal extension, but I can't figure out what is happening.
> >> >
> >> > Any ideas?
> >> >
> >> > _______________________________________________
> >> > Join us at ClueCon 2011, Aug 9-11, Chicago
> >> > http://www.cluecon.com 877-7-4ACLUE
> >> >
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> 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
> >> googletalk:conf+888 at conference.freeswitch.org
> >> pstn:+19193869900
> >>
> >> _______________________________________________
> >> Join us at ClueCon 2011, Aug 9-11, Chicago
> >> http://www.cluecon.com 877-7-4ACLUE
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/7e1283f0/attachment-0001.html 

From msc at freeswitch.org  Wed Aug  3 03:41:06 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 2 Aug 2011 16:41:06 -0700
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <1312327583.59336.YahooMailNeo@web161019.mail.bf1.yahoo.com>
References: <1312307011.32126.YahooMailNeo@web161004.mail.bf1.yahoo.com>
	<CAKzWOxVoX9U6gfbikbj0npSZj5JKzzj94vjT8hrNDtZjxD-Uxw@mail.gmail.com>
	<1312313437.77623.YahooMailNeo@web161003.mail.bf1.yahoo.com>
	<CAKzWOxW40E4TSJ0p865iUiVLMd4PuROOx+UdN2KunBwrbsdW1Q@mail.gmail.com>
	<1312316397.65960.YahooMailNeo@web161007.mail.bf1.yahoo.com>
	<CAKzWOxUc=4rfTgnAxpRd7EFUA3F9Q8iBZMneNoTzyjFNeLCv3Q@mail.gmail.com>
	<1312317497.83886.YahooMailNeo@web161017.mail.bf1.yahoo.com>
	<CAKzWOxVPE13SgeOwPZE3mqOVfoxqzwkQN-8Fc3z_Pc8neYnV9A@mail.gmail.com>
	<1312325304.39236.YahooMailNeo@web161019.mail.bf1.yahoo.com>
	<CAKzWOxVjWde-1isxbrbjOhFMrSwQUzsx7H6yHoXLedbe2b0N3Q@mail.gmail.com>
	<1312327583.59336.YahooMailNeo@web161019.mail.bf1.yahoo.com>
Message-ID: <CAKzWOxV09m=+rzZueogrs8poLdto9PgzOCRynO0SaBb8Y_0N-Q@mail.gmail.com>

okay, i'm out of ideas, other than to ask... why are you doing this in a
script and not the dialplan?
-MC

On Tue, Aug 2, 2011 at 4:26 PM, Sam <lakersman2006 at yahoo.com> wrote:

> No, still same behavior.
>
> ------------------------------
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Tuesday, August 2, 2011 3:56 PM
>
> *Subject:* Re: [Freeswitch-users] (no subject)
>
> One last thing... put this as the last line of your script:
>
> 1;
>
> In other words, end it with w "true" value. I saw that on the wiki, and as
> you know, wikis are NEVER wrong. :P
>
> -MC
>
> On Tue, Aug 2, 2011 at 3:48 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> I made the corrections you suggested in my code, but still the same
> behavior. The bridge app seems to be in some type of internal loop and it
> does not end.
>
> ------------------------------
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Tuesday, August 2, 2011 3:14 PM
>
> *Subject:* Re: [Freeswitch-users] (no subject)
>
> Just a thought... try adding another "breakout" for your loop...
>
> #!/usr/bin/perl -w
>
> our $session;
>
> my $ringback_tone = "%(2000,4000,440,480)";    #US RINGBACK TONE
> my $end_call = 0;
> if ( $session->ready() && !$end_call )
> {
>     #set bridge settings
>     $session->execute("set", "ringback=$ringback_tone");    #set the
> ringback tone type
>     $session->execute("set", "instant_ringback=true");    #set to ring
> instantly
>     $session->execute("set", "ignore_early_media=false");    #set to NOT
> ignore early media
>     $session->execute("set", "call_timeout=20");        #only works if
> "ignore_early_media=true"
>     $session->execute("set", "bridge_answer_timeout=20");
>
>     $session->execute("set", "progress_timeout=15");
>     $session->execute("set", "continue_on_fail=false");
>     $session->execute("set", "hangup_after_bridge=true");
>     $session->execute("set", "bridge_pre_execute_bleg_app=info");
>     $session->execute("bridge", "sofia/gateway/carrier1/5214498052059");
>     $session->hangup();
>     $end_call = 1;
> }
>
> Also, I don't know if it was a typo or not, but you had this:
>
> if ($session->ready ())
>
> as opposed to
>
> if ($session->ready())
>
> Make sure that you fix that before testing further. :)
>
> -MC
>
> On Tue, Aug 2, 2011 at 1:38 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> In the pastebin, that was only 1 single call that was executed with the
> bridge app. The issue is that once the call channel is shutdown, a new
> channel gets created even though the call should be hung up. And I have
> noticed that this occurs when I leave out $session->answer(); in the
> beginning of the script. I was told by someone else on the mailing list to
> leave out $session->answer(); since the call has already been answered by
> the dialplan.
>
> ------------------------------
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Tuesday, August 2, 2011 1:29 PM
>
> *Subject:* Re: [Freeswitch-users] (no subject)
>
> In your pastebin of the call log, how many different calls was that? I saw
> like 4 incoming calls, however I couldn't specifically see the issue you
> were experiencing. Can you look at your log and see if you can isolate the
> approximate log lines where the issue is occurring?
>
> -MC
>
> On Tue, Aug 2, 2011 at 1:19 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> Here is my dialplan, pretty straight forward.
>
> <extension name="public_did">
>     <condition field="destination_number" expression="^(\d{11})|(\d{10})$">
>       <action application="perl" data="/usr/local/freeswitch/scripts/
> test2.pl"/>
>     </condition>
>   </extension>
>
> ------------------------------
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Tuesday, August 2, 2011 1:08 PM
> *Subject:* Re: [Freeswitch-users] (no subject)
>
>
>
> On Tue, Aug 2, 2011 at 12:30 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> MC,
>
> Here is the link to the console/ sip trace log
> http://pastebin.freeswitch.org/16945
>
>
> Not sure what is happening. Pastebin the dialplan extension(s) that handle
> this call. Something interesting must be going on.
>
> -MC
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/98b93209/attachment.html 

From moises.silva at gmail.com  Wed Aug  3 06:36:02 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Tue, 2 Aug 2011 22:36:02 -0400
Subject: [Freeswitch-users] freetdm error message
In-Reply-To: <CAABykDrOSPH4krH45EjxURkXOFJAGMV6q0T3CehGEQcgKzPfxQ@mail.gmail.com>
References: <CAABykDrOSPH4krH45EjxURkXOFJAGMV6q0T3CehGEQcgKzPfxQ@mail.gmail.com>
Message-ID: <CAA4nhyC7s2qzd6DDypzN0xVPdVmeWompgncFcyOZhgF6FZQqyQ@mail.gmail.com>

On Tue, Aug 2, 2011 at 7:45 AM, ovvenkat <ovvenkatesan at gmail.com> wrote:
> Hi to all,
>
> In my "fs_cli"? I am getting bellow message very frequently.
> Can you any one help me where is the error?
>
> 2011-08-02 17:11:33.028452 [INFO] ftmod_sangoma_isdn_stack_rcv.c:995
> sng_isdn->s1: Invalid Q.921/Q.931 frame - ignoring len:1
> 2011-08-02 17:11:33.048455 [INFO] ftmod_sangoma_isdn_stack_rcv.c:995
> sng_isdn->s1: Invalid Q.921/Q.931 frame - ignoring len:1
> 2011-08-02 17:11:33.048455 [INFO] ftmod_sangoma_isdn_stack_rcv.c:995
> sng_isdn->s1: Invalid Q.921/Q.931 frame - ignoring len:1
> 2011-08-02 17:11:33.068455 [INFO] ftmod_sangoma_isdn_stack_rcv.c:995
> sng_isdn->s1: Invalid Q.921/Q.931 frame - ignoring len:1
> 2011-08-02 17:11:33.068455 [INFO] ftmod_sangoma_isdn_stack_rcv.c:995
> sng_isdn->s1: Invalid Q.921/Q.931 frame - ignoring len:1
> 2011-08-02 17:11:33.068455 [INFO] ftmod_sangoma_isdn_stack_rcv.c:995
> sng_isdn->s1: Invalid Q.921/Q.931 frame - ignoring len:1

That really looks like some sort of garbage on the line (although is
not strictly only garbage since they seem to be valid HDLC frames).

How often do you get this? the word "frequently" means nothing from an
engineering point of view. Every minute you get a burst of these? or
is it a constant flow? can you receive/place calls despite this
message?

Try taking a pcap file when you're receiving those messages:
http://wiki.sangoma.com/wanpipe-wireshark-pcap-pri-bri-wan-t1-e1-tracing

Moises Silva
Senior Software Engineer, Software Development Manager
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
L3R 9R6 Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com


From bino at indoakses-online.com  Wed Aug  3 08:09:45 2011
From: bino at indoakses-online.com (bino oetomo)
Date: Wed, 03 Aug 2011 11:09:45 +0700
Subject: [Freeswitch-users] How to for video call ?
In-Reply-To: <CALXCt0ouywkiXe-cit_q4=JmVCk9nwMdOq7Z-EM5ZG-6kcHHmw@mail.gmail.com>
References: <4E3655DF.4060207@indoakses-online.com>	<CAHKqMb60DcNm8P8FztXyJOhmf0h6wQEMgZw_ami_vbWho2CP8Q@mail.gmail.com>	<4E376A92.9060201@indoakses-online.com>
	<CALXCt0ouywkiXe-cit_q4=JmVCk9nwMdOq7Z-EM5ZG-6kcHHmw@mail.gmail.com>
Message-ID: <4E38CA09.8050804@indoakses-online.com>

On 08/02/2011 12:45 PM, Giovanni Maruzzelli wrote:
> Use extensions 1001 to 1020 (those are configured in the default dialplan).
>
> -giovanni
>

Dear Giovanni, Nandy , and ALL ..

I really appreciate your fast response.
Ok .. I made a clean re-install ... and now it work.
It's a plain freeswitch install .. no management GUI.

Since it work .. I made a complete recursive copy of "conf" directory

Next, I'll try to re install fusionPBX on top of it.
Well .. my target is Freeswitch system with:
- Single but humanizedc domain, i.e : use "mycompany" instead of 
"192.168.10.238" as domain name.
- Postgres backend
- simple management GUI

Is there any simpler alternative to FusionPBX ?

I'll give the report as soon as it's installed

Sincerely
-bino-


From fieldpeak at gmail.com  Wed Aug  3 08:32:39 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Wed, 3 Aug 2011 12:32:39 +0800
Subject: [Freeswitch-users] Mod_rad_auth issue for FS working with
 FreeRadius server
In-Reply-To: <CAJoKKdXSwF02S9tBkorGwOjj57AtkRpU0c177qXASEpodKVk=w@mail.gmail.com>
References: <CAOTmbMzvd5OHL8z=_8vu6Orp2_prGxaw1gmXYCM2=2+pgzs_WA@mail.gmail.com>
	<CAJoKKdX01Qc8ibs25PUELi8GrzMDNOrHb_bQvYdq3ZqA4XLy4w@mail.gmail.com>
	<CAOTmbMxL3pLOuw655fyT5VzAON_N0PrDN+-i-++=oZoYU6+Fmg@mail.gmail.com>
	<CAOTmbMyvyDA-CRpaMC=WRGVkiyTG=Rs+n4hyeQWdyTZpqysSzw@mail.gmail.com>
	<CAJoKKdXSwF02S9tBkorGwOjj57AtkRpU0c177qXASEpodKVk=w@mail.gmail.com>
Message-ID: <CAOTmbMzu2qDYh2T24JSpCYn1xH2J4iGiK7UkUB4Kq=-KdcDWGQ@mail.gmail.com>

Hi Tihomir,

Sorry, i missed your mail in gmail before, just now saw it, and after using
your dictionary.all, the dictionary issue was resolved, very appreciated for
your kindly help! however, it did not fully functional yet,

Attached are configuration files that i used, when i dial 601 to trigger to
auth, the freeradius server shows log below, the supecious log is the value
User-Password, it should be '1111' that i've set in the mysql db of
freeradisu server for the user 1001 .

i searched in google, for "known good" password issue, i suggest change
user-password to cleartext-password, however, i did not find where it is.
and also the Auth-Type, where to configure it...

Freeradius server log:

rad_recv: Access-Request packet from host 127.0.0.1 port 52684, id=49,
length=111
        User-Name = "1001"
        User-Password = "?\210\365@\263\t\306\343\243iT?\311C\t\002"
        Called-Station-Id = "888"
        h323-conf-id = "749d2b5a-16ad-48e4-af58-24011949d1b5"
        Calling-Station-Id = "1001"
        NAS-Port = 0
        NAS-IP-Address = 127.0.0.1
# Executing section authorize from file
/usr/local/etc/raddb/sites-enabled/default
+- entering group authorize {...}
++[preprocess] returns ok
[auth_log]      expand:
/usr/local/var/log/radius/radacct/%{Client-IP-Address}/auth-detail-%Y%m%d ->
/usr/local/var/log/radius/radacct/127.0.0.1/auth-detail-20110803
[auth_log]
/usr/local/var/log/radius/radacct/%{Client-IP-Address}/auth-detail-%Y%m%d
expands to /usr/local/var/log/radius/radacct/127.0.0.1/auth-detail-20110803
[auth_log]      expand: %t -> Wed Aug  3 12:06:33 2011
++[auth_log] returns ok
++[chap] returns noop
++[mschap] returns noop
++[digest] returns noop
[suffix] No '@' in User-Name = "1001", looking up realm NULL
[suffix] No such realm "NULL"
++[suffix] returns noop
[eap] No EAP-Message, not doing EAP
++[eap] returns noop
++[unix] returns notfound
++[files] returns noop
[sql]   expand: %{User-Name} -> 1001
[sql] sql_set_user escaped user --> '1001'
rlm_sql (sql): Reserving sql socket id: 4
[sql]   expand: SELECT id, username, attribute, value, op           FROM
radcheck           WHERE username = '%{SQL-User-Name}'           ORDER BY id
-> SELECT id, username, attribute, value, op           FROM
radcheck           WHERE username = '1001'           ORDER BY id
[sql]   expand: SELECT groupname           FROM radusergroup           WHERE
username = '%{SQL-User-Name}'           ORDER BY priority -> SELECT
groupname           FROM radusergroup           WHERE username =
'1001'           ORDER BY priority
rlm_sql (sql): Released sql socket id: 4
[sql] User 1001 not found
++[sql] returns notfound
++[expiration] returns noop
++[logintime] returns noop
[pap] WARNING! No "known good" password found for the user.  Authentication
may fail because of this.
++[pap] returns noop
ERROR: No authenticate method (Auth-Type) found for the request: Rejecting
the user
Failed to authenticate the user.
  WARNING: Unprintable characters in the password.        Double-check the
shared secret on the server and the NAS!
Using Post-Auth-Type Reject
# Executing group from file /usr/local/etc/raddb/sites-enabled/default
+- entering group REJECT {...}
[attr_filter.access_reject]     expand: %{User-Name} -> 1001
 attr_filter: Matched entry DEFAULT at line 11
++[attr_filter.access_reject] returns updated
Delaying reject of request 8 for 1 seconds
Going to the next request
Waking up in 0.9 seconds.
Sending delayed reject for request 8
Sending Access-Reject of id 49 to 127.0.0.1 port 52684
Waking up in 4.9 seconds.
Cleaning up request 8 ID 49 with timestamp +7674
Ready to process requests.
WARNING! No "known good" password found for the user

Regards,
Charles

2011/8/3 Tihomir Culjaga <tculjaga at gmail.com>

> did u use the dictionary i have attached ?
>
>
> On Tue, Aug 2, 2011 at 10:08 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>
>> i tried change to 'h323-conf-id' to 'h323-call-origin' in
>> 02_unitest_rad-ANI-auth.xml, rad_auth.conf.xml, however, it still prompt
>> '[ERR] mod_rad_auth.c:428 Unknown attribute: key:h323-conf-id, not found
>> in dictionary', so where the mod_rad_auth read out the 'h323-conf-id'? very
>> very strange, which dictionary it was using...
>>
>> Regards,
>> Charles
>>
>>
>> 2011/8/2 fieldpeak <fieldpeak at gmail.com>
>>
>>> Hi Tihomir,
>>>
>>> Finally the answer coming, i see the hope, thanks for your reply, :)
>>>
>>> As your advise, i only use one attribute(h323-conf-id) in my dialplan,
>>> and only one attribute(h323-conf-id) in rad_auth.conf.xml, and using the
>>> attached dictionary (from ciso) which contains this attribute, however, it
>>> still prompt 'unknown attribute', so i suspected if it was reading
>>> /usr/local/etc/radiusclient/dictionary, so i copy the same dictionary to
>>> /usr/local/freeswitch/radius/, it did not any help at all... very strange...
>>>
>>> Log:
>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set default_realm
>>> := .
>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set radius_timeout
>>> := 3.
>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set radius_retries
>>> := 2.
>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set radius_deadtime
>>> := 0.
>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set bindaddr := *.
>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:371 ... radius:
>>> User-Name: 38516060333
>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:380 ... radius:
>>> User-Password: 003282
>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:396 ... radius:
>>> Called-station-Id: 16094191500
>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:413 Handle attribute:
>>> h323-conf-id
>>> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:428 Unknown attribute:
>>> key:h323-conf-id, not found in dictionary
>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:538 abort sending
>>> radius packet.
>>> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:546 An error occured
>>> during RADIUS Authentication(RC=-1)
>>> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:702 An error occured
>>> during radius authorization.
>>>
>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  AUTH_RESULT=)
>>>
>>>
>>>
>>>   <extension name="unitest_rad-ANI-auth">
>>>     <condition field="destination_number" expression="^601$">
>>>       <!-- <action application="log" data="INFO  Before Auth "/> -->
>>>
>>>       <action inline="true" application="set" data="CALLID=h323-conf-id
>>> =${uuid}"/>
>>>
>>>       <action inline="true" application="set" data="USERNAME=1001"/>
>>>       <action inline="true" application="set" data="PASSWD=1111"/>
>>>
>>>
>>>       <action application="sleep" data="2000"/>
>>>       <action application="auth_function" data="in ${DIALED_NUMBER}, in
>>> ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>
>>>
>>>     </condition>
>>>   </extension>
>>>
>>>
>>>
>>> <configuration name="rad_auth.conf" description="radius authentification
>>> module">
>>>   <settings>
>>>
>>>   </settings>
>>>
>>>   <client>
>>>     <param name="authserver" value="127.0.0.1:1812:gateway"/>
>>>     <param name="dictionary"
>>> value="/usr/local/etc/radiusclient/dictionary"/>
>>>     <param name="seqfile" value="/var/run/radius.seq"/>
>>>     <param name="mapfile"
>>> value="/usr/local/etc/radiusclient/port-id-map"/>
>>>     <param name="default_realm" value=""/>
>>>     <param name="radius_timeout" value="3"/>
>>>     <param name="radius_retries" value="2"/>
>>>     <param name="radius_deadtime" value="0"/>
>>>     <param name="bindaddr" value="*"/>
>>>   </client>
>>>
>>>   <vsas>
>>>
>>>
>>>     <param name="h323-conf-id" id="24" value="CALLID" pec="9" expr="1"
>>> direction="in"/>
>>>
>>>   </vsas>
>>>  </configuration>
>>>
>>>
>>>
>>> 2011/8/2 Tihomir Culjaga <tculjaga at gmail.com>
>>>
>>>> hi,
>>>>
>>>> dictionary.all is just the name of a file containing all attributes i
>>>> needed at that time.
>>>>
>>>> you can include other dictionaries by putting #INCLUDE <pathname> at the
>>>> end of the dictionary file you reference in rad_auth.conf.xml.
>>>> if the INCLUDE doesn't work, just append dictionary.cisco to your
>>>> dictionary file... and make your own file.
>>>>
>>>>
>>>> check inline comments down below...
>>>>
>>>>
>>>> T.
>>>>
>>>>
>>>> On Sun, Jul 31, 2011 at 10:46 AM, fieldpeak <fieldpeak at gmail.com>wrote:
>>>>
>>>>> Hello Gurus,
>>>>>
>>>>> i met a issue when using
>>>>> mod_rad_auth(http://wiki.freeswitch.org/wiki/Mod_rad_auth) to works
>>>>> with freeradius server+mysql for AAA, the details is below, Could
>>>>> anyone give any hints, Thanks in advance.
>>>>>
>>>>> i setup a dial plan "unitest_rad-ANI-auth" as wiki above, however,
>>>>> when i dialed 601 to trigger the dial plan, the console show errors,
>>>>> it looks "h323-conf-id" is not in the directory, then i tried to add
>>>>> this attribute to the dictionary, however, it does not help, in the
>>>>> wiki, it mentioned the rad_auth.conf.xml contains <param
>>>>> name="dictionary"
>>>>> value="/usr/local/etc/radiusclient/dictionary.all"/>, however i did
>>>>> not find the file "dictionary.all" at that directory, so i use
>>>>> dictionary. BTW, the freeradius server + mysql works well.
>>>>>
>>>>
>>>> i just appended the information needed into dictionary.all file...
>>>> (vendor and attribute definition).
>>>>
>>>>
>>>>
>>>>>
>>>>> console errors:
>>>>>
>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 auth_function(in , in
>>>>> 38516060333, in 003282, out AUTH_RESULT)
>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:301 allocate initial
>>>>> structure.
>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:313 initialzed
>>>>> configuration.
>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set authserver
>>>>> := 127.0.0.1:1812:gateway.
>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set dictionary
>>>>> := /usr/local/etc/radiusclient/dictionary.
>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set seqfile :=
>>>>> /var/run/radius.seq.
>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set mapfile :=
>>>>> /usr/local/etc/radiusclient/port-id-map.
>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set default_realm
>>>>> := .
>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>> radius_timeout := 3.
>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>> radius_retries := 2.
>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>> radius_deadtime := 0.
>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set bindaddr :=
>>>>> *.
>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:371 ... radius:
>>>>> User-Name: 38516060333
>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:380 ... radius:
>>>>> User-Password: 003282
>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:391 ... radius:
>>>>> Called-station-Id is empty, ignoring...
>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:413 Handle
>>>>> attribute: h323-conf-id
>>>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:428 Unknown attribute:
>>>>> key:h323-conf-id, not found in dictionary
>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:538 abort sending
>>>>> radius packet.
>>>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:546 An error occured
>>>>> during RADIUS Authentication(RC=-1)
>>>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:702 An error occured
>>>>> during radius authorization.
>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  AUTH_RESULT=)
>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  AUTH_RESULT=
>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  billing_model=)
>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  billing_model=
>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  credit_amount=)
>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  credit_amount=
>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  currency=)
>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  currency=
>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  preffered_lang=)
>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  preffered_lang=
>>>>>
>>>>> added below in the dictionary(/usr/local/etc/radiusclient/dictionary):
>>>>>
>>>>> ATTRIBUTE       h323-conf-id            1008    string
>>>>>
>>>>
>>>> you need the vendor definition as well
>>>>
>>>>
>>>>>
>>>>>
>>>>> dial plan:
>>>>> <extension name="unitest_rad-ANI-auth">
>>>>>    <condition field="destination_number" expression="^601$">
>>>>>      <action application="log" data="INFO  Before Auth "/>
>>>>>
>>>>>      <action inline="true" application="set"
>>>>> data="CALLID=h323-conf-id=${uuid}"/>
>>>>>      <action inline="true" application="set"
>>>>> data="SERVICENUM=h323-prompt-id=${destination_number}"/>
>>>>>      <action inline="true" application="set"
>>>>> data="TRANSACTIONID=h323-ivr-out=transactionID:1234"/>
>>>>>  <!--      <action inline="true" application="set"
>>>>> data="CALLINGNUMBER=${caller_id_number}"/> -->
>>>>>      <action inline="true" application="set"
>>>>> data="CALLINGNUMBER=38516060333"/>
>>>>>      <action inline="true" application="set"
>>>>> data="USERNAME=38516060333"/>
>>>>>  <!--      <action inline="true" application="set"
>>>>> data="USERNAME=209354"/> -->
>>>>>      <action inline="true" application="set" data="PASSWD=003282"/>
>>>>>  <!--      <action inline="true" application="set"
>>>>> data="DIALED_NUMBER=16094191500"/>  -->
>>>>>
>>>>>      <action application="sleep" data="2000"/>
>>>>>      <action application="auth_function" data="in ${DIALED_NUMBER},
>>>>> in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>
>>>>>
>>>>>
>>>>>      <action application="log" data="INFO
>>>>>  AUTH_RESULT=${AUTH_RESULT}"/>
>>>>>      <action application="log" data="INFO
>>>>>  billing_model=${billing_model}"/>
>>>>>      <action application="log" data="INFO
>>>>>  credit_amount=${credit_amount}"/>
>>>>>      <action application="log" data="INFO  currency=${currency}"/>
>>>>>      <action application="log" data="INFO
>>>>>  preffered_lang=${preffered_lang}"/>
>>>>>      <action application="log" data="INFO
>>>>>  credit_time=${credit_time}"/>
>>>>>      <action application="log" data="INFO
>>>>> h323_ivr_duration=${h323_ivr_duration}"/>
>>>>>      <action application="log" data="INFO
>>>>>  return_code=${return_code}"/>
>>>>>      <!-- <action application="execute_extension" data="AUTH XML
>>>>> default"/> -->
>>>>>    </condition>
>>>>>  </extension>
>>>>>
>>>>>  radius_cdr.conf.xml:
>>>>>  <configuration name="radius_cdr.conf" description="RADIUS CDR
>>>>> Configuration">
>>>>>
>>>>>        <settings>
>>>>>
>>>>>                <!-- location of the radius dictionary files -->
>>>>>
>>>>>                <param name="dictionary"
>>>>> value="/usr/local/freeswitch/conf/radius/dictionary"/>
>>>>>
>>>>>
>>>> your dictionary file need to contain all the attributes you are trying
>>>> to use or to include other dictionaries (In this case dictionary.cisco) from
>>>> the dictionary file you are referencing here.
>>>>
>>>>
>>>>>                <!-- number of retries for each server -->
>>>>>
>>>>>                <param name="radius_retries" value="3"/>
>>>>>
>>>>>                <!-- number of seconds to wait between retries -->
>>>>>
>>>>>                <param name="radius_timeout" value="5"/>
>>>>>
>>>>>                <!-- accounting servers, up to 8 allowed -->
>>>>>
>>>>>                <!-- value is "host:port:secret", port is optional -->
>>>>>
>>>>>                <!-- use IP ADDRESSES, not hostnames -->
>>>>>
>>>>>                <param name="acct_server" value="127.0.0.1:1813
>>>>> :testing123"/>
>>>>>
>>>>>
>>>>>        </settings>
>>>>>
>>>>> </configuration>
>>>>>
>>>>>  the FS version:
>>>>>  FreeSWITCH Version 1.0.head (git-492bc6b 2011-07-23 12-53-04 -0400)
>>>>>
>>>>>  Regards,
>>>>>  Charles
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>>
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110803/b381a3f6/attachment-0001.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: 02_unitest_rad-ANI-auth.xml
Type: text/xml
Size: 1934 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110803/b381a3f6/attachment-0002.xml 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: rad_auth.conf.xml
Type: text/xml
Size: 2779 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110803/b381a3f6/attachment-0003.xml 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: mysql_user_info.png
Type: image/png
Size: 114727 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110803/b381a3f6/attachment-0001.png 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: radiusclient.conf
Type: application/octet-stream
Size: 3302 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110803/b381a3f6/attachment-0001.obj 

From dhairya.blogs at gmail.com  Wed Aug  3 09:10:23 2011
From: dhairya.blogs at gmail.com (Dhairya Vora)
Date: Wed, 3 Aug 2011 10:40:23 +0530
Subject: [Freeswitch-users] Calls not established : See the error details
Message-ID: <CA+P_fE8Fd5=2c4b48npp-27aQGGJN1vhFRE8FTvvgp+4xx0CXw@mail.gmail.com>

I don't understand why two users (registered directly to freeswitch) are
unable to connect each other?


When a registered user 1001 calls registered user 1002, call fails giving
this error
************************************************************************************************************************************************************
freeswitch at localhost.localdomain> 2011-08-03 10:07:01.941089 [WARNING]
sofia_reg.c:1337 SIP auth challenge (INVITE) on sofia profile 'internal' for
[1002 at 172.16.10.211] from ip 172.16.10.248
2011-08-03 10:07:01.941089 [NOTICE] switch_channel.c:897 New Channel
sofia/internal/1000 at 172.16.10.211 [0bf31c06-8379-4831-88d7-a7df3c4ecd49]
2011-08-03 10:07:01.941089 [INFO] mod_dialplan_xml.c:336 Processing 1000
<1000>->1002 in context default
2011-08-03 10:07:01.941089 [INFO] switch_core_session.c:1281 sofia/internal/
1000 at 172.16.10.211 setting session heartbeat to 60 second(s).
2011-08-03 10:07:01.941089 [ERR] mod_event_socket.c:457 Socket Error!
2011-08-03 10:07:01.941089 [NOTICE] switch_core_state_machine.c:189
sofia/internal/1000 at 172.16.10.211 has executed the last dialplan
instruction, hanging up.
2011-08-03 10:07:01.941089 [NOTICE] switch_core_state_machine.c:191 Hangup
sofia/internal/1000 at 172.16.10.211 [CS_EXECUTE] [NORMAL_CLEARING]
2011-08-03 10:07:01.961094 [NOTICE] switch_core_session.c:1347 Session 25
(sofia/internal/1000 at 172.16.10.211) Ended
2011-08-03 10:07:01.961094 [NOTICE] switch_core_session.c:1349 Close Channel
sofia/internal/1000 at 172.16.10.211 [CS_DESTROY]
************************************************************************************************************************************************************



When I make an outbound call to my mobile, calls are not going through the
custom gateway
************************************************************************************************************************************************************
freeswitch at localhost.localdomain> 2011-08-03 10:03:43.608234 [WARNING]
sofia_reg.c:1337 SIP auth challenge (INVITE) on sofia profile 'internal' for
[00919876543210 at 172.16.10.211] from ip 172.16.10.213
2011-08-03 10:03:43.728232 [NOTICE] switch_channel.c:897 New Channel
sofia/internal/1002 at 172.16.10.211 [334906be-9cb0-44f2-9660-f9377c2ecfba]
2011-08-03 10:03:43.728232 [INFO] mod_dialplan_xml.c:336 Processing 1002
<1002>->00919876543210 in context default
2011-08-03 10:03:43.728232 [INFO] switch_core_session.c:1281 sofia/internal/
1002 at 172.16.10.211 setting session heartbeat to 60 second(s).
2011-08-03 10:03:43.728232 [ERR] mod_event_socket.c:457 Socket Error!
2011-08-03 10:03:43.728232 [NOTICE] switch_core_state_machine.c:189
sofia/internal/1002 at 172.16.10.211 has executed the last dialplan
instruction, hanging up.
2011-08-03 10:03:43.728232 [NOTICE] switch_core_state_machine.c:191 Hangup
sofia/internal/1002 at 172.16.10.211 [CS_EXECUTE] [NORMAL_CLEARING]
2011-08-03 10:03:43.728232 [NOTICE] switch_core_session.c:1347 Session 24
(sofia/internal/1002 at 172.16.10.211) Ended
2011-08-03 10:03:43.728232 [NOTICE] switch_core_session.c:1349 Close Channel
sofia/internal/1002 at 172.16.10.211 [CS_DESTROY]
************************************************************************************************************************************************************
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110803/055f8471/attachment.html 

From msc at freeswitch.org  Wed Aug  3 09:48:50 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 2 Aug 2011 22:48:50 -0700
Subject: [Freeswitch-users] conference caller id number&name
In-Reply-To: <4E13710A.5060108@gosilverplus.com>
References: <4E13710A.5060108@gosilverplus.com>
Message-ID: <CAKzWOxW80XDJsjjX2o_8UkJzLMts1aZ=VGu+ikXtym=4+cecMA@mail.gmail.com>

Zhang,

My apologies - I was going through my emails and I realized I never actually
looked into your question. I think there is a way to do what you want. It is
an alternative to doing a "bridging" conference. It's the auto outcall
feature:

http://wiki.freeswitch.org/wiki/Conference_set_auto_outcall

I think you will be able to do what you want by using the auto outcall and
setting the caller id info. (See the example.) Let us know if that works for
your application.

-MC

On Tue, Jul 5, 2011 at 1:16 PM, ran zhang <rzhang at gosilverplus.com> wrote:

> I want when the bridging conference is inviting someone to start the
> conference, the caller is shown as the real caller id number&name rather
> than
> whats stored in 'caller-id-number' and 'caller-id-name' in
> conference.conf.xml.
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/650204e4/attachment.html 

From msc at freeswitch.org  Wed Aug  3 09:52:13 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 2 Aug 2011 22:52:13 -0700
Subject: [Freeswitch-users] Calls not established : See the error details
In-Reply-To: <CA+P_fE8Fd5=2c4b48npp-27aQGGJN1vhFRE8FTvvgp+4xx0CXw@mail.gmail.com>
References: <CA+P_fE8Fd5=2c4b48npp-27aQGGJN1vhFRE8FTvvgp+4xx0CXw@mail.gmail.com>
Message-ID: <CAKzWOxUE=Pp+D9pw6_BsasvRUw3M4x=D8VrE6qkpBJqSHV0LgQ@mail.gmail.com>

Best thing here is to turn on debugging and a siptrace:
console loglevel debug
sofia global siptrace on

Do a test call and then put it into pastebin.freeswitch.org. Be sure to use
"FreeSWITCH Log" as the syntax highlighting.

-MC

On Tue, Aug 2, 2011 at 10:10 PM, Dhairya Vora <dhairya.blogs at gmail.com>wrote:

> I don't understand why two users (registered directly to freeswitch) are
> unable to connect each other?
>
>
> When a registered user 1001 calls registered user 1002, call fails giving
> this error
>
> ************************************************************************************************************************************************************
> freeswitch at localhost.localdomain> 2011-08-03 10:07:01.941089 [WARNING]
> sofia_reg.c:1337 SIP auth challenge (INVITE) on sofia profile 'internal' for
> [1002 at 172.16.10.211] from ip 172.16.10.248
> 2011-08-03 10:07:01.941089 [NOTICE] switch_channel.c:897 New Channel
> sofia/internal/1000 at 172.16.10.211 [0bf31c06-8379-4831-88d7-a7df3c4ecd49]
> 2011-08-03 10:07:01.941089 [INFO] mod_dialplan_xml.c:336 Processing 1000
> <1000>->1002 in context default
> 2011-08-03 10:07:01.941089 [INFO] switch_core_session.c:1281
> sofia/internal/1000 at 172.16.10.211 setting session heartbeat to 60
> second(s).
> 2011-08-03 10:07:01.941089 [ERR] mod_event_socket.c:457 Socket Error!
> 2011-08-03 10:07:01.941089 [NOTICE] switch_core_state_machine.c:189
> sofia/internal/1000 at 172.16.10.211 has executed the last dialplan
> instruction, hanging up.
> 2011-08-03 10:07:01.941089 [NOTICE] switch_core_state_machine.c:191 Hangup
> sofia/internal/1000 at 172.16.10.211 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-08-03 10:07:01.961094 [NOTICE] switch_core_session.c:1347 Session 25
> (sofia/internal/1000 at 172.16.10.211) Ended
> 2011-08-03 10:07:01.961094 [NOTICE] switch_core_session.c:1349 Close
> Channel sofia/internal/1000 at 172.16.10.211 [CS_DESTROY]
>
> ************************************************************************************************************************************************************
>
>
>
> When I make an outbound call to my mobile, calls are not going through the
> custom gateway
>
> ************************************************************************************************************************************************************
> freeswitch at localhost.localdomain> 2011-08-03 10:03:43.608234 [WARNING]
> sofia_reg.c:1337 SIP auth challenge (INVITE) on sofia profile 'internal' for
> [00919876543210 at 172.16.10.211] from ip 172.16.10.213
> 2011-08-03 10:03:43.728232 [NOTICE] switch_channel.c:897 New Channel
> sofia/internal/1002 at 172.16.10.211 [334906be-9cb0-44f2-9660-f9377c2ecfba]
> 2011-08-03 10:03:43.728232 [INFO] mod_dialplan_xml.c:336 Processing 1002
> <1002>->00919876543210 in context default
> 2011-08-03 10:03:43.728232 [INFO] switch_core_session.c:1281
> sofia/internal/1002 at 172.16.10.211 setting session heartbeat to 60
> second(s).
> 2011-08-03 10:03:43.728232 [ERR] mod_event_socket.c:457 Socket Error!
> 2011-08-03 10:03:43.728232 [NOTICE] switch_core_state_machine.c:189
> sofia/internal/1002 at 172.16.10.211 has executed the last dialplan
> instruction, hanging up.
> 2011-08-03 10:03:43.728232 [NOTICE] switch_core_state_machine.c:191 Hangup
> sofia/internal/1002 at 172.16.10.211 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-08-03 10:03:43.728232 [NOTICE] switch_core_session.c:1347 Session 24
> (sofia/internal/1002 at 172.16.10.211) Ended
> 2011-08-03 10:03:43.728232 [NOTICE] switch_core_session.c:1349 Close
> Channel sofia/internal/1002 at 172.16.10.211 [CS_DESTROY]
>
> ************************************************************************************************************************************************************
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110802/3c1c2c3b/attachment.html 

From dhairya.blogs at gmail.com  Wed Aug  3 10:36:34 2011
From: dhairya.blogs at gmail.com (Dhairya Vora)
Date: Wed, 3 Aug 2011 12:06:34 +0530
Subject: [Freeswitch-users] Calls not established : See the error details
In-Reply-To: <CAKzWOxUE=Pp+D9pw6_BsasvRUw3M4x=D8VrE6qkpBJqSHV0LgQ@mail.gmail.com>
References: <CA+P_fE8Fd5=2c4b48npp-27aQGGJN1vhFRE8FTvvgp+4xx0CXw@mail.gmail.com>
	<CAKzWOxUE=Pp+D9pw6_BsasvRUw3M4x=D8VrE6qkpBJqSHV0LgQ@mail.gmail.com>
Message-ID: <CA+P_fE_Bymr7M7_m8x7ETiFZrKSHbBhojCfiJmNeECceBPONng@mail.gmail.com>

Thanks a lot mc,
please check http://pastebin.freeswitch.org/16962 for internal call error
and
http://pastebin.freeswitch.org/16963 for external call error


(I used pastebin for the first time and just got impressed. Amazing
product.)


On Wed, Aug 3, 2011 at 11:22 AM, Michael Collins <msc at freeswitch.org> wrote:

> Best thing here is to turn on debugging and a siptrace:
> console loglevel debug
> sofia global siptrace on
>
> Do a test call and then put it into pastebin.freeswitch.org. Be sure to
> use "FreeSWITCH Log" as the syntax highlighting.
>
> -MC
>
> On Tue, Aug 2, 2011 at 10:10 PM, Dhairya Vora <dhairya.blogs at gmail.com>wrote:
>
>> I don't understand why two users (registered directly to freeswitch) are
>> unable to connect each other?
>>
>>
>> When a registered user 1001 calls registered user 1002, call fails giving
>> this error
>>
>> ************************************************************************************************************************************************************
>> freeswitch at localhost.localdomain> 2011-08-03 10:07:01.941089 [WARNING]
>> sofia_reg.c:1337 SIP auth challenge (INVITE) on sofia profile 'internal' for
>> [1002 at 172.16.10.211] from ip 172.16.10.248
>> 2011-08-03 10:07:01.941089 [NOTICE] switch_channel.c:897 New Channel
>> sofia/internal/1000 at 172.16.10.211 [0bf31c06-8379-4831-88d7-a7df3c4ecd49]
>> 2011-08-03 10:07:01.941089 [INFO] mod_dialplan_xml.c:336 Processing 1000
>> <1000>->1002 in context default
>> 2011-08-03 10:07:01.941089 [INFO] switch_core_session.c:1281
>> sofia/internal/1000 at 172.16.10.211 setting session heartbeat to 60
>> second(s).
>> 2011-08-03 10:07:01.941089 [ERR] mod_event_socket.c:457 Socket Error!
>> 2011-08-03 10:07:01.941089 [NOTICE] switch_core_state_machine.c:189
>> sofia/internal/1000 at 172.16.10.211 has executed the last dialplan
>> instruction, hanging up.
>> 2011-08-03 10:07:01.941089 [NOTICE] switch_core_state_machine.c:191 Hangup
>> sofia/internal/1000 at 172.16.10.211 [CS_EXECUTE] [NORMAL_CLEARING]
>> 2011-08-03 10:07:01.961094 [NOTICE] switch_core_session.c:1347 Session 25
>> (sofia/internal/1000 at 172.16.10.211) Ended
>> 2011-08-03 10:07:01.961094 [NOTICE] switch_core_session.c:1349 Close
>> Channel sofia/internal/1000 at 172.16.10.211 [CS_DESTROY]
>>
>> ************************************************************************************************************************************************************
>>
>>
>>
>> When I make an outbound call to my mobile, calls are not going through the
>> custom gateway
>>
>> ************************************************************************************************************************************************************
>> freeswitch at localhost.localdomain> 2011-08-03 10:03:43.608234 [WARNING]
>> sofia_reg.c:1337 SIP auth challenge (INVITE) on sofia profile 'internal' for
>> [00919876543210 at 172.16.10.211] from ip 172.16.10.213
>> 2011-08-03 10:03:43.728232 [NOTICE] switch_channel.c:897 New Channel
>> sofia/internal/1002 at 172.16.10.211 [334906be-9cb0-44f2-9660-f9377c2ecfba]
>> 2011-08-03 10:03:43.728232 [INFO] mod_dialplan_xml.c:336 Processing 1002
>> <1002>->00919876543210 in context default
>> 2011-08-03 10:03:43.728232 [INFO] switch_core_session.c:1281
>> sofia/internal/1002 at 172.16.10.211 setting session heartbeat to 60
>> second(s).
>> 2011-08-03 10:03:43.728232 [ERR] mod_event_socket.c:457 Socket Error!
>> 2011-08-03 10:03:43.728232 [NOTICE] switch_core_state_machine.c:189
>> sofia/internal/1002 at 172.16.10.211 has executed the last dialplan
>> instruction, hanging up.
>> 2011-08-03 10:03:43.728232 [NOTICE] switch_core_state_machine.c:191 Hangup
>> sofia/internal/1002 at 172.16.10.211 [CS_EXECUTE] [NORMAL_CLEARING]
>> 2011-08-03 10:03:43.728232 [NOTICE] switch_core_session.c:1347 Session 24
>> (sofia/internal/1002 at 172.16.10.211) Ended
>> 2011-08-03 10:03:43.728232 [NOTICE] switch_core_session.c:1349 Close
>> Channel sofia/internal/1002 at 172.16.10.211 [CS_DESTROY]
>>
>> ************************************************************************************************************************************************************
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110803/27c5ef65/attachment-0001.html 

From mays.david at gmail.com  Wed Aug  3 11:10:07 2011
From: mays.david at gmail.com (dma)
Date: Wed, 3 Aug 2011 00:10:07 -0700 (PDT)
Subject: [Freeswitch-users] Receiving junk ESL events
Message-ID: <1312355407424-6647776.post@n2.nabble.com>

Hello All,

I have some problem with receiving and handling ESL events.

My code is like:

void handle_eslswitch_event_plain(esl_handle_t *handle, esl_event_t *event)
{
        ... ...
        TRACE_DEBUG("ESL event [%d] %s: UUID: %s",
                event->event_id, fs_get_event_header(event, "Event-Name"),
uniqueid);
         print_event(event);
        ... ...
}

void handle_event(esl_handle_t *handle, esl_event_t *last_event)
{
        ... ...
        if (!strcasecmp(type, "text/event-plain")) {
            handle_eslswitch_event_plain(handle, handle->last_ievent);
        }
        ... ...
}

int main()
{
        ... ...
        //receive from event socket
        if (handle.last_event) handle_event(&handle, handle.last_event);
        ... ...

Usually I receive correct events. But I occasionally receive incorrect
event. See below the 1st is the normal event and the 2nd is the wrong event:

<2011-07-29 14:25:29> [DEBUG] ESL event [8] CHANNEL_HANGUP_COMPLETE: UUID:
f320e4a6-db23-46d0-8d89-9957eccbd4c9
<2011-07-29 14:25:29> [DEBUG] RECV EVENT
Event-Name: CHANNEL_HANGUP_COMPLETE
Core-UUID: 26b77cba-8fdd-486d-90ec-6844bca58c72
FreeSWITCH-Hostname: fs01
FreeSWITCH-IPv4: 10.1.1.46
FreeSWITCH-IPv6: ::1
Event-Date-Local: 2011-07-29 14:25:29

However, usually after failure in executing "hangup" and getting "-ERR ..."
in the event->last_sr_reply, I have wrong event, but not always (an -ERR
returned from executing "hangup" doesn't always result in a wrong event).
Here it is:

<2011-07-29 14:25:29> [NOTICE] -ERR invalid session id
[f320e4a6-db23-46d0-8d89-9957eccbd4c9]
<2011-07-28 15:55:23> [DEBUG] ESL event [0] : UUID:
<2011-07-28 15:55:23> [DEBUG] RECV EVENT
Content-Length: 6485
Content-Type: text/event-plain

Event-Name: CHANNEL_HANGUP
Core-UUID: 26b77cba-8fdd-486d-90ec-6844bca58c72
FreeSWITCH-Hostname: fs01
FreeSWITCH-IPv4: 10.1.1.46
FreeSWITCH-IPv6: %3A%3A1
Event-Date-Local: 2011-07-28%2015%3A55%3A23

In the above case, both Event ID and UUID are invalid.

I think i need more information regarding how to use the data elements in
ESL event, especially how to use the following:

    char last_reply[1024];
    /*! Las command reply when called with esl_send_recv */
    char last_sr_reply[1024];
    /*! Last event received. Only populated when **save_event is NULL */
    esl_event_t *last_event;
    /*! Last event received when called by esl_send_recv */
    esl_event_t *last_sr_event;
    /*! This will hold already processed events queued by esl_recv_event */
    esl_event_t *race_event;
    /*! Events that have content-type == text/plain and a body */
    esl_event_t *last_ievent;

My questions, what event should I check if I am interested in only
channel/call related events, and how? Do I check handle.last_event, or
handle.last_ievent, or what?

Else, when checking command reply, do I check last_sr_reply, or
last_sr_event->body?

Please kindly advice.

Thanks,
D.Ma



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Receiving-junk-ESL-events-tp6647776p6647776.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From jcgpoza at gmail.com  Wed Aug  3 11:13:40 2011
From: jcgpoza at gmail.com (Dissident)
Date: Wed, 3 Aug 2011 00:13:40 -0700 (PDT)
Subject: [Freeswitch-users] Calls not established : See the error details
In-Reply-To: <CA+P_fE8Fd5=2c4b48npp-27aQGGJN1vhFRE8FTvvgp+4xx0CXw@mail.gmail.com>
References: <CA+P_fE8Fd5=2c4b48npp-27aQGGJN1vhFRE8FTvvgp+4xx0CXw@mail.gmail.com>
Message-ID: <1312355620178-6647784.post@n2.nabble.com>

**2011-08-03 11:59:44.451984 [DEBUG] sofia.c:7066 IP 172.16.10.213 Rejected
by acl "domains". Falling back to Digest auth.

I'm new to this Freeswitch experience so my help could be misleading but it
seems as if the ACL is preventing you from making the call.

Have you checked your acl.conf.xml?

http://wiki.freeswitch.org/wiki/Acl

Best Regards



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Calls-not-established-See-the-error-details-tp6647547p6647784.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From lloydie.t at gmail.com  Wed Aug  3 11:41:45 2011
From: lloydie.t at gmail.com (lloyd thomas)
Date: Wed, 3 Aug 2011 08:41:45 +0100
Subject: [Freeswitch-users] Help setting up SIP reg
In-Reply-To: <CAKzWOxVC51tge2o1xEAW2NTvA2ewUD-XnanLRgRjsymtrRT-Og@mail.gmail.com>
References: <CAE=3qOYPpTuRTvq7RTJeUgzsCtvrfCjMiKFGu1fg0PHyZRE+-g@mail.gmail.com>
	<CAFiqYu=c8mY5BMWPumy1CgZHZ81kMRPgoNiF-6mjFAkW1q30fA@mail.gmail.com>
	<CAE=3qOZxwERT3CQEFHetO_o2s=aQUwYZ54TmMRPx9eYD=A7NoQ@mail.gmail.com>
	<CAKzWOxUEsGUKojF-vvstL6FvcDMhePqYGdXEQXQ4x-EJ81Bk6w@mail.gmail.com>
	<CAE=3qOacZh50iepxJKJD5UvtesxZAsN7RnPKNe1zdLO9VmFsuA@mail.gmail.com>
	<CAE=3qOZsNuiimFBZJTf4SbPRA8bRq_hJfPUunEnT8rxZj3qTnA@mail.gmail.com>
	<CAE=3qObVuLOhp2GD29Da-GfzB_qkbkyDP4kPmRxjdbJi7XeirQ@mail.gmail.com>
	<CAKzWOxUyZoBS5Evp4b61vP8fm64Lef0n01ti3KmVraQO_BqYtA@mail.gmail.com>
	<CAE=3qObsDBDTsKY2ycK_NierbS_A+-f896yMW4OArTkgauYgRA@mail.gmail.com>
	<CAKzWOxXYi-yO9DNDw1rHCduKq=h9q=BddFmn2dKuqALKb=t1fA@mail.gmail.com>
	<CAE=3qObVbDQi6g1m80WLtG5MhyosV2YYnqmOoausiz-yC+NmQQ@mail.gmail.com>
	<CAKzWOxVC51tge2o1xEAW2NTvA2ewUD-XnanLRgRjsymtrRT-Og@mail.gmail.com>
Message-ID: <CAE=3qObd_GZohA=zbWgGPx5bU1GLNsGkjnZPNeTf8NAxjZnVOw@mail.gmail.com>

Getting a little further, but calls are still failing.
Not sure why something to do with 404 error. could it be wrong number
format?

changed dialplan to <action application="bridge" data="sofia/external/01869$
1 at 80.40.150.150:5060"/> just in case the port number was incorrect.


On 2 August 2011 19:16, Michael Collins <msc at freeswitch.org> wrote:

> Lloyd,
>
> The gateway you created I think has a few issues. Let's have you bypass it
> altogether with a single change to your dialplan. Change your bridge line to
> this:
>
> <action application="bridge" data="sofia/external/01869$1 at 80.40.150.150"/>
>
> This bypasses the gateway and sends the call straight out the external
> profile. (You could also send it out the internal profile.)
>
> Try that and see what happens. If you have issues then do the usual console
> log and siptrace and put it into pastebin.freeswitch.org. Be sure to
> choose "FreeSWITCH Log" for the syntax highlighting.
>
> -MC
>
>
> On Tue, Aug 2, 2011 at 3:41 AM, lloyd thomas <lloydie.t at gmail.com> wrote:
>
>> Just did a test, but no joy. I suspect I may have to dispense with the
>> gateway settings and just bridge straight from the dial plan, but it is just
>> a guess.
>>
>>
>> dialplan
>> ----------------------------------------
>> <extension name="Local Dial">
>>    <condition field="destination_number" expression="^9([2-9][0-9]{5})$">
>>      <action application="bridge"
>> data="sofia/gateway/phisys-2circles/01869$1"/>
>>    </condition>
>> </extension>
>>
>> gateway
>> ---------------------------------------
>> <include>
>>   <gateway name="phisys-2circles">
>>     <param name="from-domain" value="80.40.150.150"/>
>>     <param name="proxy" value="80.40.150.150"/>
>>     <param name="sip-port" value="5060"/>
>>   </gateway>
>> </include>
>>
>>
>>
>> errors
>> ------------------------------------
>> 2011-08-02 11:32:42.056302 [DEBUG] mod_dptools.c:1059 sofia/internal/
>> 200 at phisys.tele.phi.co.uk SET [RFC2822_DATE]=[Tue, 02 Aug 2011 11:32:42
>> +0100]
>> EXECUTE sofia/internal/200 at phisys.tele.phi.co.ukbridge(sofia/gateway/phisys-2circles/01869321110)
>> 2011-08-02 11:32:42.073667 [ERR] mod_sofia.c:3940 Invalid Gateway
>> 2011-08-02 11:32:42.073667 [NOTICE] mod_sofia.c:4282 Close Channel N/A
>> [CS_NEW]
>> 2011-08-02 11:32:42.076523 [DEBUG] switch_core_state_machine.c:452 ()
>> Running State Change CS_DESTROY
>> 2011-08-02 11:32:42.079574 [DEBUG] switch_core_state_machine.c:462 (N/A)
>> State DESTROY
>> 2011-08-02 11:32:42.079574 [DEBUG] mod_sofia.c:362 N/A SOFIA DESTROY
>> 2011-08-02 11:32:42.081946 [DEBUG] switch_core_state_machine.c:462 (N/A)
>> State DESTROY going to sleep
>> 2011-08-02 11:32:42.084166 [ERR] switch_ivr_originate.c:2640 Cannot create
>> outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]
>> 2011-08-02 11:32:42.085649 [DEBUG] switch_ivr_originate.c:3506 Originate
>> Resulted in Error Cause: 28 [INVALID_NUMBER_FORMAT]
>> 2011-08-02 11:32:42.087275 [INFO] mod_dptools.c:2623 Originate Failed.
>> Cause: INVALID_NUMBER_FORMAT
>> 2011-08-02 11:32:42.088652 [DEBUG] switch_channel.c:2559 (sofia/internal/
>> 200 at phisys.tele.phi.co.uk) Callstate Change RINGING -> HANGUP
>> 2011-08-02 11:32:42.093248 [NOTICE] mod_dptools.c:2686 Hangup
>> sofia/internal/200 at phisys.tele.phi.co.uk [CS_EXECUTE]
>> [INVALID_NUMBER_FORMAT]
>> 2011-08-02 11:32:42.096925 [DEBUG] switch_channel.c:2575 Send signal
>> sofia/internal/200 at phisys.tele.phi.co.uk [KILL]
>>
>>
>> On 1 August 2011 20:50, Michael Collins <msc at freeswitch.org> wrote:
>>
>>> Okay, so what happens when you dial out? Sorry, it's been a few days and
>>> I don't recall where we left off. Be sure to include console log w/ siptrace
>>> on pastebin.freeswitch.org.
>>>
>>> -MC
>>>
>>>
>>> On Mon, Aug 1, 2011 at 12:35 PM, lloyd thomas <lloydie.t at gmail.com>wrote:
>>>
>>>> I think they have my IP on a white list.
>>>>
>>>>
>>>> On 1 August 2011 16:24, Michael Collins <msc at freeswitch.org> wrote:
>>>>
>>>>> Do they challenge you (digest auth) or do they have your IP address on
>>>>> a white list? That's a critical piece of information that only your provider
>>>>> can supply.
>>>>>
>>>>> -MC
>>>>>
>>>>>
>>>>> On Fri, Jul 29, 2011 at 9:31 PM, lloyd thomas <lloydie.t at gmail.com>wrote:
>>>>>
>>>>>> OK Inbound working with:
>>>>>>
>>>>>>   <!--2circles -->
>>>>>>     <list name="domains" default="allow">
>>>>>>       <node type="allow" cidr="80.40.150.150/32"/>
>>>>>>     </list>
>>>>>>
>>>>>> Just need to sort outbound.
>>>>>>
>>>>>>
>>>>>> On 30 July 2011 04:59, lloyd thomas <lloydie.t at gmail.com> wrote:
>>>>>>
>>>>>>> Hi, dialling in produces the following error.
>>>>>>>
>>>>>>> 2011-07-30 04:56:07.818936 [DEBUG] sofia.c:6517 IP 80.40.150.150
>>>>>>> Rejected by acl "domains". Falling back to Digest auth.
>>>>>>> 2011-07-30 04:56:07.826367 [WARNING] sofia_reg.c:1246 SIP auth
>>>>>>> challenge (INVITE) on sofia profile 'internal' for
>>>>>>> [01869******@172.16.XXX.XXX] from ip 80.40.150.150
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On 30 July 2011 04:34, lloyd thomas <lloydie.t at gmail.com> wrote:
>>>>>>>
>>>>>>>> I am registering with a them. I could not find suitable example in
>>>>>>>> http://wiki.freeswitch.org/wiki/SIP_Provider_Examples which
>>>>>>>>
>>>>>>>>
>>>>>>>> On 29 July 2011 21:57, Michael Collins <msc at freeswitch.org> wrote:
>>>>>>>>
>>>>>>>>> Are you registering with the provider or are they registering with
>>>>>>>>> you? If they register with you then a user example is appropriate. If you
>>>>>>>>> are registering with them then all you need is a gateway configured.
>>>>>>>>> -MC
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On Fri, Jul 29, 2011 at 1:40 PM, lloyd thomas <lloydie.t at gmail.com
>>>>>>>>> > wrote:
>>>>>>>>>
>>>>>>>>>> Sorry, example is not clear to me.
>>>>>>>>>> I don't understand why a user config is relevant to sip
>>>>>>>>>> registration for a provider.
>>>>>>>>>> An example will help me more. Maybe CIDR attribute in a
>>>>>>>>>> sip_profile gateway could help.
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> On 29 July 2011 19:55, Steven Ayre <steveayre at gmail.com> wrote:
>>>>>>>>>>
>>>>>>>>>>> Look at the cidr attribute in the user directory to authenticate
>>>>>>>>>>> by IP:
>>>>>>>>>>> http://wiki.freeswitch.org/wiki/Acl#Users
>>>>>>>>>>>
>>>>>>>>>>> -Steve
>>>>>>>>>>>
>>>>>>>>>>> On 29 July 2011 19:38, lloyd thomas <lloydie.t at gmail.com> wrote:
>>>>>>>>>>>
>>>>>>>>>>>> *Hi I need a little help setting up a SIP registration for a
>>>>>>>>>>>> provider that does not use auth.*
>>>>>>>>>>>>
>>>>>>>>>>>> *All I have is info below.*
>>>>>>>>>>>> **
>>>>>>>>>>>>
>>>>>>>>>>>> *
>>>>>>>>>>>> *
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> SBC/Proxy IP:        80.40.150.150:5060
>>>>>>>>>>>>
>>>>>>>>>>>> Authentication:      Trusted IP ? 88.221.85.33
>>>>>>>>>>>>
>>>>>>>>>>>> Assigned DDI:        01869******, 01869******
>>>>>>>>>>>>
>>>>>>>>>>>> DTMF Method:         RFC2833
>>>>>>>>>>>>
>>>>>>>>>>>> Status:              Live
>>>>>>>>>>>>
>>>>>>>>>>>> No. of trunks:       2x
>>>>>>>>>>>>
>>>>>>>>>>>> Session Timer:       1800
>>>>>>>>>>>>
>>>>>>>>>>>> Profile*:*             Generic (35060)
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> Apparently the following is used for *
>>>>>>>>>>>>
>>>>>>>>>>>> [vibe]
>>>>>>>>>>>>
>>>>>>>>>>>> type = friend
>>>>>>>>>>>>
>>>>>>>>>>>> host = 80.40.150.150
>>>>>>>>>>>>
>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>>>
>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>>
>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110803/7ae4b897/attachment-0001.html 

From lloydie.t at gmail.com  Wed Aug  3 11:42:20 2011
From: lloydie.t at gmail.com (lloyd thomas)
Date: Wed, 3 Aug 2011 08:42:20 +0100
Subject: [Freeswitch-users] Help setting up SIP reg
In-Reply-To: <CAE=3qObd_GZohA=zbWgGPx5bU1GLNsGkjnZPNeTf8NAxjZnVOw@mail.gmail.com>
References: <CAE=3qOYPpTuRTvq7RTJeUgzsCtvrfCjMiKFGu1fg0PHyZRE+-g@mail.gmail.com>
	<CAFiqYu=c8mY5BMWPumy1CgZHZ81kMRPgoNiF-6mjFAkW1q30fA@mail.gmail.com>
	<CAE=3qOZxwERT3CQEFHetO_o2s=aQUwYZ54TmMRPx9eYD=A7NoQ@mail.gmail.com>
	<CAKzWOxUEsGUKojF-vvstL6FvcDMhePqYGdXEQXQ4x-EJ81Bk6w@mail.gmail.com>
	<CAE=3qOacZh50iepxJKJD5UvtesxZAsN7RnPKNe1zdLO9VmFsuA@mail.gmail.com>
	<CAE=3qOZsNuiimFBZJTf4SbPRA8bRq_hJfPUunEnT8rxZj3qTnA@mail.gmail.com>
	<CAE=3qObVuLOhp2GD29Da-GfzB_qkbkyDP4kPmRxjdbJi7XeirQ@mail.gmail.com>
	<CAKzWOxUyZoBS5Evp4b61vP8fm64Lef0n01ti3KmVraQO_BqYtA@mail.gmail.com>
	<CAE=3qObsDBDTsKY2ycK_NierbS_A+-f896yMW4OArTkgauYgRA@mail.gmail.com>
	<CAKzWOxXYi-yO9DNDw1rHCduKq=h9q=BddFmn2dKuqALKb=t1fA@mail.gmail.com>
	<CAE=3qObVbDQi6g1m80WLtG5MhyosV2YYnqmOoausiz-yC+NmQQ@mail.gmail.com>
	<CAKzWOxVC51tge2o1xEAW2NTvA2ewUD-XnanLRgRjsymtrRT-Og@mail.gmail.com>
	<CAE=3qObd_GZohA=zbWgGPx5bU1GLNsGkjnZPNeTf8NAxjZnVOw@mail.gmail.com>
Message-ID: <CAE=3qOYtCeFWPs9k-GEQwz8BMu8WipEBYwu8AfrW_Gx+qxAXeg@mail.gmail.com>

Log is here
http://pastebin.freeswitch.org/16964


On 3 August 2011 08:41, lloyd thomas <lloydie.t at gmail.com> wrote:

> Getting a little further, but calls are still failing.
> Not sure why something to do with 404 error. could it be wrong number
> format?
>
> changed dialplan to <action application="bridge"
> data="sofia/external/01869$1 at 80.40.150.150:5060"/> just in case the port
> number was incorrect.
>
>
>
> On 2 August 2011 19:16, Michael Collins <msc at freeswitch.org> wrote:
>
>> Lloyd,
>>
>> The gateway you created I think has a few issues. Let's have you bypass it
>> altogether with a single change to your dialplan. Change your bridge line to
>> this:
>>
>> <action application="bridge" data="sofia/external/01869$1 at 80.40.150.150
>> "/>
>>
>> This bypasses the gateway and sends the call straight out the external
>> profile. (You could also send it out the internal profile.)
>>
>> Try that and see what happens. If you have issues then do the usual
>> console log and siptrace and put it into pastebin.freeswitch.org. Be sure
>> to choose "FreeSWITCH Log" for the syntax highlighting.
>>
>> -MC
>>
>>
>> On Tue, Aug 2, 2011 at 3:41 AM, lloyd thomas <lloydie.t at gmail.com> wrote:
>>
>>> Just did a test, but no joy. I suspect I may have to dispense with the
>>> gateway settings and just bridge straight from the dial plan, but it is just
>>> a guess.
>>>
>>>
>>> dialplan
>>> ----------------------------------------
>>> <extension name="Local Dial">
>>>    <condition field="destination_number" expression="^9([2-9][0-9]{5})$">
>>>      <action application="bridge"
>>> data="sofia/gateway/phisys-2circles/01869$1"/>
>>>    </condition>
>>> </extension>
>>>
>>> gateway
>>> ---------------------------------------
>>> <include>
>>>   <gateway name="phisys-2circles">
>>>     <param name="from-domain" value="80.40.150.150"/>
>>>     <param name="proxy" value="80.40.150.150"/>
>>>     <param name="sip-port" value="5060"/>
>>>   </gateway>
>>> </include>
>>>
>>>
>>>
>>> errors
>>> ------------------------------------
>>> 2011-08-02 11:32:42.056302 [DEBUG] mod_dptools.c:1059 sofia/internal/
>>> 200 at phisys.tele.phi.co.uk SET [RFC2822_DATE]=[Tue, 02 Aug 2011 11:32:42
>>> +0100]
>>> EXECUTE sofia/internal/200 at phisys.tele.phi.co.ukbridge(sofia/gateway/phisys-2circles/01869321110)
>>> 2011-08-02 11:32:42.073667 [ERR] mod_sofia.c:3940 Invalid Gateway
>>> 2011-08-02 11:32:42.073667 [NOTICE] mod_sofia.c:4282 Close Channel N/A
>>> [CS_NEW]
>>> 2011-08-02 11:32:42.076523 [DEBUG] switch_core_state_machine.c:452 ()
>>> Running State Change CS_DESTROY
>>> 2011-08-02 11:32:42.079574 [DEBUG] switch_core_state_machine.c:462 (N/A)
>>> State DESTROY
>>> 2011-08-02 11:32:42.079574 [DEBUG] mod_sofia.c:362 N/A SOFIA DESTROY
>>> 2011-08-02 11:32:42.081946 [DEBUG] switch_core_state_machine.c:462 (N/A)
>>> State DESTROY going to sleep
>>> 2011-08-02 11:32:42.084166 [ERR] switch_ivr_originate.c:2640 Cannot
>>> create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]
>>> 2011-08-02 11:32:42.085649 [DEBUG] switch_ivr_originate.c:3506 Originate
>>> Resulted in Error Cause: 28 [INVALID_NUMBER_FORMAT]
>>> 2011-08-02 11:32:42.087275 [INFO] mod_dptools.c:2623 Originate Failed.
>>> Cause: INVALID_NUMBER_FORMAT
>>> 2011-08-02 11:32:42.088652 [DEBUG] switch_channel.c:2559 (sofia/internal/
>>> 200 at phisys.tele.phi.co.uk) Callstate Change RINGING -> HANGUP
>>> 2011-08-02 11:32:42.093248 [NOTICE] mod_dptools.c:2686 Hangup
>>> sofia/internal/200 at phisys.tele.phi.co.uk [CS_EXECUTE]
>>> [INVALID_NUMBER_FORMAT]
>>> 2011-08-02 11:32:42.096925 [DEBUG] switch_channel.c:2575 Send signal
>>> sofia/internal/200 at phisys.tele.phi.co.uk [KILL]
>>>
>>>
>>> On 1 August 2011 20:50, Michael Collins <msc at freeswitch.org> wrote:
>>>
>>>> Okay, so what happens when you dial out? Sorry, it's been a few days and
>>>> I don't recall where we left off. Be sure to include console log w/ siptrace
>>>> on pastebin.freeswitch.org.
>>>>
>>>> -MC
>>>>
>>>>
>>>> On Mon, Aug 1, 2011 at 12:35 PM, lloyd thomas <lloydie.t at gmail.com>wrote:
>>>>
>>>>> I think they have my IP on a white list.
>>>>>
>>>>>
>>>>> On 1 August 2011 16:24, Michael Collins <msc at freeswitch.org> wrote:
>>>>>
>>>>>> Do they challenge you (digest auth) or do they have your IP address on
>>>>>> a white list? That's a critical piece of information that only your provider
>>>>>> can supply.
>>>>>>
>>>>>> -MC
>>>>>>
>>>>>>
>>>>>> On Fri, Jul 29, 2011 at 9:31 PM, lloyd thomas <lloydie.t at gmail.com>wrote:
>>>>>>
>>>>>>> OK Inbound working with:
>>>>>>>
>>>>>>>   <!--2circles -->
>>>>>>>     <list name="domains" default="allow">
>>>>>>>       <node type="allow" cidr="80.40.150.150/32"/>
>>>>>>>     </list>
>>>>>>>
>>>>>>> Just need to sort outbound.
>>>>>>>
>>>>>>>
>>>>>>> On 30 July 2011 04:59, lloyd thomas <lloydie.t at gmail.com> wrote:
>>>>>>>
>>>>>>>> Hi, dialling in produces the following error.
>>>>>>>>
>>>>>>>> 2011-07-30 04:56:07.818936 [DEBUG] sofia.c:6517 IP 80.40.150.150
>>>>>>>> Rejected by acl "domains". Falling back to Digest auth.
>>>>>>>> 2011-07-30 04:56:07.826367 [WARNING] sofia_reg.c:1246 SIP auth
>>>>>>>> challenge (INVITE) on sofia profile 'internal' for
>>>>>>>> [01869******@172.16.XXX.XXX] from ip 80.40.150.150
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> On 30 July 2011 04:34, lloyd thomas <lloydie.t at gmail.com> wrote:
>>>>>>>>
>>>>>>>>> I am registering with a them. I could not find suitable example in
>>>>>>>>> http://wiki.freeswitch.org/wiki/SIP_Provider_Examples which
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On 29 July 2011 21:57, Michael Collins <msc at freeswitch.org> wrote:
>>>>>>>>>
>>>>>>>>>> Are you registering with the provider or are they registering with
>>>>>>>>>> you? If they register with you then a user example is appropriate. If you
>>>>>>>>>> are registering with them then all you need is a gateway configured.
>>>>>>>>>> -MC
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> On Fri, Jul 29, 2011 at 1:40 PM, lloyd thomas <
>>>>>>>>>> lloydie.t at gmail.com> wrote:
>>>>>>>>>>
>>>>>>>>>>> Sorry, example is not clear to me.
>>>>>>>>>>> I don't understand why a user config is relevant to sip
>>>>>>>>>>> registration for a provider.
>>>>>>>>>>> An example will help me more. Maybe CIDR attribute in a
>>>>>>>>>>> sip_profile gateway could help.
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> On 29 July 2011 19:55, Steven Ayre <steveayre at gmail.com> wrote:
>>>>>>>>>>>
>>>>>>>>>>>> Look at the cidr attribute in the user directory to authenticate
>>>>>>>>>>>> by IP:
>>>>>>>>>>>> http://wiki.freeswitch.org/wiki/Acl#Users
>>>>>>>>>>>>
>>>>>>>>>>>> -Steve
>>>>>>>>>>>>
>>>>>>>>>>>> On 29 July 2011 19:38, lloyd thomas <lloydie.t at gmail.com>wrote:
>>>>>>>>>>>>
>>>>>>>>>>>>> *Hi I need a little help setting up a SIP registration for a
>>>>>>>>>>>>> provider that does not use auth.*
>>>>>>>>>>>>>
>>>>>>>>>>>>> *All I have is info below.*
>>>>>>>>>>>>> **
>>>>>>>>>>>>>
>>>>>>>>>>>>> *
>>>>>>>>>>>>> *
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> SBC/Proxy IP:        80.40.150.150:5060
>>>>>>>>>>>>>
>>>>>>>>>>>>> Authentication:      Trusted IP ? 88.221.85.33
>>>>>>>>>>>>>
>>>>>>>>>>>>> Assigned DDI:        01869******, 01869******
>>>>>>>>>>>>>
>>>>>>>>>>>>> DTMF Method:         RFC2833
>>>>>>>>>>>>>
>>>>>>>>>>>>> Status:              Live
>>>>>>>>>>>>>
>>>>>>>>>>>>> No. of trunks:       2x
>>>>>>>>>>>>>
>>>>>>>>>>>>> Session Timer:       1800
>>>>>>>>>>>>>
>>>>>>>>>>>>> Profile*:*             Generic (35060)
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> Apparently the following is used for *
>>>>>>>>>>>>>
>>>>>>>>>>>>> [vibe]
>>>>>>>>>>>>>
>>>>>>>>>>>>> type = friend
>>>>>>>>>>>>>
>>>>>>>>>>>>> host = 80.40.150.150
>>>>>>>>>>>>>
>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>>>>
>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>>>
>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>>
>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110803/3da8c9a5/attachment-0001.html 

From ovvenkatesan at gmail.com  Wed Aug  3 11:52:38 2011
From: ovvenkatesan at gmail.com (ovvenkat)
Date: Wed, 3 Aug 2011 13:22:38 +0530
Subject: [Freeswitch-users] freetdm error message
In-Reply-To: <CAA4nhyC7s2qzd6DDypzN0xVPdVmeWompgncFcyOZhgF6FZQqyQ@mail.gmail.com>
References: <CAABykDrOSPH4krH45EjxURkXOFJAGMV6q0T3CehGEQcgKzPfxQ@mail.gmail.com>
	<CAA4nhyC7s2qzd6DDypzN0xVPdVmeWompgncFcyOZhgF6FZQqyQ@mail.gmail.com>
Message-ID: <CAABykDqDBmg3c9xrReErzk=Put+g+7cKqe2ooOyDDc5JEuXxLw@mail.gmail.com>

Hi,
Thank you for your response.
Here I have created 2 pcap files.

1. isdn.pcap file captured only signaling status.

2. isdn_oncall.pcap file captured at the time of outbound call.
    (call got disconnected)

Since Its live server, plz reply me ASAP.

Here is the System configuration.

freeswitch at 192.168.1.110@internal> version
FreeSWITCH Version 1.0.head (git-a7c6fa9 2011-08-02 00-27-38 -0500)


[root at localhost var]# wanrouter status
Devices currently active:
        wanpipe1


Wanpipe Config:

Device name | Protocol Map | Adapter  | IRQ | Slot/IO | If's | CLK | Baud
rate |
wanpipe1    | N/A          | A101/1D/A102/2D/4/4D/8| 21  | 0       | 1    |
N/A | 0         |

Wanrouter Status:

Device name | Protocol | Station | Status        |
wanpipe1    | AFT TE1  | N/A     | Connected     |


[root at localhost var]# wanrouter hwprobe
-------------------------------
| Wanpipe Hardware Probe Info |
-------------------------------
1 . AFT-A101-SH : SLOT=0 : BUS=17 : IRQ=21 : CPU=A : PORT=1 : HWEC=0 : V=37

Card Cnt: A101-2=1


[root at localhost var]# wanpipemon -i w1g1 -c Ta

***** w1g1: E1 Rx Alarms (Framer) *****

ALOS:   OFF     | LOS:  OFF
RED:    OFF     | AIS:  OFF
LOF:    OFF     | RAI:  OFF* ( some time its ON )*



***** w1g1: E1 Rx Alarms (LIU) *****

Short Circuit:  OFF
Open Circuit:   OFF
Loss of Signal: OFF

***** w1g1: E1 Tx Alarms *****

AIS:    OFF     | YEL:  OFF


***** w1g1: E1 Performance Monitoring Counters *****

Line Code Violation     : 1373
Far End Block Errors    : 0
CRC4 Errors             : 0
FAS Errors              : 2822


Rx Level        : > -2.5db





Regards,
Venkat.



On Wed, Aug 3, 2011 at 8:06 AM, Moises Silva <moises.silva at gmail.com> wrote:

> On Tue, Aug 2, 2011 at 7:45 AM, ovvenkat <ovvenkatesan at gmail.com> wrote:
> > Hi to all,
> >
> > In my "fs_cli"  I am getting bellow message very frequently.
> > Can you any one help me where is the error?
> >
> > 2011-08-02 17:11:33.028452 [INFO] ftmod_sangoma_isdn_stack_rcv.c:995
> > sng_isdn->s1: Invalid Q.921/Q.931 frame - ignoring len:1
> > 2011-08-02 17:11:33.048455 [INFO] ftmod_sangoma_isdn_stack_rcv.c:995
> > sng_isdn->s1: Invalid Q.921/Q.931 frame - ignoring len:1
> > 2011-08-02 17:11:33.048455 [INFO] ftmod_sangoma_isdn_stack_rcv.c:995
> > sng_isdn->s1: Invalid Q.921/Q.931 frame - ignoring len:1
> > 2011-08-02 17:11:33.068455 [INFO] ftmod_sangoma_isdn_stack_rcv.c:995
> > sng_isdn->s1: Invalid Q.921/Q.931 frame - ignoring len:1
> > 2011-08-02 17:11:33.068455 [INFO] ftmod_sangoma_isdn_stack_rcv.c:995
> > sng_isdn->s1: Invalid Q.921/Q.931 frame - ignoring len:1
> > 2011-08-02 17:11:33.068455 [INFO] ftmod_sangoma_isdn_stack_rcv.c:995
> > sng_isdn->s1: Invalid Q.921/Q.931 frame - ignoring len:1
>
> That really looks like some sort of garbage on the line (although is
> not strictly only garbage since they seem to be valid HDLC frames).
>
> How often do you get this? the word "frequently" means nothing from an
> engineering point of view. Every minute you get a burst of these? or
> is it a constant flow? can you receive/place calls despite this
> message?
>
> Try taking a pcap file when you're receiving those messages:
> http://wiki.sangoma.com/wanpipe-wireshark-pcap-pri-bri-wan-t1-e1-tracing
>
> Moises Silva
> Senior Software Engineer, Software Development Manager
> Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
> L3R 9R6 Canada
> t. 1 905 474 1990 x128 | e. moy at sangoma.com
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

If you have come to help me, you are wasting your time.
If you have come to because your liberation is bound up in mine, we can work
together.


Regards
Venkatesan OV.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110803/0f5f1c41/attachment-0001.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: isdn.pcap
Type: application/octet-stream
Size: 6643 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110803/0f5f1c41/attachment-0002.obj 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: isdn_oncall.pca
Type: application/octet-stream
Size: 13285 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110803/0f5f1c41/attachment-0003.obj 

From steveayre at gmail.com  Wed Aug  3 11:54:43 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 3 Aug 2011 08:54:43 +0100
Subject: [Freeswitch-users] Calls not established : See the error details
In-Reply-To: <CA+P_fE8Fd5=2c4b48npp-27aQGGJN1vhFRE8FTvvgp+4xx0CXw@mail.gmail.com>
References: <CA+P_fE8Fd5=2c4b48npp-27aQGGJN1vhFRE8FTvvgp+4xx0CXw@mail.gmail.com>
Message-ID: <CAFiqYun8ppGVpngCb78QCi29EsxM7=EouMWPc2D-j+sUxz2dtw@mail.gmail.com>

The debug-level log would be helpful.

What does your dialplan look like? It looks like you're executing a dialplan
extension that's trying to connect somewhere with the event socket (to have
the logic executed externally) but failing to connect to a ESL server:
mod_event_socket.c:457 Socket Error!

That suggests to me your dialplan might not be doing what you expect it to.

-Steve



On 3 August 2011 06:10, Dhairya Vora <dhairya.blogs at gmail.com> wrote:

> I don't understand why two users (registered directly to freeswitch) are
> unable to connect each other?
>
>
> When a registered user 1001 calls registered user 1002, call fails giving
> this error
>
> ************************************************************************************************************************************************************
> freeswitch at localhost.localdomain> 2011-08-03 10:07:01.941089 [WARNING]
> sofia_reg.c:1337 SIP auth challenge (INVITE) on sofia profile 'internal' for
> [1002 at 172.16.10.211] from ip 172.16.10.248
> 2011-08-03 10:07:01.941089 [NOTICE] switch_channel.c:897 New Channel
> sofia/internal/1000 at 172.16.10.211 [0bf31c06-8379-4831-88d7-a7df3c4ecd49]
> 2011-08-03 10:07:01.941089 [INFO] mod_dialplan_xml.c:336 Processing 1000
> <1000>->1002 in context default
> 2011-08-03 10:07:01.941089 [INFO] switch_core_session.c:1281
> sofia/internal/1000 at 172.16.10.211 setting session heartbeat to 60
> second(s).
> 2011-08-03 10:07:01.941089 [ERR] mod_event_socket.c:457 Socket Error!
> 2011-08-03 10:07:01.941089 [NOTICE] switch_core_state_machine.c:189
> sofia/internal/1000 at 172.16.10.211 has executed the last dialplan
> instruction, hanging up.
> 2011-08-03 10:07:01.941089 [NOTICE] switch_core_state_machine.c:191 Hangup
> sofia/internal/1000 at 172.16.10.211 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-08-03 10:07:01.961094 [NOTICE] switch_core_session.c:1347 Session 25
> (sofia/internal/1000 at 172.16.10.211) Ended
> 2011-08-03 10:07:01.961094 [NOTICE] switch_core_session.c:1349 Close
> Channel sofia/internal/1000 at 172.16.10.211 [CS_DESTROY]
>
> ************************************************************************************************************************************************************
>
>
>
> When I make an outbound call to my mobile, calls are not going through the
> custom gateway
>
> ************************************************************************************************************************************************************
> freeswitch at localhost.localdomain> 2011-08-03 10:03:43.608234 [WARNING]
> sofia_reg.c:1337 SIP auth challenge (INVITE) on sofia profile 'internal' for
> [00919876543210 at 172.16.10.211] from ip 172.16.10.213
> 2011-08-03 10:03:43.728232 [NOTICE] switch_channel.c:897 New Channel
> sofia/internal/1002 at 172.16.10.211 [334906be-9cb0-44f2-9660-f9377c2ecfba]
> 2011-08-03 10:03:43.728232 [INFO] mod_dialplan_xml.c:336 Processing 1002
> <1002>->00919876543210 in context default
> 2011-08-03 10:03:43.728232 [INFO] switch_core_session.c:1281
> sofia/internal/1002 at 172.16.10.211 setting session heartbeat to 60
> second(s).
> 2011-08-03 10:03:43.728232 [ERR] mod_event_socket.c:457 Socket Error!
> 2011-08-03 10:03:43.728232 [NOTICE] switch_core_state_machine.c:189
> sofia/internal/1002 at 172.16.10.211 has executed the last dialplan
> instruction, hanging up.
> 2011-08-03 10:03:43.728232 [NOTICE] switch_core_state_machine.c:191 Hangup
> sofia/internal/1002 at 172.16.10.211 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-08-03 10:03:43.728232 [NOTICE] switch_core_session.c:1347 Session 24
> (sofia/internal/1002 at 172.16.10.211) Ended
> 2011-08-03 10:03:43.728232 [NOTICE] switch_core_session.c:1349 Close
> Channel sofia/internal/1002 at 172.16.10.211 [CS_DESTROY]
>
> ************************************************************************************************************************************************************
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110803/7051d17f/attachment.html 

From dhairya.blogs at gmail.com  Wed Aug  3 12:35:56 2011
From: dhairya.blogs at gmail.com (Dhairya Vora)
Date: Wed, 3 Aug 2011 14:05:56 +0530
Subject: [Freeswitch-users] Calls not established : See the error details
In-Reply-To: <CAFiqYun8ppGVpngCb78QCi29EsxM7=EouMWPc2D-j+sUxz2dtw@mail.gmail.com>
References: <CA+P_fE8Fd5=2c4b48npp-27aQGGJN1vhFRE8FTvvgp+4xx0CXw@mail.gmail.com>
	<CAFiqYun8ppGVpngCb78QCi29EsxM7=EouMWPc2D-j+sUxz2dtw@mail.gmail.com>
Message-ID: <CA+P_fE9QZ6e2N3aixe+V4n5sbgJJ3cHts6nbKg4oM0eEbuYmkw@mail.gmail.com>

One thing is making a confusion in my mind. Can plivo (installed on
freeswitch) be the problem ?
According to me, it would not the be reason. But still just asking for the
clarification.



On Wed, Aug 3, 2011 at 1:24 PM, Steven Ayre <steveayre at gmail.com> wrote:

> The debug-level log would be helpful.
>
> What does your dialplan look like? It looks like you're executing a
> dialplan extension that's trying to connect somewhere with the event socket
> (to have the logic executed externally) but failing to connect to a ESL
> server:
>
> mod_event_socket.c:457 Socket Error!
>
> That suggests to me your dialplan might not be doing what you expect it to.
>
> -Steve
>
>
>
> On 3 August 2011 06:10, Dhairya Vora <dhairya.blogs at gmail.com> wrote:
>
>> I don't understand why two users (registered directly to freeswitch) are
>> unable to connect each other?
>>
>>
>> When a registered user 1001 calls registered user 1002, call fails giving
>> this error
>>
>> ************************************************************************************************************************************************************
>> freeswitch at localhost.localdomain> 2011-08-03 10:07:01.941089 [WARNING]
>> sofia_reg.c:1337 SIP auth challenge (INVITE) on sofia profile 'internal' for
>> [1002 at 172.16.10.211] from ip 172.16.10.248
>> 2011-08-03 10:07:01.941089 [NOTICE] switch_channel.c:897 New Channel
>> sofia/internal/1000 at 172.16.10.211 [0bf31c06-8379-4831-88d7-a7df3c4ecd49]
>> 2011-08-03 10:07:01.941089 [INFO] mod_dialplan_xml.c:336 Processing 1000
>> <1000>->1002 in context default
>> 2011-08-03 10:07:01.941089 [INFO] switch_core_session.c:1281
>> sofia/internal/1000 at 172.16.10.211 setting session heartbeat to 60
>> second(s).
>> 2011-08-03 10:07:01.941089 [ERR] mod_event_socket.c:457 Socket Error!
>> 2011-08-03 10:07:01.941089 [NOTICE] switch_core_state_machine.c:189
>> sofia/internal/1000 at 172.16.10.211 has executed the last dialplan
>> instruction, hanging up.
>> 2011-08-03 10:07:01.941089 [NOTICE] switch_core_state_machine.c:191 Hangup
>> sofia/internal/1000 at 172.16.10.211 [CS_EXECUTE] [NORMAL_CLEARING]
>> 2011-08-03 10:07:01.961094 [NOTICE] switch_core_session.c:1347 Session 25
>> (sofia/internal/1000 at 172.16.10.211) Ended
>> 2011-08-03 10:07:01.961094 [NOTICE] switch_core_session.c:1349 Close
>> Channel sofia/internal/1000 at 172.16.10.211 [CS_DESTROY]
>>
>> ************************************************************************************************************************************************************
>>
>>
>>
>> When I make an outbound call to my mobile, calls are not going through the
>> custom gateway
>>
>> ************************************************************************************************************************************************************
>> freeswitch at localhost.localdomain> 2011-08-03 10:03:43.608234 [WARNING]
>> sofia_reg.c:1337 SIP auth challenge (INVITE) on sofia profile 'internal' for
>> [00919876543210 at 172.16.10.211] from ip 172.16.10.213
>> 2011-08-03 10:03:43.728232 [NOTICE] switch_channel.c:897 New Channel
>> sofia/internal/1002 at 172.16.10.211 [334906be-9cb0-44f2-9660-f9377c2ecfba]
>> 2011-08-03 10:03:43.728232 [INFO] mod_dialplan_xml.c:336 Processing 1002
>> <1002>->00919876543210 in context default
>> 2011-08-03 10:03:43.728232 [INFO] switch_core_session.c:1281
>> sofia/internal/1002 at 172.16.10.211 setting session heartbeat to 60
>> second(s).
>> 2011-08-03 10:03:43.728232 [ERR] mod_event_socket.c:457 Socket Error!
>> 2011-08-03 10:03:43.728232 [NOTICE] switch_core_state_machine.c:189
>> sofia/internal/1002 at 172.16.10.211 has executed the last dialplan
>> instruction, hanging up.
>> 2011-08-03 10:03:43.728232 [NOTICE] switch_core_state_machine.c:191 Hangup
>> sofia/internal/1002 at 172.16.10.211 [CS_EXECUTE] [NORMAL_CLEARING]
>> 2011-08-03 10:03:43.728232 [NOTICE] switch_core_session.c:1347 Session 24
>> (sofia/internal/1002 at 172.16.10.211) Ended
>> 2011-08-03 10:03:43.728232 [NOTICE] switch_core_session.c:1349 Close
>> Channel sofia/internal/1002 at 172.16.10.211 [CS_DESTROY]
>>
>> ************************************************************************************************************************************************************
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110803/b718ebe7/attachment-0001.html 

From sascha.daniels at amooma.de  Wed Aug  3 13:04:24 2011
From: sascha.daniels at amooma.de (Sascha Daniels)
Date: Wed, 03 Aug 2011 11:04:24 +0200
Subject: [Freeswitch-users] Limit number of voicemails
In-Reply-To: <05a901cc511a$775ee8d0$661cba70$@com>
References: <4E37CA90.4090806@amooma.de> <05a901cc511a$775ee8d0$661cba70$@com>
Message-ID: <4E390F18.8060604@amooma.de>

Hi.

Am 02.08.2011 15:45, schrieb Robert Huddleston:
> Actually http://wiki.freeswitch.org/wiki/Mod_voicemail#vm-disk-quota
>
> In the manual...
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sascha
> Daniels
>
> I know that I can limit the length. That is the first step.
>
That is the part I allready know. Limit the length.

Greets Sascha



-- 
AMOOMA GmbH - Bachstr. 124 - 56566 Neuwied  -->  http://www.amooma.de
Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister Montabaur B14998

B?cher: http://das-asterisk-buch.de - http://ruby-auf-schienen.de



From vkozak at abisoft.spb.ru  Wed Aug  3 13:11:19 2011
From: vkozak at abisoft.spb.ru (Kozak Vladimir)
Date: Wed, 3 Aug 2011 13:11:19 +0400
Subject: [Freeswitch-users] EXCHANGE_ROUTING_ERROR after ~70 transfer to
	dialplan extebsion
References: <576190BDB2B847A180F79DF6B311A49C@abisoft.biz>
	<CAKzWOxVeG7HTNmUsH0e6iRxgAd45u8WX=arDaq6B9B6gfgOoAA@mail.gmail.com>
Message-ID: <0F5E8341E87D4818894C7CA1B13A24B2@abisoft.biz>

It isn't looping. It's implimentation IVR menu. Our external system send commands to FS (transfer to suitable extension and other) for play files, say text, get dtmf, ...
And it's possible a lot of transfers.
I try increase max_forwards header. Exist something else?



----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Tuesday, August 02, 2011 11:13 PM
  Subject: Re: [Freeswitch-users] EXCHANGE_ROUTING_ERROR after ~70 transfer to dialplan extebsion


  If you have 70 hops through the dialplan then it's probably because you have a routing loop. The 70 hops acts as a "circuit breaker" to keep the call from looping forever.


  The next step for you is to determine why your call is looping through the dialplan so many times. is that truly needed?


  -MC


  2011/8/2 Kozak Vladimir <vkozak at abisoft.spb.ru>

    Hi all.
    FreeSWITCH Version 1.0.head (git-9ff8f53 2011-05-03 12-13-52 -0400)

    I have problem with transfer channel to FS extension.
    I send from external system to FS command as
       "SendMsg " + uniqueId + "\n" +
       "call-command: execute\n" +
       "execute-app-name: transfer\n" +
       "execute-app-arg: " + destination + "\n\n";

    after ~70 correct transfer actions FS send BYE to my phone and send HANGUP_EVENT with cause EXCHANGE_ROUTING_ERROR to my system. (FS log attached)

    from FS loggs:
    2011-08-09 04:48:38.525293 [DEBUG] mod_sofia.c:457 Channel sofia/internal/1000 at vkozak.starpoundtech.net hanging up, cause: EXCHANGE_ROUTING_ERROR
    2011-08-09 04:48:38.525293 [DEBUG] mod_sofia.c:500 Sending BYE to sofia/internal/1000 at vkozak.starpoundtech.net

    what for does limitation of transfer operations exist?
    how can I avoid this issue?
    is't possible to increase number of hops for transfer command?

    _______________________________________________
    Join us at ClueCon 2011, Aug 9-11, Chicago
    http://www.cluecon.com 877-7-4ACLUE

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






------------------------------------------------------------------------------


  _______________________________________________
  Join us at ClueCon 2011, Aug 9-11, Chicago
  http://www.cluecon.com 877-7-4ACLUE

  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110803/38897c44/attachment.html 

From sascha.daniels at amooma.de  Wed Aug  3 13:14:07 2011
From: sascha.daniels at amooma.de (Sascha Daniels)
Date: Wed, 03 Aug 2011 11:14:07 +0200
Subject: [Freeswitch-users] Limit number of voicemails
In-Reply-To: <CAKzWOxXj1Cn1yL9d4CwxCZxHRKXdTcmVfKw00xh2VuH8CaxJbw@mail.gmail.com>
References: <4E37CA90.4090806@amooma.de> <05a901cc511a$775ee8d0$661cba70$@com>
	<CAKzWOxXj1Cn1yL9d4CwxCZxHRKXdTcmVfKw00xh2VuH8CaxJbw@mail.gmail.com>
Message-ID: <4E39115F.9030100@amooma.de>

Hi together.

Am 02.08.2011 21:07, schrieb Michael Collins:
> This is actually better than limiting the number of messages since you
> can control actual disk space. (One REALLY long message will take more
> disk space than five really short ones.)
>

I need to do both, because I have to guarantee that a User can not use
more X mb for voicemail.

If I only limit the length, a bad boy could leave thousands of messages ...

> However, in the interests of actually answering the question, and
> demonstrating that with FreeSWITCH anything is possible, here is a
> simple dialplan trick that will allow you to limit the number messages:
>
> <extension name="check vm box limit" continue="true">
>   <condition field="destination_number" expression="^(10[01][0-9])$"
> break="never">
>     <action application="set" data="dialed_extension=$1"/>
>     <action application="set" data="dest_user=$1@${domain}"
> inline="true"/>
>     <action application="set" data="box_count=${vm_boxcount
> ${dest_user}}" inline="true">
>     <action application="log" data="INFO User ${dest_user} has
> ${box_count} message(s)"/>
>   </condition>
>
>   <condition field="${expr below(${boxcount},{$max_count})"
> expression="^1$">
>     <action application="voicemail" data="${domain} ${dialed_extension}"/>
>     <anti-action application="playback"
> data="voicemail/vm-mailbox_full.wav"/>
>   </condition>
> </extension>

That's what I was looking for!

Thanks a lot.

I will give feedback, as soon as possible.

Greets

                Sascha

-- 
AMOOMA GmbH - Bachstr. 124 - 56566 Neuwied  -->  http://www.amooma.de
Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister Montabaur B14998

B?cher: http://das-asterisk-buch.de - http://ruby-auf-schienen.de



From cmcureau at gmail.com  Wed Aug  3 18:29:45 2011
From: cmcureau at gmail.com (Chris Cureau)
Date: Wed, 3 Aug 2011 09:29:45 -0500
Subject: [Freeswitch-users] Commercial support
Message-ID: <6315100417238183580@unknownmsgid>

What's the best way to get paid support for freeswitch either on a per
incident or contract basis?


From leonardo.bidinoto at voicetechnology.com.br  Wed Aug  3 18:37:46 2011
From: leonardo.bidinoto at voicetechnology.com.br (Leonardo P. Bidinoto)
Date: Wed, 3 Aug 2011 11:37:46 -0300
Subject: [Freeswitch-users] FS Status Command Description
Message-ID: <CALV5c5PiAy341ii+Lusb=TFUcgrJ-Wt=6Xp9=9ga=FG=EsQaRg@mail.gmail.com>

Hello Dan,

I know this might be a noobie question, but .......

*fs_cli -x'status'*

UP 0 years, 1 day, 1 hour, 2 minutes, 45 seconds, 167 milliseconds, 852
microseconds
FreeSWITCH is ready
1377 session(s) since startup
153 session(s) 0/30 ==> *what means this zero before the "/30"? sometimes is
changing to 1, like this ( 153 session(s) 1/30 )*
1000 session(s) max
min idle cpu 0.00/76.00


-- 
Leonardo Pires Bidinoto
Voice Technology
www.voicetechnology.com.br
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110803/93429c64/attachment.html 

From jeff at jefflenk.com  Wed Aug  3 18:42:10 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 3 Aug 2011 07:42:10 -0700 (PDT)
Subject: [Freeswitch-users] (no subject)
In-Reply-To: <CAKzWOxV09m=+rzZueogrs8poLdto9PgzOCRynO0SaBb8Y_0N-Q@mail.gmail.com>
References: <1312313437.77623.YahooMailNeo@web161003.mail.bf1.yahoo.com>
	<CAKzWOxW40E4TSJ0p865iUiVLMd4PuROOx+UdN2KunBwrbsdW1Q@mail.gmail.com>
	<1312316397.65960.YahooMailNeo@web161007.mail.bf1.yahoo.com>
	<CAKzWOxUc=4rfTgnAxpRd7EFUA3F9Q8iBZMneNoTzyjFNeLCv3Q@mail.gmail.com>
	<1312317497.83886.YahooMailNeo@web161017.mail.bf1.yahoo.com>
	<CAKzWOxVPE13SgeOwPZE3mqOVfoxqzwkQN-8Fc3z_Pc8neYnV9A@mail.gmail.com>
	<1312325304.39236.YahooMailNeo@web161019.mail.bf1.yahoo.com>
	<CAKzWOxVjWde-1isxbrbjOhFMrSwQUzsx7H6yHoXLedbe2b0N3Q@mail.gmail.com>
	<1312327583.59336.YahooMailNeo@web161019.mail.bf1.yahoo.com>
	<CAKzWOxV09m=+rzZueogrs8poLdto9PgzOCRynO0SaBb8Y_0N-Q@mail.gmail.com>
Message-ID: <1312382530544-6648986.post@n2.nabble.com>

Sam is your trace log including everything? sofia siptrace global on?

It seems like there a things missing from the other profile?

Also please don't open discussions here and on Jira. There is only so much
volunteers can do in there spare time.


--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/no-subject-tp6645805p6648986.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From freeswitch-list at puzzled.xs4all.nl  Wed Aug  3 18:47:11 2011
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Wed, 03 Aug 2011 16:47:11 +0200
Subject: [Freeswitch-users] Commercial support
In-Reply-To: <6315100417238183580@unknownmsgid>
References: <6315100417238183580@unknownmsgid>
Message-ID: <4E395F6F.9060301@puzzled.xs4all.nl>

On 08/03/2011 04:29 PM, Chris Cureau wrote:
> What's the best way to get paid support for freeswitch either on a per
> incident or contract basis?

Email consulting at freeswitch.org or call 1-877-742-2583.

Regards,
Patrick


From jeff at jefflenk.com  Wed Aug  3 18:48:22 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 3 Aug 2011 07:48:22 -0700 (PDT)
Subject: [Freeswitch-users] FS Status Command Description
In-Reply-To: <CALV5c5PiAy341ii+Lusb=TFUcgrJ-Wt=6Xp9=9ga=FG=EsQaRg@mail.gmail.com>
References: <CALV5c5PiAy341ii+Lusb=TFUcgrJ-Wt=6Xp9=9ga=FG=EsQaRg@mail.gmail.com>
Message-ID: <1312382902614-6649009.post@n2.nabble.com>

153 session(s) 0/30 
The zero here is the last snapshot value of how many sessions per second
were occuring.
The thirty is the maximum allowed.

Please update the Wiki if this was not clear to you. Thanks

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-Status-Command-Description-tp6648983p6649009.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From rahulkrishna222 at gmail.com  Wed Aug  3 13:10:37 2011
From: rahulkrishna222 at gmail.com (rahulkrishna)
Date: Wed, 3 Aug 2011 02:10:37 -0700 (PDT)
Subject: [Freeswitch-users] Freeswitch Hangup Event Tracking
Message-ID: <CAALn-hO8+t2P-=cvFSvfw-EZxje+KuejQeiM2r-Vx8pgxoYpvA@mail.gmail.com>

Hi ... All .am Rahul..
am new to freeswitch programming.... can you help me to find out a solution
for the following Task.
My question is that. is When a user A make call to user B . and after the
conversation the User A will end the call..
I want to know that is it is possible make a delay of 10sec for the HANGUP.

and is it is possible to add some extra task at that time. need the call
still alive

Is it is possible to track the Hangup Event.. if yes please help me.
-- 
Rahul


--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-Hangup-Event-Tracking-tp6648025p6648025.html
Sent from the freeswitch-users mailing list archive at Nabble.com.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110803/ee3c5e3a/attachment-0001.html 

From msc at freeswitch.org  Wed Aug  3 19:15:59 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 3 Aug 2011 08:15:59 -0700
Subject: [Freeswitch-users] Help setting up SIP reg
In-Reply-To: <CAE=3qObd_GZohA=zbWgGPx5bU1GLNsGkjnZPNeTf8NAxjZnVOw@mail.gmail.com>
References: <CAE=3qOYPpTuRTvq7RTJeUgzsCtvrfCjMiKFGu1fg0PHyZRE+-g@mail.gmail.com>
	<CAFiqYu=c8mY5BMWPumy1CgZHZ81kMRPgoNiF-6mjFAkW1q30fA@mail.gmail.com>
	<CAE=3qOZxwERT3CQEFHetO_o2s=aQUwYZ54TmMRPx9eYD=A7NoQ@mail.gmail.com>
	<CAKzWOxUEsGUKojF-vvstL6FvcDMhePqYGdXEQXQ4x-EJ81Bk6w@mail.gmail.com>
	<CAE=3qOacZh50iepxJKJD5UvtesxZAsN7RnPKNe1zdLO9VmFsuA@mail.gmail.com>
	<CAE=3qOZsNuiimFBZJTf4SbPRA8bRq_hJfPUunEnT8rxZj3qTnA@mail.gmail.com>
	<CAE=3qObVuLOhp2GD29Da-GfzB_qkbkyDP4kPmRxjdbJi7XeirQ@mail.gmail.com>
	<CAKzWOxUyZoBS5Evp4b61vP8fm64Lef0n01ti3KmVraQO_BqYtA@mail.gmail.com>
	<CAE=3qObsDBDTsKY2ycK_NierbS_A+-f896yMW4OArTkgauYgRA@mail.gmail.com>
	<CAKzWOxXYi-yO9DNDw1rHCduKq=h9q=BddFmn2dKuqALKb=t1fA@mail.gmail.com>
	<CAE=3qObVbDQi6g1m80WLtG5MhyosV2YYnqmOoausiz-yC+NmQQ@mail.gmail.com>
	<CAKzWOxVC51tge2o1xEAW2NTvA2ewUD-XnanLRgRjsymtrRT-Og@mail.gmail.com>
	<CAE=3qObd_GZohA=zbWgGPx5bU1GLNsGkjnZPNeTf8NAxjZnVOw@mail.gmail.com>
Message-ID: <CAKzWOxWTgw0c=ucZcYEHhmD2TewTmTsSFwjArB0LyA-t4gO5ag@mail.gmail.com>

Yes, the error is "UNALLOCATED NUMBER" which means a phone number that is
not in service or possibly a wrong number format. This means that you are
actually talking to your provider now which is a good thing. Next what you
need to do is ask the provider to analyze what you're sending to see if it
is in the correct format.

-MC

On Wed, Aug 3, 2011 at 12:41 AM, lloyd thomas <lloydie.t at gmail.com> wrote:

> Getting a little further, but calls are still failing.
> Not sure why something to do with 404 error. could it be wrong number
> format?
>
> changed dialplan to <action application="bridge"
> data="sofia/external/01869$1 at 80.40.150.150:5060"/> just in case the port
> number was incorrect.
>
>
>
> On 2 August 2011 19:16, Michael Collins <msc at freeswitch.org> wrote:
>
>> Lloyd,
>>
>> The gateway you created I think has a few issues. Let's have you bypass it
>> altogether with a single change to your dialplan. Change your bridge line to
>> this:
>>
>> <action application="bridge" data="sofia/external/01869$1 at 80.40.150.150
>> "/>
>>
>> This bypasses the gateway and sends the call straight out the external
>> profile. (You could also send it out the internal profile.)
>>
>> Try that and see what happens. If you have issues then do the usual
>> console log and siptrace and put it into pastebin.freeswitch.org. Be sure
>> to choose "FreeSWITCH Log" for the syntax highlighting.
>>
>> -MC
>>
>>
>> On Tue, Aug 2, 2011 at 3:41 AM, lloyd thomas <lloydie.t at gmail.com> wrote:
>>
>>> Just did a test, but no joy. I suspect I may have to dispense with the
>>> gateway settings and just bridge straight from the dial plan, but it is just
>>> a guess.
>>>
>>>
>>> dialplan
>>> ----------------------------------------
>>> <extension name="Local Dial">
>>>    <condition field="destination_number" expression="^9([2-9][0-9]{5})$">
>>>      <action application="bridge"
>>> data="sofia/gateway/phisys-2circles/01869$1"/>
>>>    </condition>
>>> </extension>
>>>
>>> gateway
>>> ---------------------------------------
>>> <include>
>>>   <gateway name="phisys-2circles">
>>>     <param name="from-domain" value="80.40.150.150"/>
>>>     <param name="proxy" value="80.40.150.150"/>
>>>     <param name="sip-port" value="5060"/>
>>>   </gateway>
>>> </include>
>>>
>>>
>>>
>>> errors
>>> ------------------------------------
>>> 2011-08-02 11:32:42.056302 [DEBUG] mod_dptools.c:1059 sofia/internal/
>>> 200 at phisys.tele.phi.co.uk SET [RFC2822_DATE]=[Tue, 02 Aug 2011 11:32:42
>>> +0100]
>>> EXECUTE sofia/internal/200 at phisys.tele.phi.co.ukbridge(sofia/gateway/phisys-2circles/01869321110)
>>> 2011-08-02 11:32:42.073667 [ERR] mod_sofia.c:3940 Invalid Gateway
>>> 2011-08-02 11:32:42.073667 [NOTICE] mod_sofia.c:4282 Close Channel N/A
>>> [CS_NEW]
>>> 2011-08-02 11:32:42.076523 [DEBUG] switch_core_state_machine.c:452 ()
>>> Running State Change CS_DESTROY
>>> 2011-08-02 11:32:42.079574 [DEBUG] switch_core_state_machine.c:462 (N/A)
>>> State DESTROY
>>> 2011-08-02 11:32:42.079574 [DEBUG] mod_sofia.c:362 N/A SOFIA DESTROY
>>> 2011-08-02 11:32:42.081946 [DEBUG] switch_core_state_machine.c:462 (N/A)
>>> State DESTROY going to sleep
>>> 2011-08-02 11:32:42.084166 [ERR] switch_ivr_originate.c:2640 Cannot
>>> create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]
>>> 2011-08-02 11:32:42.085649 [DEBUG] switch_ivr_originate.c:3506 Originate
>>> Resulted in Error Cause: 28 [INVALID_NUMBER_FORMAT]
>>> 2011-08-02 11:32:42.087275 [INFO] mod_dptools.c:2623 Originate Failed.
>>> Cause: INVALID_NUMBER_FORMAT
>>> 2011-08-02 11:32:42.088652 [DEBUG] switch_channel.c:2559 (sofia/internal/
>>> 200 at phisys.tele.phi.co.uk) Callstate Change RINGING -> HANGUP
>>> 2011-08-02 11:32:42.093248 [NOTICE] mod_dptools.c:2686 Hangup
>>> sofia/internal/200 at phisys.tele.phi.co.uk [CS_EXECUTE]
>>> [INVALID_NUMBER_FORMAT]
>>> 2011-08-02 11:32:42.096925 [DEBUG] switch_channel.c:2575 Send signal
>>> sofia/internal/200 at phisys.tele.phi.co.uk [KILL]
>>>
>>>
>>> On 1 August 2011 20:50, Michael Collins <msc at freeswitch.org> wrote:
>>>
>>>> Okay, so what happens when you dial out? Sorry, it's been a few days and
>>>> I don't recall where we left off. Be sure to include console log w/ siptrace
>>>> on pastebin.freeswitch.org.
>>>>
>>>> -MC
>>>>
>>>>
>>>> On Mon, Aug 1, 2011 at 12:35 PM, lloyd thomas <lloydie.t at gmail.com>wrote:
>>>>
>>>>> I think they have my IP on a white list.
>>>>>
>>>>>
>>>>> On 1 August 2011 16:24, Michael Collins <msc at freeswitch.org> wrote:
>>>>>
>>>>>> Do they challenge you (digest auth) or do they have your IP address on
>>>>>> a white list? That's a critical piece of information that only your provider
>>>>>> can supply.
>>>>>>
>>>>>> -MC
>>>>>>
>>>>>>
>>>>>> On Fri, Jul 29, 2011 at 9:31 PM, lloyd thomas <lloydie.t at gmail.com>wrote:
>>>>>>
>>>>>>> OK Inbound working with:
>>>>>>>
>>>>>>>   <!--2circles -->
>>>>>>>     <list name="domains" default="allow">
>>>>>>>       <node type="allow" cidr="80.40.150.150/32"/>
>>>>>>>     </list>
>>>>>>>
>>>>>>> Just need to sort outbound.
>>>>>>>
>>>>>>>
>>>>>>> On 30 July 2011 04:59, lloyd thomas <lloydie.t at gmail.com> wrote:
>>>>>>>
>>>>>>>> Hi, dialling in produces the following error.
>>>>>>>>
>>>>>>>> 2011-07-30 04:56:07.818936 [DEBUG] sofia.c:6517 IP 80.40.150.150
>>>>>>>> Rejected by acl "domains". Falling back to Digest auth.
>>>>>>>> 2011-07-30 04:56:07.826367 [WARNING] sofia_reg.c:1246 SIP auth
>>>>>>>> challenge (INVITE) on sofia profile 'internal' for
>>>>>>>> [01869******@172.16.XXX.XXX] from ip 80.40.150.150
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> On 30 July 2011 04:34, lloyd thomas <lloydie.t at gmail.com> wrote:
>>>>>>>>
>>>>>>>>> I am registering with a them. I could not find suitable example in
>>>>>>>>> http://wiki.freeswitch.org/wiki/SIP_Provider_Examples which
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On 29 July 2011 21:57, Michael Collins <msc at freeswitch.org> wrote:
>>>>>>>>>
>>>>>>>>>> Are you registering with the provider or are they registering with
>>>>>>>>>> you? If they register with you then a user example is appropriate. If you
>>>>>>>>>> are registering with them then all you need is a gateway configured.
>>>>>>>>>> -MC
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> On Fri, Jul 29, 2011 at 1:40 PM, lloyd thomas <
>>>>>>>>>> lloydie.t at gmail.com> wrote:
>>>>>>>>>>
>>>>>>>>>>> Sorry, example is not clear to me.
>>>>>>>>>>> I don't understand why a user config is relevant to sip
>>>>>>>>>>> registration for a provider.
>>>>>>>>>>> An example will help me more. Maybe CIDR attribute in a
>>>>>>>>>>> sip_profile gateway could help.
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> On 29 July 2011 19:55, Steven Ayre <steveayre at gmail.com> wrote:
>>>>>>>>>>>
>>>>>>>>>>>> Look at the cidr attribute in the user directory to authenticate
>>>>>>>>>>>> by IP:
>>>>>>>>>>>> http://wiki.freeswitch.org/wiki/Acl#Users
>>>>>>>>>>>>
>>>>>>>>>>>> -Steve
>>>>>>>>>>>>
>>>>>>>>>>>> On 29 July 2011 19:38, lloyd thomas <lloydie.t at gmail.com>wrote:
>>>>>>>>>>>>
>>>>>>>>>>>>> *Hi I need a little help setting up a SIP registration for a
>>>>>>>>>>>>> provider that does not use auth.*
>>>>>>>>>>>>>
>>>>>>>>>>>>> *All I have is info below.*
>>>>>>>>>>>>> **
>>>>>>>>>>>>>
>>>>>>>>>>>>> *
>>>>>>>>>>>>> *
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> SBC/Proxy IP:        80.40.150.150:5060
>>>>>>>>>>>>>
>>>>>>>>>>>>> Authentication:      Trusted IP ? 88.221.85.33
>>>>>>>>>>>>>
>>>>>>>>>>>>> Assigned DDI:        01869******, 01869******
>>>>>>>>>>>>>
>>>>>>>>>>>>> DTMF Method:         RFC2833
>>>>>>>>>>>>>
>>>>>>>>>>>>> Status:              Live
>>>>>>>>>>>>>
>>>>>>>>>>>>> No. of trunks:       2x
>>>>>>>>>>>>>
>>>>>>>>>>>>> Session Timer:       1800
>>>>>>>>>>>>>
>>>>>>>>>>>>> Profile*:*             Generic (35060)
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> Apparently the following is used for *
>>>>>>>>>>>>>
>>>>>>>>>>>>> [vibe]
>>>>>>>>>>>>>
>>>>>>>>>>>>> type = friend
>>>>>>>>>>>>>
>>>>>>>>>>>>> host = 80.40.150.150
>>>>>>>>>>>>>
>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>>>>
>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>>>
>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>>
>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110803/fd1276c3/attachment-0001.html 

From msc at freeswitch.org  Wed Aug  3 19:20:45 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 3 Aug 2011 08:20:45 -0700
Subject: [Freeswitch-users] Freeswitch Hangup Event Tracking
In-Reply-To: <CAALn-hO8+t2P-=cvFSvfw-EZxje+KuejQeiM2r-Vx8pgxoYpvA@mail.gmail.com>
References: <CAALn-hO8+t2P-=cvFSvfw-EZxje+KuejQeiM2r-Vx8pgxoYpvA@mail.gmail.com>
Message-ID: <CAKzWOxUR9xLLK_njx7Rtrknxd=fOcKmVBnWERjFXySE3uSVjGQ@mail.gmail.com>

If User A hangs up the call then the call is over - there's nothing you can
do about that. However, you can use api_hangup_hook channel variable to do
stuff after the call is over.

What problem are you solving by delaying the hangup for 10 seconds? That
sounds unusual.

-MC

On Wed, Aug 3, 2011 at 2:10 AM, rahulkrishna <rahulkrishna222 at gmail.com>wrote:

> Hi ... All .am Rahul..
> am new to freeswitch programming.... can you help me to find out a solution
> for the following Task.
> My question is that. is When a user A make call to user B . and after the
> conversation the User A will end the call..
> I want to know that is it is possible make a delay of 10sec for the HANGUP.
>
> and is it is possible to add some extra task at that time. need the call
> still alive
>
> Is it is possible to track the Hangup Event.. if yes please help me.
> --
> Rahul
>
> ------------------------------
> View this message in context: Freeswitch Hangup Event Tracking<http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-Hangup-Event-Tracking-tp6648025p6648025.html>
> Sent from the freeswitch-users mailing list archive<http://freeswitch-users.2379917.n2.nabble.com/>at Nabble.com.
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110803/67cd21bd/attachment.html 

From roger.castaldo at gmail.com  Wed Aug  3 19:20:56 2011
From: roger.castaldo at gmail.com (Roger Castaldo)
Date: Wed, 3 Aug 2011 11:20:56 -0400
Subject: [Freeswitch-users] Freeswitch Hangup Event Tracking
In-Reply-To: <CAALn-hO8+t2P-=cvFSvfw-EZxje+KuejQeiM2r-Vx8pgxoYpvA@mail.gmail.com>
References: <CAALn-hO8+t2P-=cvFSvfw-EZxje+KuejQeiM2r-Vx8pgxoYpvA@mail.gmail.com>
Message-ID: <CAAYf2LuJhRtq-x=p_Mn1FxyF++fMAJb3iuMbw=DT6r907Zk5CQ@mail.gmail.com>

I am not sure about delaying the hangup, however using the event socket you
can track the channel events which includes the hangup, please refer to the
wiki http://wiki.freeswitch.org/wiki/Event_list for more information on the
events trackable in the event socket.

On Wed, Aug 3, 2011 at 5:10 AM, rahulkrishna <rahulkrishna222 at gmail.com>wrote:

> Hi ... All .am Rahul..
> am new to freeswitch programming.... can you help me to find out a solution
> for the following Task.
> My question is that. is When a user A make call to user B . and after the
> conversation the User A will end the call..
> I want to know that is it is possible make a delay of 10sec for the HANGUP.
>
> and is it is possible to add some extra task at that time. need the call
> still alive
>
> Is it is possible to track the Hangup Event.. if yes please help me.
> --
> Rahul
>
> ------------------------------
> View this message in context: Freeswitch Hangup Event Tracking<http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-Hangup-Event-Tracking-tp6648025p6648025.html>
> Sent from the freeswitch-users mailing list archive<http://freeswitch-users.2379917.n2.nabble.com/>at Nabble.com.
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110803/fa5f8420/attachment.html 

From msc at freeswitch.org  Wed Aug  3 19:25:40 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 3 Aug 2011 08:25:40 -0700
Subject: [Freeswitch-users] Calls not established : See the error details
In-Reply-To: <CA+P_fE9QZ6e2N3aixe+V4n5sbgJJ3cHts6nbKg4oM0eEbuYmkw@mail.gmail.com>
References: <CA+P_fE8Fd5=2c4b48npp-27aQGGJN1vhFRE8FTvvgp+4xx0CXw@mail.gmail.com>
	<CAFiqYun8ppGVpngCb78QCi29EsxM7=EouMWPc2D-j+sUxz2dtw@mail.gmail.com>
	<CA+P_fE9QZ6e2N3aixe+V4n5sbgJJ3cHts6nbKg4oM0eEbuYmkw@mail.gmail.com>
Message-ID: <CAKzWOxVhv1BzX188pf_APUZDF90WH6Uk=X9R=fhVg47TuCHwDA@mail.gmail.com>

Yes, Plivo uses the event socket. This sounds very much like an issue with
the Plivo stuff not running/responding.  Recommend checking plivo.org or
#plivo IRC channel.

-MC

On Wed, Aug 3, 2011 at 1:35 AM, Dhairya Vora <dhairya.blogs at gmail.com>wrote:

> One thing is making a confusion in my mind. Can plivo (installed on
> freeswitch) be the problem ?
> According to me, it would not the be reason. But still just asking for the
> clarification.
>
>
>
>
> On Wed, Aug 3, 2011 at 1:24 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> The debug-level log would be helpful.
>>
>> What does your dialplan look like? It looks like you're executing a
>> dialplan extension that's trying to connect somewhere with the event socket
>> (to have the logic executed externally) but failing to connect to a ESL
>> server:
>>
>> mod_event_socket.c:457 Socket Error!
>>
>> That suggests to me your dialplan might not be doing what you expect it
>> to.
>>
>> -Steve
>>
>>
>>
>> On 3 August 2011 06:10, Dhairya Vora <dhairya.blogs at gmail.com> wrote:
>>
>>> I don't understand why two users (registered directly to freeswitch) are
>>> unable to connect each other?
>>>
>>>
>>> When a registered user 1001 calls registered user 1002, call fails giving
>>> this error
>>>
>>> ************************************************************************************************************************************************************
>>> freeswitch at localhost.localdomain> 2011-08-03 10:07:01.941089 [WARNING]
>>> sofia_reg.c:1337 SIP auth challenge (INVITE) on sofia profile 'internal' for
>>> [1002 at 172.16.10.211] from ip 172.16.10.248
>>> 2011-08-03 10:07:01.941089 [NOTICE] switch_channel.c:897 New Channel
>>> sofia/internal/1000 at 172.16.10.211 [0bf31c06-8379-4831-88d7-a7df3c4ecd49]
>>> 2011-08-03 10:07:01.941089 [INFO] mod_dialplan_xml.c:336 Processing 1000
>>> <1000>->1002 in context default
>>> 2011-08-03 10:07:01.941089 [INFO] switch_core_session.c:1281
>>> sofia/internal/1000 at 172.16.10.211 setting session heartbeat to 60
>>> second(s).
>>> 2011-08-03 10:07:01.941089 [ERR] mod_event_socket.c:457 Socket Error!
>>> 2011-08-03 10:07:01.941089 [NOTICE] switch_core_state_machine.c:189
>>> sofia/internal/1000 at 172.16.10.211 has executed the last dialplan
>>> instruction, hanging up.
>>> 2011-08-03 10:07:01.941089 [NOTICE] switch_core_state_machine.c:191
>>> Hangup sofia/internal/1000 at 172.16.10.211 [CS_EXECUTE] [NORMAL_CLEARING]
>>> 2011-08-03 10:07:01.961094 [NOTICE] switch_core_session.c:1347 Session 25
>>> (sofia/internal/1000 at 172.16.10.211) Ended
>>> 2011-08-03 10:07:01.961094 [NOTICE] switch_core_session.c:1349 Close
>>> Channel sofia/internal/1000 at 172.16.10.211 [CS_DESTROY]
>>>
>>> ************************************************************************************************************************************************************
>>>
>>>
>>>
>>> When I make an outbound call to my mobile, calls are not going through
>>> the custom gateway
>>>
>>> ************************************************************************************************************************************************************
>>> freeswitch at localhost.localdomain> 2011-08-03 10:03:43.608234 [WARNING]
>>> sofia_reg.c:1337 SIP auth challenge (INVITE) on sofia profile 'internal' for
>>> [00919876543210 at 172.16.10.211] from ip 172.16.10.213
>>> 2011-08-03 10:03:43.728232 [NOTICE] switch_channel.c:897 New Channel
>>> sofia/internal/1002 at 172.16.10.211 [334906be-9cb0-44f2-9660-f9377c2ecfba]
>>> 2011-08-03 10:03:43.728232 [INFO] mod_dialplan_xml.c:336 Processing 1002
>>> <1002>->00919876543210 in context default
>>> 2011-08-03 10:03:43.728232 [INFO] switch_core_session.c:1281
>>> sofia/internal/1002 at 172.16.10.211 setting session heartbeat to 60
>>> second(s).
>>> 2011-08-03 10:03:43.728232 [ERR] mod_event_socket.c:457 Socket Error!
>>> 2011-08-03 10:03:43.728232 [NOTICE] switch_core_state_machine.c:189
>>> sofia/internal/1002 at 172.16.10.211 has executed the last dialplan
>>> instruction, hanging up.
>>> 2011-08-03 10:03:43.728232 [NOTICE] switch_core_state_machine.c:191
>>> Hangup sofia/internal/1002 at 172.16.10.211 [CS_EXECUTE] [NORMAL_CLEARING]
>>> 2011-08-03 10:03:43.728232 [NOTICE] switch_core_session.c:1347 Session 24
>>> (sofia/internal/1002 at 172.16.10.211) Ended
>>> 2011-08-03 10:03:43.728232 [NOTICE] switch_core_session.c:1349 Close
>>> Channel sofia/internal/1002 at 172.16.10.211 [CS_DESTROY]
>>>
>>> ************************************************************************************************************************************************************
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110803/9d351141/attachment-0001.html 

From cmcureau at gmail.com  Wed Aug  3 19:28:02 2011
From: cmcureau at gmail.com (Chris Cureau)
Date: Wed, 3 Aug 2011 10:28:02 -0500
Subject: [Freeswitch-users] Cancel sent on outgoing call completion
Message-ID: <5185360865201181730@unknownmsgid>

I'm hoping that someone might be able to figure this one out...I've
tried everything that I know of!

On an outbound call, I can see the buildup and the connection to the
gateway. The calked party begins to ring very briefly, and then it
looks like a CANCEL is being sent. I'm pretty sure it isn't isolated
to a phone since all my phones are doing it now.

This is using current git as of last night.

http://pastebin.freeswitch.org/16953

Any hints appreciated!
--
Chris


From Joshua.Foshee at LogixCom.com  Wed Aug  3 19:29:03 2011
From: Joshua.Foshee at LogixCom.com (Joshua Foshee)
Date: Wed, 3 Aug 2011 10:29:03 -0500
Subject: [Freeswitch-users] Grandstreams and TLS
Message-ID: <06502C073AD9394AADB3CA7FD94931BC077906FA@okc1x1.Logixcom.com>

Has anyone got the GXV3175 or GXV3140 to work with TLS? I have tried
TLSV1 and SSL23 on the TLS but neither one of them will get to
authenticate correctly. If you have can you post the Freeswitch TLS
config and the phone settings that you got to work?

 

Thanks,
Josh

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110803/7b453c49/attachment.html 

From ijurado at econcept.es  Wed Aug  3 19:49:40 2011
From: ijurado at econcept.es (Isaac Jurado)
Date: Wed, 3 Aug 2011 17:49:40 +0200
Subject: [Freeswitch-users] Hangup hooks on B legs
Message-ID: <CABj_V-mOmm05ycYJZRPaeePeVzB4LWdTfzjNXwM3O=Kjmrn5tQ@mail.gmail.com>

Hi,

I've been playing with the "api_hangup_hook" dialplan setting.  However,
it only seems to be executing when the A-leg (originator) hangs up.  Is
there any way to execute such a hook from all the potential B-legs of
the call?

Thanks in advance.

-- 
Isaac Jurado
Internet Busines Solutions eConcept


From lloydie.t at gmail.com  Wed Aug  3 19:58:23 2011
From: lloydie.t at gmail.com (lloyd thomas)
Date: Wed, 3 Aug 2011 16:58:23 +0100
Subject: [Freeswitch-users] Help setting up SIP reg
In-Reply-To: <CAKzWOxWTgw0c=ucZcYEHhmD2TewTmTsSFwjArB0LyA-t4gO5ag@mail.gmail.com>
References: <CAE=3qOYPpTuRTvq7RTJeUgzsCtvrfCjMiKFGu1fg0PHyZRE+-g@mail.gmail.com>
	<CAFiqYu=c8mY5BMWPumy1CgZHZ81kMRPgoNiF-6mjFAkW1q30fA@mail.gmail.com>
	<CAE=3qOZxwERT3CQEFHetO_o2s=aQUwYZ54TmMRPx9eYD=A7NoQ@mail.gmail.com>
	<CAKzWOxUEsGUKojF-vvstL6FvcDMhePqYGdXEQXQ4x-EJ81Bk6w@mail.gmail.com>
	<CAE=3qOacZh50iepxJKJD5UvtesxZAsN7RnPKNe1zdLO9VmFsuA@mail.gmail.com>
	<CAE=3qOZsNuiimFBZJTf4SbPRA8bRq_hJfPUunEnT8rxZj3qTnA@mail.gmail.com>
	<CAE=3qObVuLOhp2GD29Da-GfzB_qkbkyDP4kPmRxjdbJi7XeirQ@mail.gmail.com>
	<CAKzWOxUyZoBS5Evp4b61vP8fm64Lef0n01ti3KmVraQO_BqYtA@mail.gmail.com>
	<CAE=3qObsDBDTsKY2ycK_NierbS_A+-f896yMW4OArTkgauYgRA@mail.gmail.com>
	<CAKzWOxXYi-yO9DNDw1rHCduKq=h9q=BddFmn2dKuqALKb=t1fA@mail.gmail.com>
	<CAE=3qObVbDQi6g1m80WLtG5MhyosV2YYnqmOoausiz-yC+NmQQ@mail.gmail.com>
	<CAKzWOxVC51tge2o1xEAW2NTvA2ewUD-XnanLRgRjsymtrRT-Og@mail.gmail.com>
	<CAE=3qObd_GZohA=zbWgGPx5bU1GLNsGkjnZPNeTf8NAxjZnVOw@mail.gmail.com>
	<CAKzWOxWTgw0c=ucZcYEHhmD2TewTmTsSFwjArB0LyA-t4gO5ag@mail.gmail.com>
Message-ID: <CAE=3qOYiqnW4rGSjPkVCOjzwOADy2eX9C_Vuiz2sN-LdaAU-jg@mail.gmail.com>

Looks like I am sending request on 5080 instead of 5060. I thought the
following would take care of that.

<action application="bridge" data="sofia/external/01869$1 at 80.40.150.150
:5060"/>

Should I set the port number elsewhere?


Please find response from provider  below.

------------------------------------------------------------

Looks like there a really small typo wrong in your configuration, the port
number appears to be 5080 instead of 5060. I have attached the packet
capture above in PCAP format showing both calls entering our Soft Switch. We
reply with a 404 not found message if the wrong port or wrong ip address is
used as we authenticate on IP Address and Port number.


On 3 August 2011 16:15, Michael Collins <msc at freeswitch.org> wrote:

> Yes, the error is "UNALLOCATED NUMBER" which means a phone number that is
> not in service or possibly a wrong number format. This means that you are
> actually talking to your provider now which is a good thing. Next what you
> need to do is ask the provider to analyze what you're sending to see if it
> is in the correct format.
>
> -MC
>
>
> On Wed, Aug 3, 2011 at 12:41 AM, lloyd thomas <lloydie.t at gmail.com> wrote:
>
>> Getting a little further, but calls are still failing.
>> Not sure why something to do with 404 error. could it be wrong number
>> format?
>>
>> changed dialplan to <action application="bridge"
>> data="sofia/external/01869$1 at 80.40.150.150:5060"/> just in case the port
>> number was incorrect.
>>
>>
>>
>> On 2 August 2011 19:16, Michael Collins <msc at freeswitch.org> wrote:
>>
>>> Lloyd,
>>>
>>> The gateway you created I think has a few issues. Let's have you bypass
>>> it altogether with a single change to your dialplan. Change your bridge line
>>> to this:
>>>
>>> <action application="bridge" data="sofia/external/01869$1 at 80.40.150.150
>>> "/>
>>>
>>> This bypasses the gateway and sends the call straight out the external
>>> profile. (You could also send it out the internal profile.)
>>>
>>> Try that and see what happens. If you have issues then do the usual
>>> console log and siptrace and put it into pastebin.freeswitch.org. Be
>>> sure to choose "FreeSWITCH Log" for the syntax highlighting.
>>>
>>> -MC
>>>
>>>
>>> On Tue, Aug 2, 2011 at 3:41 AM, lloyd thomas <lloydie.t at gmail.com>wrote:
>>>
>>>> Just did a test, but no joy. I suspect I may have to dispense with the
>>>> gateway settings and just bridge straight from the dial plan, but it is just
>>>> a guess.
>>>>
>>>>
>>>> dialplan
>>>> ----------------------------------------
>>>> <extension name="Local Dial">
>>>>    <condition field="destination_number"
>>>> expression="^9([2-9][0-9]{5})$">
>>>>      <action application="bridge"
>>>> data="sofia/gateway/phisys-2circles/01869$1"/>
>>>>    </condition>
>>>> </extension>
>>>>
>>>> gateway
>>>> ---------------------------------------
>>>> <include>
>>>>   <gateway name="phisys-2circles">
>>>>     <param name="from-domain" value="80.40.150.150"/>
>>>>     <param name="proxy" value="80.40.150.150"/>
>>>>     <param name="sip-port" value="5060"/>
>>>>   </gateway>
>>>> </include>
>>>>
>>>>
>>>>
>>>> errors
>>>> ------------------------------------
>>>> 2011-08-02 11:32:42.056302 [DEBUG] mod_dptools.c:1059 sofia/internal/
>>>> 200 at phisys.tele.phi.co.uk SET [RFC2822_DATE]=[Tue, 02 Aug 2011 11:32:42
>>>> +0100]
>>>> EXECUTE sofia/internal/200 at phisys.tele.phi.co.ukbridge(sofia/gateway/phisys-2circles/01869321110)
>>>> 2011-08-02 11:32:42.073667 [ERR] mod_sofia.c:3940 Invalid Gateway
>>>> 2011-08-02 11:32:42.073667 [NOTICE] mod_sofia.c:4282 Close Channel N/A
>>>> [CS_NEW]
>>>> 2011-08-02 11:32:42.076523 [DEBUG] switch_core_state_machine.c:452 ()
>>>> Running State Change CS_DESTROY
>>>> 2011-08-02 11:32:42.079574 [DEBUG] switch_core_state_machine.c:462 (N/A)
>>>> State DESTROY
>>>> 2011-08-02 11:32:42.079574 [DEBUG] mod_sofia.c:362 N/A SOFIA DESTROY
>>>> 2011-08-02 11:32:42.081946 [DEBUG] switch_core_state_machine.c:462 (N/A)
>>>> State DESTROY going to sleep
>>>> 2011-08-02 11:32:42.084166 [ERR] switch_ivr_originate.c:2640 Cannot
>>>> create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]
>>>> 2011-08-02 11:32:42.085649 [DEBUG] switch_ivr_originate.c:3506 Originate
>>>> Resulted in Error Cause: 28 [INVALID_NUMBER_FORMAT]
>>>> 2011-08-02 11:32:42.087275 [INFO] mod_dptools.c:2623 Originate Failed.
>>>> Cause: INVALID_NUMBER_FORMAT
>>>> 2011-08-02 11:32:42.088652 [DEBUG] switch_channel.c:2559
>>>> (sofia/internal/200 at phisys.tele.phi.co.uk) Callstate Change RINGING ->
>>>> HANGUP
>>>> 2011-08-02 11:32:42.093248 [NOTICE] mod_dptools.c:2686 Hangup
>>>> sofia/internal/200 at phisys.tele.phi.co.uk [CS_EXECUTE]
>>>> [INVALID_NUMBER_FORMAT]
>>>> 2011-08-02 11:32:42.096925 [DEBUG] switch_channel.c:2575 Send signal
>>>> sofia/internal/200 at phisys.tele.phi.co.uk [KILL]
>>>>
>>>>
>>>> On 1 August 2011 20:50, Michael Collins <msc at freeswitch.org> wrote:
>>>>
>>>>> Okay, so what happens when you dial out? Sorry, it's been a few days
>>>>> and I don't recall where we left off. Be sure to include console log w/
>>>>> siptrace on pastebin.freeswitch.org.
>>>>>
>>>>> -MC
>>>>>
>>>>>
>>>>> On Mon, Aug 1, 2011 at 12:35 PM, lloyd thomas <lloydie.t at gmail.com>wrote:
>>>>>
>>>>>> I think they have my IP on a white list.
>>>>>>
>>>>>>
>>>>>> On 1 August 2011 16:24, Michael Collins <msc at freeswitch.org> wrote:
>>>>>>
>>>>>>> Do they challenge you (digest auth) or do they have your IP address
>>>>>>> on a white list? That's a critical piece of information that only your
>>>>>>> provider can supply.
>>>>>>>
>>>>>>> -MC
>>>>>>>
>>>>>>>
>>>>>>> On Fri, Jul 29, 2011 at 9:31 PM, lloyd thomas <lloydie.t at gmail.com>wrote:
>>>>>>>
>>>>>>>> OK Inbound working with:
>>>>>>>>
>>>>>>>>   <!--2circles -->
>>>>>>>>     <list name="domains" default="allow">
>>>>>>>>       <node type="allow" cidr="80.40.150.150/32"/>
>>>>>>>>     </list>
>>>>>>>>
>>>>>>>> Just need to sort outbound.
>>>>>>>>
>>>>>>>>
>>>>>>>> On 30 July 2011 04:59, lloyd thomas <lloydie.t at gmail.com> wrote:
>>>>>>>>
>>>>>>>>> Hi, dialling in produces the following error.
>>>>>>>>>
>>>>>>>>> 2011-07-30 04:56:07.818936 [DEBUG] sofia.c:6517 IP 80.40.150.150
>>>>>>>>> Rejected by acl "domains". Falling back to Digest auth.
>>>>>>>>> 2011-07-30 04:56:07.826367 [WARNING] sofia_reg.c:1246 SIP auth
>>>>>>>>> challenge (INVITE) on sofia profile 'internal' for
>>>>>>>>> [01869******@172.16.XXX.XXX] from ip 80.40.150.150
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On 30 July 2011 04:34, lloyd thomas <lloydie.t at gmail.com> wrote:
>>>>>>>>>
>>>>>>>>>> I am registering with a them. I could not find suitable example in
>>>>>>>>>> http://wiki.freeswitch.org/wiki/SIP_Provider_Examples which
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> On 29 July 2011 21:57, Michael Collins <msc at freeswitch.org>wrote:
>>>>>>>>>>
>>>>>>>>>>> Are you registering with the provider or are they registering
>>>>>>>>>>> with you? If they register with you then a user example is appropriate. If
>>>>>>>>>>> you are registering with them then all you need is a gateway configured.
>>>>>>>>>>> -MC
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> On Fri, Jul 29, 2011 at 1:40 PM, lloyd thomas <
>>>>>>>>>>> lloydie.t at gmail.com> wrote:
>>>>>>>>>>>
>>>>>>>>>>>> Sorry, example is not clear to me.
>>>>>>>>>>>> I don't understand why a user config is relevant to sip
>>>>>>>>>>>> registration for a provider.
>>>>>>>>>>>> An example will help me more. Maybe CIDR attribute in a
>>>>>>>>>>>> sip_profile gateway could help.
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> On 29 July 2011 19:55, Steven Ayre <steveayre at gmail.com> wrote:
>>>>>>>>>>>>
>>>>>>>>>>>>> Look at the cidr attribute in the user directory to
>>>>>>>>>>>>> authenticate by IP:
>>>>>>>>>>>>> http://wiki.freeswitch.org/wiki/Acl#Users
>>>>>>>>>>>>>
>>>>>>>>>>>>> -Steve
>>>>>>>>>>>>>
>>>>>>>>>>>>> On 29 July 2011 19:38, lloyd thomas <lloydie.t at gmail.com>wrote:
>>>>>>>>>>>>>
>>>>>>>>>>>>>> *Hi I need a little help setting up a SIP registration for a
>>>>>>>>>>>>>> provider that does not use auth.*
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> *All I have is info below.*
>>>>>>>>>>>>>> **
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> *
>>>>>>>>>>>>>> *
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> SBC/Proxy IP:        80.40.150.150:5060
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Authentication:      Trusted IP ? 88.221.85.33
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Assigned DDI:        01869******, 01869******
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> DTMF Method:         RFC2833
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Status:              Live
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> No. of trunks:       2x
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Session Timer:       1800
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Profile*:*             Generic (35060)
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> Apparently the following is used for *
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> [vibe]
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> type = friend
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> host = 80.40.150.150
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>>>>
>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>>>
>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>>
>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110803/c4d4cb0d/attachment-0001.html 

From msc at freeswitch.org  Wed Aug  3 20:06:09 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 3 Aug 2011 09:06:09 -0700
Subject: [Freeswitch-users] Hangup hooks on B legs
In-Reply-To: <CABj_V-mOmm05ycYJZRPaeePeVzB4LWdTfzjNXwM3O=Kjmrn5tQ@mail.gmail.com>
References: <CABj_V-mOmm05ycYJZRPaeePeVzB4LWdTfzjNXwM3O=Kjmrn5tQ@mail.gmail.com>
Message-ID: <CAKzWOxV=q6y_Z74R0_FHxg3Lij5roSbkwPTT-WesJE+evtnYeg@mail.gmail.com>

What version of FreeSWITCH are you running?
-MC

On Wed, Aug 3, 2011 at 8:49 AM, Isaac Jurado <ijurado at econcept.es> wrote:

> Hi,
>
> I've been playing with the "api_hangup_hook" dialplan setting.  However,
> it only seems to be executing when the A-leg (originator) hangs up.  Is
> there any way to execute such a hook from all the potential B-legs of
> the call?
>
> Thanks in advance.
>
> --
> Isaac Jurado
> Internet Busines Solutions eConcept
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110803/68280eae/attachment.html 

From msc at freeswitch.org  Wed Aug  3 20:06:53 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 3 Aug 2011 09:06:53 -0700
Subject: [Freeswitch-users] Help setting up SIP reg
In-Reply-To: <CAE=3qOYiqnW4rGSjPkVCOjzwOADy2eX9C_Vuiz2sN-LdaAU-jg@mail.gmail.com>
References: <CAE=3qOYPpTuRTvq7RTJeUgzsCtvrfCjMiKFGu1fg0PHyZRE+-g@mail.gmail.com>
	<CAFiqYu=c8mY5BMWPumy1CgZHZ81kMRPgoNiF-6mjFAkW1q30fA@mail.gmail.com>
	<CAE=3qOZxwERT3CQEFHetO_o2s=aQUwYZ54TmMRPx9eYD=A7NoQ@mail.gmail.com>
	<CAKzWOxUEsGUKojF-vvstL6FvcDMhePqYGdXEQXQ4x-EJ81Bk6w@mail.gmail.com>
	<CAE=3qOacZh50iepxJKJD5UvtesxZAsN7RnPKNe1zdLO9VmFsuA@mail.gmail.com>
	<CAE=3qOZsNuiimFBZJTf4SbPRA8bRq_hJfPUunEnT8rxZj3qTnA@mail.gmail.com>
	<CAE=3qObVuLOhp2GD29Da-GfzB_qkbkyDP4kPmRxjdbJi7XeirQ@mail.gmail.com>
	<CAKzWOxUyZoBS5Evp4b61vP8fm64Lef0n01ti3KmVraQO_BqYtA@mail.gmail.com>
	<CAE=3qObsDBDTsKY2ycK_NierbS_A+-f896yMW4OArTkgauYgRA@mail.gmail.com>
	<CAKzWOxXYi-yO9DNDw1rHCduKq=h9q=BddFmn2dKuqALKb=t1fA@mail.gmail.com>
	<CAE=3qObVbDQi6g1m80WLtG5MhyosV2YYnqmOoausiz-yC+NmQQ@mail.gmail.com>
	<CAKzWOxVC51tge2o1xEAW2NTvA2ewUD-XnanLRgRjsymtrRT-Og@mail.gmail.com>
	<CAE=3qObd_GZohA=zbWgGPx5bU1GLNsGkjnZPNeTf8NAxjZnVOw@mail.gmail.com>
	<CAKzWOxWTgw0c=ucZcYEHhmD2TewTmTsSFwjArB0LyA-t4gO5ag@mail.gmail.com>
	<CAE=3qOYiqnW4rGSjPkVCOjzwOADy2eX9C_Vuiz2sN-LdaAU-jg@mail.gmail.com>
Message-ID: <CAKzWOxV-QBEuGn1uH7uWLwB=NQXGZWzxC+KbwzYqwE+FN_UBxg@mail.gmail.com>

Well, if they want to see it come from 5060 just use "internal" instead of
"external" in your bridge data and you'll be fine.
-MC

On Wed, Aug 3, 2011 at 8:58 AM, lloyd thomas <lloydie.t at gmail.com> wrote:

> Looks like I am sending request on 5080 instead of 5060. I thought the
> following would take care of that.
>
> <action application="bridge" data="sofia/external/01869$1 at 80.40.150.150
> :5060"/>
>
> Should I set the port number elsewhere?
>
>
> Please find response from provider  below.
>
> ------------------------------------------------------------
>
> Looks like there a really small typo wrong in your configuration, the port
> number appears to be 5080 instead of 5060. I have attached the packet
> capture above in PCAP format showing both calls entering our Soft Switch. We
> reply with a 404 not found message if the wrong port or wrong ip address is
> used as we authenticate on IP Address and Port number.
>
>
> On 3 August 2011 16:15, Michael Collins <msc at freeswitch.org> wrote:
>
>> Yes, the error is "UNALLOCATED NUMBER" which means a phone number that is
>> not in service or possibly a wrong number format. This means that you are
>> actually talking to your provider now which is a good thing. Next what you
>> need to do is ask the provider to analyze what you're sending to see if it
>> is in the correct format.
>>
>> -MC
>>
>>
>> On Wed, Aug 3, 2011 at 12:41 AM, lloyd thomas <lloydie.t at gmail.com>wrote:
>>
>>> Getting a little further, but calls are still failing.
>>> Not sure why something to do with 404 error. could it be wrong number
>>> format?
>>>
>>> changed dialplan to <action application="bridge"
>>> data="sofia/external/01869$1 at 80.40.150.150:5060"/> just in case the port
>>> number was incorrect.
>>>
>>>
>>>
>>> On 2 August 2011 19:16, Michael Collins <msc at freeswitch.org> wrote:
>>>
>>>> Lloyd,
>>>>
>>>> The gateway you created I think has a few issues. Let's have you bypass
>>>> it altogether with a single change to your dialplan. Change your bridge line
>>>> to this:
>>>>
>>>> <action application="bridge" data="sofia/external/01869$1 at 80.40.150.150
>>>> "/>
>>>>
>>>> This bypasses the gateway and sends the call straight out the external
>>>> profile. (You could also send it out the internal profile.)
>>>>
>>>> Try that and see what happens. If you have issues then do the usual
>>>> console log and siptrace and put it into pastebin.freeswitch.org. Be
>>>> sure to choose "FreeSWITCH Log" for the syntax highlighting.
>>>>
>>>> -MC
>>>>
>>>>
>>>> On Tue, Aug 2, 2011 at 3:41 AM, lloyd thomas <lloydie.t at gmail.com>wrote:
>>>>
>>>>> Just did a test, but no joy. I suspect I may have to dispense with the
>>>>> gateway settings and just bridge straight from the dial plan, but it is just
>>>>> a guess.
>>>>>
>>>>>
>>>>> dialplan
>>>>> ----------------------------------------
>>>>> <extension name="Local Dial">
>>>>>    <condition field="destination_number"
>>>>> expression="^9([2-9][0-9]{5})$">
>>>>>      <action application="bridge"
>>>>> data="sofia/gateway/phisys-2circles/01869$1"/>
>>>>>    </condition>
>>>>> </extension>
>>>>>
>>>>> gateway
>>>>> ---------------------------------------
>>>>> <include>
>>>>>   <gateway name="phisys-2circles">
>>>>>     <param name="from-domain" value="80.40.150.150"/>
>>>>>     <param name="proxy" value="80.40.150.150"/>
>>>>>     <param name="sip-port" value="5060"/>
>>>>>   </gateway>
>>>>> </include>
>>>>>
>>>>>
>>>>>
>>>>> errors
>>>>> ------------------------------------
>>>>> 2011-08-02 11:32:42.056302 [DEBUG] mod_dptools.c:1059 sofia/internal/
>>>>> 200 at phisys.tele.phi.co.uk SET [RFC2822_DATE]=[Tue, 02 Aug 2011
>>>>> 11:32:42 +0100]
>>>>> EXECUTE sofia/internal/200 at phisys.tele.phi.co.ukbridge(sofia/gateway/phisys-2circles/01869321110)
>>>>> 2011-08-02 11:32:42.073667 [ERR] mod_sofia.c:3940 Invalid Gateway
>>>>> 2011-08-02 11:32:42.073667 [NOTICE] mod_sofia.c:4282 Close Channel N/A
>>>>> [CS_NEW]
>>>>> 2011-08-02 11:32:42.076523 [DEBUG] switch_core_state_machine.c:452 ()
>>>>> Running State Change CS_DESTROY
>>>>> 2011-08-02 11:32:42.079574 [DEBUG] switch_core_state_machine.c:462
>>>>> (N/A) State DESTROY
>>>>> 2011-08-02 11:32:42.079574 [DEBUG] mod_sofia.c:362 N/A SOFIA DESTROY
>>>>> 2011-08-02 11:32:42.081946 [DEBUG] switch_core_state_machine.c:462
>>>>> (N/A) State DESTROY going to sleep
>>>>> 2011-08-02 11:32:42.084166 [ERR] switch_ivr_originate.c:2640 Cannot
>>>>> create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]
>>>>> 2011-08-02 11:32:42.085649 [DEBUG] switch_ivr_originate.c:3506
>>>>> Originate Resulted in Error Cause: 28 [INVALID_NUMBER_FORMAT]
>>>>> 2011-08-02 11:32:42.087275 [INFO] mod_dptools.c:2623 Originate Failed.
>>>>> Cause: INVALID_NUMBER_FORMAT
>>>>> 2011-08-02 11:32:42.088652 [DEBUG] switch_channel.c:2559
>>>>> (sofia/internal/200 at phisys.tele.phi.co.uk) Callstate Change RINGING ->
>>>>> HANGUP
>>>>> 2011-08-02 11:32:42.093248 [NOTICE] mod_dptools.c:2686 Hangup
>>>>> sofia/internal/200 at phisys.tele.phi.co.uk [CS_EXECUTE]
>>>>> [INVALID_NUMBER_FORMAT]
>>>>> 2011-08-02 11:32:42.096925 [DEBUG] switch_channel.c:2575 Send signal
>>>>> sofia/internal/200 at phisys.tele.phi.co.uk [KILL]
>>>>>
>>>>>
>>>>> On 1 August 2011 20:50, Michael Collins <msc at freeswitch.org> wrote:
>>>>>
>>>>>> Okay, so what happens when you dial out? Sorry, it's been a few days
>>>>>> and I don't recall where we left off. Be sure to include console log w/
>>>>>> siptrace on pastebin.freeswitch.org.
>>>>>>
>>>>>> -MC
>>>>>>
>>>>>>
>>>>>> On Mon, Aug 1, 2011 at 12:35 PM, lloyd thomas <lloydie.t at gmail.com>wrote:
>>>>>>
>>>>>>> I think they have my IP on a white list.
>>>>>>>
>>>>>>>
>>>>>>> On 1 August 2011 16:24, Michael Collins <msc at freeswitch.org> wrote:
>>>>>>>
>>>>>>>> Do they challenge you (digest auth) or do they have your IP address
>>>>>>>> on a white list? That's a critical piece of information that only your
>>>>>>>> provider can supply.
>>>>>>>>
>>>>>>>> -MC
>>>>>>>>
>>>>>>>>
>>>>>>>> On Fri, Jul 29, 2011 at 9:31 PM, lloyd thomas <lloydie.t at gmail.com>wrote:
>>>>>>>>
>>>>>>>>> OK Inbound working with:
>>>>>>>>>
>>>>>>>>>   <!--2circles -->
>>>>>>>>>     <list name="domains" default="allow">
>>>>>>>>>       <node type="allow" cidr="80.40.150.150/32"/>
>>>>>>>>>     </list>
>>>>>>>>>
>>>>>>>>> Just need to sort outbound.
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On 30 July 2011 04:59, lloyd thomas <lloydie.t at gmail.com> wrote:
>>>>>>>>>
>>>>>>>>>> Hi, dialling in produces the following error.
>>>>>>>>>>
>>>>>>>>>> 2011-07-30 04:56:07.818936 [DEBUG] sofia.c:6517 IP 80.40.150.150
>>>>>>>>>> Rejected by acl "domains". Falling back to Digest auth.
>>>>>>>>>> 2011-07-30 04:56:07.826367 [WARNING] sofia_reg.c:1246 SIP auth
>>>>>>>>>> challenge (INVITE) on sofia profile 'internal' for
>>>>>>>>>> [01869******@172.16.XXX.XXX] from ip 80.40.150.150
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> On 30 July 2011 04:34, lloyd thomas <lloydie.t at gmail.com> wrote:
>>>>>>>>>>
>>>>>>>>>>> I am registering with a them. I could not find suitable example
>>>>>>>>>>> in http://wiki.freeswitch.org/wiki/SIP_Provider_Examples which
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> On 29 July 2011 21:57, Michael Collins <msc at freeswitch.org>wrote:
>>>>>>>>>>>
>>>>>>>>>>>> Are you registering with the provider or are they registering
>>>>>>>>>>>> with you? If they register with you then a user example is appropriate. If
>>>>>>>>>>>> you are registering with them then all you need is a gateway configured.
>>>>>>>>>>>> -MC
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> On Fri, Jul 29, 2011 at 1:40 PM, lloyd thomas <
>>>>>>>>>>>> lloydie.t at gmail.com> wrote:
>>>>>>>>>>>>
>>>>>>>>>>>>> Sorry, example is not clear to me.
>>>>>>>>>>>>> I don't understand why a user config is relevant to sip
>>>>>>>>>>>>> registration for a provider.
>>>>>>>>>>>>> An example will help me more. Maybe CIDR attribute in a
>>>>>>>>>>>>> sip_profile gateway could help.
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> On 29 July 2011 19:55, Steven Ayre <steveayre at gmail.com>wrote:
>>>>>>>>>>>>>
>>>>>>>>>>>>>> Look at the cidr attribute in the user directory to
>>>>>>>>>>>>>> authenticate by IP:
>>>>>>>>>>>>>> http://wiki.freeswitch.org/wiki/Acl#Users
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> -Steve
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> On 29 July 2011 19:38, lloyd thomas <lloydie.t at gmail.com>wrote:
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> *Hi I need a little help setting up a SIP registration for a
>>>>>>>>>>>>>>> provider that does not use auth.*
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> *All I have is info below.*
>>>>>>>>>>>>>>> **
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> *
>>>>>>>>>>>>>>> *
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> SBC/Proxy IP:        80.40.150.150:5060
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Authentication:      Trusted IP ? 88.221.85.33
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Assigned DDI:        01869******, 01869******
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> DTMF Method:         RFC2833
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Status:              Live
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> No. of trunks:       2x
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Session Timer:       1800
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Profile*:*             Generic (35060)
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Apparently the following is used for *
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> [vibe]
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> type = friend
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> host = 80.40.150.150
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>>>>
>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>>>
>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110803/f47d6c0e/attachment-0001.html 

From avi at avimarcus.net  Wed Aug  3 20:09:58 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 3 Aug 2011 19:09:58 +0300
Subject: [Freeswitch-users] Hangup hooks on B legs
In-Reply-To: <CABj_V-mOmm05ycYJZRPaeePeVzB4LWdTfzjNXwM3O=Kjmrn5tQ@mail.gmail.com>
References: <CABj_V-mOmm05ycYJZRPaeePeVzB4LWdTfzjNXwM3O=Kjmrn5tQ@mail.gmail.com>
Message-ID: <CALNZuwm7Zh1FZpeavRECM6VjPUERsJODi=gRxWsXboR4hKEiLg@mail.gmail.com>

Yes, you can export it, "nolocal" to activate only on the B leg, e.g.:

<action application="export" data="nolocal:api_hangup_hook=lua file.lua"/>

Or.. set the variable in the bridge string.

-Avi


On Wed, Aug 3, 2011 at 6:49 PM, Isaac Jurado <ijurado at econcept.es> wrote:
>
> Hi,
>
> I've been playing with the "api_hangup_hook" dialplan setting. ?However,
> it only seems to be executing when the A-leg (originator) hangs up. ?Is
> there any way to execute such a hook from all the potential B-legs of
> the call?
>
> Thanks in advance.
>
> --
> Isaac Jurado
> Internet Busines Solutions eConcept
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From msc at freeswitch.org  Wed Aug  3 20:10:50 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 3 Aug 2011 09:10:50 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today
Message-ID: <CAKzWOxXkGY+5OOvpAYoQ--K-CDCmSLiPNaeJ1uPgcixftdQu0Q@mail.gmail.com>

Hello all! Today's agenda is:

http://wiki.freeswitch.org/wiki/FS_weekly_2011_08_03

We have a few things to discuss plus a sneak peek at a new SIP logging
project that will be presented at ClueCon. (We'll have the author on a
future conf call to talk to us.)

-Michael
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110803/657cbc40/attachment.html 

From lloydie.t at gmail.com  Wed Aug  3 20:27:12 2011
From: lloydie.t at gmail.com (lloyd thomas)
Date: Wed, 3 Aug 2011 17:27:12 +0100
Subject: [Freeswitch-users] Help setting up SIP reg
In-Reply-To: <CAKzWOxV-QBEuGn1uH7uWLwB=NQXGZWzxC+KbwzYqwE+FN_UBxg@mail.gmail.com>
References: <CAE=3qOYPpTuRTvq7RTJeUgzsCtvrfCjMiKFGu1fg0PHyZRE+-g@mail.gmail.com>
	<CAFiqYu=c8mY5BMWPumy1CgZHZ81kMRPgoNiF-6mjFAkW1q30fA@mail.gmail.com>
	<CAE=3qOZxwERT3CQEFHetO_o2s=aQUwYZ54TmMRPx9eYD=A7NoQ@mail.gmail.com>
	<CAKzWOxUEsGUKojF-vvstL6FvcDMhePqYGdXEQXQ4x-EJ81Bk6w@mail.gmail.com>
	<CAE=3qOacZh50iepxJKJD5UvtesxZAsN7RnPKNe1zdLO9VmFsuA@mail.gmail.com>
	<CAE=3qOZsNuiimFBZJTf4SbPRA8bRq_hJfPUunEnT8rxZj3qTnA@mail.gmail.com>
	<CAE=3qObVuLOhp2GD29Da-GfzB_qkbkyDP4kPmRxjdbJi7XeirQ@mail.gmail.com>
	<CAKzWOxUyZoBS5Evp4b61vP8fm64Lef0n01ti3KmVraQO_BqYtA@mail.gmail.com>
	<CAE=3qObsDBDTsKY2ycK_NierbS_A+-f896yMW4OArTkgauYgRA@mail.gmail.com>
	<CAKzWOxXYi-yO9DNDw1rHCduKq=h9q=BddFmn2dKuqALKb=t1fA@mail.gmail.com>
	<CAE=3qObVbDQi6g1m80WLtG5MhyosV2YYnqmOoausiz-yC+NmQQ@mail.gmail.com>
	<CAKzWOxVC51tge2o1xEAW2NTvA2ewUD-XnanLRgRjsymtrRT-Og@mail.gmail.com>
	<CAE=3qObd_GZohA=zbWgGPx5bU1GLNsGkjnZPNeTf8NAxjZnVOw@mail.gmail.com>
	<CAKzWOxWTgw0c=ucZcYEHhmD2TewTmTsSFwjArB0LyA-t4gO5ag@mail.gmail.com>
	<CAE=3qOYiqnW4rGSjPkVCOjzwOADy2eX9C_Vuiz2sN-LdaAU-jg@mail.gmail.com>
	<CAKzWOxV-QBEuGn1uH7uWLwB=NQXGZWzxC+KbwzYqwE+FN_UBxg@mail.gmail.com>
Message-ID: <CAE=3qObGnL=hpCz0+G6CRdf9xdaKw2cKvnny23Ffn+hhW2a1Bg@mail.gmail.com>

Thanks Michael,
                        That worked fine, but I would like to know how to
structure a new xml file should I require to use a different port number
than those. Should I use internal.xml as template?

Lloyd T

On 3 August 2011 17:06, Michael Collins <msc at freeswitch.org> wrote:

> Well, if they want to see it come from 5060 just use "internal" instead of
> "external" in your bridge data and you'll be fine.
> -MC
>
>
> On Wed, Aug 3, 2011 at 8:58 AM, lloyd thomas <lloydie.t at gmail.com> wrote:
>
>> Looks like I am sending request on 5080 instead of 5060. I thought the
>> following would take care of that.
>>
>> <action application="bridge" data="sofia/external/01869$1 at 80.40.150.150
>> :5060"/>
>>
>> Should I set the port number elsewhere?
>>
>>
>> Please find response from provider  below.
>>
>> ------------------------------------------------------------
>>
>> Looks like there a really small typo wrong in your configuration, the port
>> number appears to be 5080 instead of 5060. I have attached the packet
>> capture above in PCAP format showing both calls entering our Soft Switch. We
>> reply with a 404 not found message if the wrong port or wrong ip address is
>> used as we authenticate on IP Address and Port number.
>>
>>
>> On 3 August 2011 16:15, Michael Collins <msc at freeswitch.org> wrote:
>>
>>> Yes, the error is "UNALLOCATED NUMBER" which means a phone number that is
>>> not in service or possibly a wrong number format. This means that you are
>>> actually talking to your provider now which is a good thing. Next what you
>>> need to do is ask the provider to analyze what you're sending to see if it
>>> is in the correct format.
>>>
>>> -MC
>>>
>>>
>>> On Wed, Aug 3, 2011 at 12:41 AM, lloyd thomas <lloydie.t at gmail.com>wrote:
>>>
>>>> Getting a little further, but calls are still failing.
>>>> Not sure why something to do with 404 error. could it be wrong number
>>>> format?
>>>>
>>>> changed dialplan to <action application="bridge"
>>>> data="sofia/external/01869$1 at 80.40.150.150:5060"/> just in case the
>>>> port number was incorrect.
>>>>
>>>>
>>>>
>>>> On 2 August 2011 19:16, Michael Collins <msc at freeswitch.org> wrote:
>>>>
>>>>> Lloyd,
>>>>>
>>>>> The gateway you created I think has a few issues. Let's have you bypass
>>>>> it altogether with a single change to your dialplan. Change your bridge line
>>>>> to this:
>>>>>
>>>>> <action application="bridge" data="sofia/external/01869$
>>>>> 1 at 80.40.150.150"/>
>>>>>
>>>>> This bypasses the gateway and sends the call straight out the external
>>>>> profile. (You could also send it out the internal profile.)
>>>>>
>>>>> Try that and see what happens. If you have issues then do the usual
>>>>> console log and siptrace and put it into pastebin.freeswitch.org. Be
>>>>> sure to choose "FreeSWITCH Log" for the syntax highlighting.
>>>>>
>>>>> -MC
>>>>>
>>>>>
>>>>> On Tue, Aug 2, 2011 at 3:41 AM, lloyd thomas <lloydie.t at gmail.com>wrote:
>>>>>
>>>>>> Just did a test, but no joy. I suspect I may have to dispense with the
>>>>>> gateway settings and just bridge straight from the dial plan, but it is just
>>>>>> a guess.
>>>>>>
>>>>>>
>>>>>> dialplan
>>>>>> ----------------------------------------
>>>>>> <extension name="Local Dial">
>>>>>>    <condition field="destination_number"
>>>>>> expression="^9([2-9][0-9]{5})$">
>>>>>>      <action application="bridge"
>>>>>> data="sofia/gateway/phisys-2circles/01869$1"/>
>>>>>>    </condition>
>>>>>> </extension>
>>>>>>
>>>>>> gateway
>>>>>> ---------------------------------------
>>>>>> <include>
>>>>>>   <gateway name="phisys-2circles">
>>>>>>     <param name="from-domain" value="80.40.150.150"/>
>>>>>>     <param name="proxy" value="80.40.150.150"/>
>>>>>>     <param name="sip-port" value="5060"/>
>>>>>>   </gateway>
>>>>>> </include>
>>>>>>
>>>>>>
>>>>>>
>>>>>> errors
>>>>>> ------------------------------------
>>>>>> 2011-08-02 11:32:42.056302 [DEBUG] mod_dptools.c:1059 sofia/internal/
>>>>>> 200 at phisys.tele.phi.co.uk SET [RFC2822_DATE]=[Tue, 02 Aug 2011
>>>>>> 11:32:42 +0100]
>>>>>> EXECUTE sofia/internal/200 at phisys.tele.phi.co.ukbridge(sofia/gateway/phisys-2circles/01869321110)
>>>>>> 2011-08-02 11:32:42.073667 [ERR] mod_sofia.c:3940 Invalid Gateway
>>>>>> 2011-08-02 11:32:42.073667 [NOTICE] mod_sofia.c:4282 Close Channel N/A
>>>>>> [CS_NEW]
>>>>>> 2011-08-02 11:32:42.076523 [DEBUG] switch_core_state_machine.c:452 ()
>>>>>> Running State Change CS_DESTROY
>>>>>> 2011-08-02 11:32:42.079574 [DEBUG] switch_core_state_machine.c:462
>>>>>> (N/A) State DESTROY
>>>>>> 2011-08-02 11:32:42.079574 [DEBUG] mod_sofia.c:362 N/A SOFIA DESTROY
>>>>>> 2011-08-02 11:32:42.081946 [DEBUG] switch_core_state_machine.c:462
>>>>>> (N/A) State DESTROY going to sleep
>>>>>> 2011-08-02 11:32:42.084166 [ERR] switch_ivr_originate.c:2640 Cannot
>>>>>> create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]
>>>>>> 2011-08-02 11:32:42.085649 [DEBUG] switch_ivr_originate.c:3506
>>>>>> Originate Resulted in Error Cause: 28 [INVALID_NUMBER_FORMAT]
>>>>>> 2011-08-02 11:32:42.087275 [INFO] mod_dptools.c:2623 Originate
>>>>>> Failed.  Cause: INVALID_NUMBER_FORMAT
>>>>>> 2011-08-02 11:32:42.088652 [DEBUG] switch_channel.c:2559
>>>>>> (sofia/internal/200 at phisys.tele.phi.co.uk) Callstate Change RINGING
>>>>>> -> HANGUP
>>>>>> 2011-08-02 11:32:42.093248 [NOTICE] mod_dptools.c:2686 Hangup
>>>>>> sofia/internal/200 at phisys.tele.phi.co.uk [CS_EXECUTE]
>>>>>> [INVALID_NUMBER_FORMAT]
>>>>>> 2011-08-02 11:32:42.096925 [DEBUG] switch_channel.c:2575 Send signal
>>>>>> sofia/internal/200 at phisys.tele.phi.co.uk [KILL]
>>>>>>
>>>>>>
>>>>>> On 1 August 2011 20:50, Michael Collins <msc at freeswitch.org> wrote:
>>>>>>
>>>>>>> Okay, so what happens when you dial out? Sorry, it's been a few days
>>>>>>> and I don't recall where we left off. Be sure to include console log w/
>>>>>>> siptrace on pastebin.freeswitch.org.
>>>>>>>
>>>>>>> -MC
>>>>>>>
>>>>>>>
>>>>>>> On Mon, Aug 1, 2011 at 12:35 PM, lloyd thomas <lloydie.t at gmail.com>wrote:
>>>>>>>
>>>>>>>> I think they have my IP on a white list.
>>>>>>>>
>>>>>>>>
>>>>>>>> On 1 August 2011 16:24, Michael Collins <msc at freeswitch.org> wrote:
>>>>>>>>
>>>>>>>>> Do they challenge you (digest auth) or do they have your IP address
>>>>>>>>> on a white list? That's a critical piece of information that only your
>>>>>>>>> provider can supply.
>>>>>>>>>
>>>>>>>>> -MC
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On Fri, Jul 29, 2011 at 9:31 PM, lloyd thomas <lloydie.t at gmail.com
>>>>>>>>> > wrote:
>>>>>>>>>
>>>>>>>>>> OK Inbound working with:
>>>>>>>>>>
>>>>>>>>>>   <!--2circles -->
>>>>>>>>>>     <list name="domains" default="allow">
>>>>>>>>>>       <node type="allow" cidr="80.40.150.150/32"/>
>>>>>>>>>>     </list>
>>>>>>>>>>
>>>>>>>>>> Just need to sort outbound.
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> On 30 July 2011 04:59, lloyd thomas <lloydie.t at gmail.com> wrote:
>>>>>>>>>>
>>>>>>>>>>> Hi, dialling in produces the following error.
>>>>>>>>>>>
>>>>>>>>>>> 2011-07-30 04:56:07.818936 [DEBUG] sofia.c:6517 IP 80.40.150.150
>>>>>>>>>>> Rejected by acl "domains". Falling back to Digest auth.
>>>>>>>>>>> 2011-07-30 04:56:07.826367 [WARNING] sofia_reg.c:1246 SIP auth
>>>>>>>>>>> challenge (INVITE) on sofia profile 'internal' for
>>>>>>>>>>> [01869******@172.16.XXX.XXX] from ip 80.40.150.150
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> On 30 July 2011 04:34, lloyd thomas <lloydie.t at gmail.com> wrote:
>>>>>>>>>>>
>>>>>>>>>>>> I am registering with a them. I could not find suitable example
>>>>>>>>>>>> in http://wiki.freeswitch.org/wiki/SIP_Provider_Examples which
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> On 29 July 2011 21:57, Michael Collins <msc at freeswitch.org>wrote:
>>>>>>>>>>>>
>>>>>>>>>>>>> Are you registering with the provider or are they registering
>>>>>>>>>>>>> with you? If they register with you then a user example is appropriate. If
>>>>>>>>>>>>> you are registering with them then all you need is a gateway configured.
>>>>>>>>>>>>> -MC
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> On Fri, Jul 29, 2011 at 1:40 PM, lloyd thomas <
>>>>>>>>>>>>> lloydie.t at gmail.com> wrote:
>>>>>>>>>>>>>
>>>>>>>>>>>>>> Sorry, example is not clear to me.
>>>>>>>>>>>>>> I don't understand why a user config is relevant to sip
>>>>>>>>>>>>>> registration for a provider.
>>>>>>>>>>>>>> An example will help me more. Maybe CIDR attribute in a
>>>>>>>>>>>>>> sip_profile gateway could help.
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> On 29 July 2011 19:55, Steven Ayre <steveayre at gmail.com>wrote:
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Look at the cidr attribute in the user directory to
>>>>>>>>>>>>>>> authenticate by IP:
>>>>>>>>>>>>>>> http://wiki.freeswitch.org/wiki/Acl#Users
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> -Steve
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> On 29 July 2011 19:38, lloyd thomas <lloydie.t at gmail.com>wrote:
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> *Hi I need a little help setting up a SIP registration for
>>>>>>>>>>>>>>>> a provider that does not use auth.*
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> *All I have is info below.*
>>>>>>>>>>>>>>>> **
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> *
>>>>>>>>>>>>>>>> *
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> SBC/Proxy IP:        80.40.150.150:5060
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Authentication:      Trusted IP ? 88.221.85.33
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Assigned DDI:        01869******, 01869******
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> DTMF Method:         RFC2833
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Status:              Live
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> No. of trunks:       2x
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Session Timer:       1800
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Profile*:*             Generic (35060)
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Apparently the following is used for *
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> [vibe]
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> type = friend
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> host = 80.40.150.150
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>>>>
>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110803/e69c92ab/attachment-0001.html 

From msc at freeswitch.org  Wed Aug  3 20:33:36 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 3 Aug 2011 09:33:36 -0700
Subject: [Freeswitch-users] Help setting up SIP reg
In-Reply-To: <CAE=3qObGnL=hpCz0+G6CRdf9xdaKw2cKvnny23Ffn+hhW2a1Bg@mail.gmail.com>
References: <CAE=3qOYPpTuRTvq7RTJeUgzsCtvrfCjMiKFGu1fg0PHyZRE+-g@mail.gmail.com>
	<CAFiqYu=c8mY5BMWPumy1CgZHZ81kMRPgoNiF-6mjFAkW1q30fA@mail.gmail.com>
	<CAE=3qOZxwERT3CQEFHetO_o2s=aQUwYZ54TmMRPx9eYD=A7NoQ@mail.gmail.com>
	<CAKzWOxUEsGUKojF-vvstL6FvcDMhePqYGdXEQXQ4x-EJ81Bk6w@mail.gmail.com>
	<CAE=3qOacZh50iepxJKJD5UvtesxZAsN7RnPKNe1zdLO9VmFsuA@mail.gmail.com>
	<CAE=3qOZsNuiimFBZJTf4SbPRA8bRq_hJfPUunEnT8rxZj3qTnA@mail.gmail.com>
	<CAE=3qObVuLOhp2GD29Da-GfzB_qkbkyDP4kPmRxjdbJi7XeirQ@mail.gmail.com>
	<CAKzWOxUyZoBS5Evp4b61vP8fm64Lef0n01ti3KmVraQO_BqYtA@mail.gmail.com>
	<CAE=3qObsDBDTsKY2ycK_NierbS_A+-f896yMW4OArTkgauYgRA@mail.gmail.com>
	<CAKzWOxXYi-yO9DNDw1rHCduKq=h9q=BddFmn2dKuqALKb=t1fA@mail.gmail.com>
	<CAE=3qObVbDQi6g1m80WLtG5MhyosV2YYnqmOoausiz-yC+NmQQ@mail.gmail.com>
	<CAKzWOxVC51tge2o1xEAW2NTvA2ewUD-XnanLRgRjsymtrRT-Og@mail.gmail.com>
	<CAE=3qObd_GZohA=zbWgGPx5bU1GLNsGkjnZPNeTf8NAxjZnVOw@mail.gmail.com>
	<CAKzWOxWTgw0c=ucZcYEHhmD2TewTmTsSFwjArB0LyA-t4gO5ag@mail.gmail.com>
	<CAE=3qOYiqnW4rGSjPkVCOjzwOADy2eX9C_Vuiz2sN-LdaAU-jg@mail.gmail.com>
	<CAKzWOxV-QBEuGn1uH7uWLwB=NQXGZWzxC+KbwzYqwE+FN_UBxg@mail.gmail.com>
	<CAE=3qObGnL=hpCz0+G6CRdf9xdaKw2cKvnny23Ffn+hhW2a1Bg@mail.gmail.com>
Message-ID: <CAKzWOxVheq+oj-2QSnqXqKtJQQmeg=FBg9CE-V9G6gRY8HVsHg@mail.gmail.com>

Use internal.xml if your Ethernet port is behind NAT; use external.xml if
not.
-MC

On Wed, Aug 3, 2011 at 9:27 AM, lloyd thomas <lloydie.t at gmail.com> wrote:

> Thanks Michael,
>                         That worked fine, but I would like to know how to
> structure a new xml file should I require to use a different port number
> than those. Should I use internal.xml as template?
>
> Lloyd T
>
>
> On 3 August 2011 17:06, Michael Collins <msc at freeswitch.org> wrote:
>
>> Well, if they want to see it come from 5060 just use "internal" instead of
>> "external" in your bridge data and you'll be fine.
>> -MC
>>
>>
>> On Wed, Aug 3, 2011 at 8:58 AM, lloyd thomas <lloydie.t at gmail.com> wrote:
>>
>>> Looks like I am sending request on 5080 instead of 5060. I thought the
>>> following would take care of that.
>>>
>>> <action application="bridge" data="sofia/external/01869$1 at 80.40.150.150
>>> :5060"/>
>>>
>>> Should I set the port number elsewhere?
>>>
>>>
>>> Please find response from provider  below.
>>>
>>> ------------------------------------------------------------
>>>
>>> Looks like there a really small typo wrong in your configuration, the
>>> port number appears to be 5080 instead of 5060. I have attached the packet
>>> capture above in PCAP format showing both calls entering our Soft Switch. We
>>> reply with a 404 not found message if the wrong port or wrong ip address is
>>> used as we authenticate on IP Address and Port number.
>>>
>>>
>>> On 3 August 2011 16:15, Michael Collins <msc at freeswitch.org> wrote:
>>>
>>>> Yes, the error is "UNALLOCATED NUMBER" which means a phone number that
>>>> is not in service or possibly a wrong number format. This means that you are
>>>> actually talking to your provider now which is a good thing. Next what you
>>>> need to do is ask the provider to analyze what you're sending to see if it
>>>> is in the correct format.
>>>>
>>>> -MC
>>>>
>>>>
>>>> On Wed, Aug 3, 2011 at 12:41 AM, lloyd thomas <lloydie.t at gmail.com>wrote:
>>>>
>>>>> Getting a little further, but calls are still failing.
>>>>> Not sure why something to do with 404 error. could it be wrong number
>>>>> format?
>>>>>
>>>>> changed dialplan to <action application="bridge"
>>>>> data="sofia/external/01869$1 at 80.40.150.150:5060"/> just in case the
>>>>> port number was incorrect.
>>>>>
>>>>>
>>>>>
>>>>> On 2 August 2011 19:16, Michael Collins <msc at freeswitch.org> wrote:
>>>>>
>>>>>> Lloyd,
>>>>>>
>>>>>> The gateway you created I think has a few issues. Let's have you
>>>>>> bypass it altogether with a single change to your dialplan. Change your
>>>>>> bridge line to this:
>>>>>>
>>>>>> <action application="bridge" data="sofia/external/01869$
>>>>>> 1 at 80.40.150.150"/>
>>>>>>
>>>>>> This bypasses the gateway and sends the call straight out the external
>>>>>> profile. (You could also send it out the internal profile.)
>>>>>>
>>>>>> Try that and see what happens. If you have issues then do the usual
>>>>>> console log and siptrace and put it into pastebin.freeswitch.org. Be
>>>>>> sure to choose "FreeSWITCH Log" for the syntax highlighting.
>>>>>>
>>>>>> -MC
>>>>>>
>>>>>>
>>>>>> On Tue, Aug 2, 2011 at 3:41 AM, lloyd thomas <lloydie.t at gmail.com>wrote:
>>>>>>
>>>>>>> Just did a test, but no joy. I suspect I may have to dispense with
>>>>>>> the gateway settings and just bridge straight from the dial plan, but it is
>>>>>>> just a guess.
>>>>>>>
>>>>>>>
>>>>>>> dialplan
>>>>>>> ----------------------------------------
>>>>>>> <extension name="Local Dial">
>>>>>>>    <condition field="destination_number"
>>>>>>> expression="^9([2-9][0-9]{5})$">
>>>>>>>      <action application="bridge"
>>>>>>> data="sofia/gateway/phisys-2circles/01869$1"/>
>>>>>>>    </condition>
>>>>>>> </extension>
>>>>>>>
>>>>>>> gateway
>>>>>>> ---------------------------------------
>>>>>>> <include>
>>>>>>>   <gateway name="phisys-2circles">
>>>>>>>     <param name="from-domain" value="80.40.150.150"/>
>>>>>>>     <param name="proxy" value="80.40.150.150"/>
>>>>>>>     <param name="sip-port" value="5060"/>
>>>>>>>   </gateway>
>>>>>>> </include>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> errors
>>>>>>> ------------------------------------
>>>>>>> 2011-08-02 11:32:42.056302 [DEBUG] mod_dptools.c:1059 sofia/internal/
>>>>>>> 200 at phisys.tele.phi.co.uk SET [RFC2822_DATE]=[Tue, 02 Aug 2011
>>>>>>> 11:32:42 +0100]
>>>>>>> EXECUTE sofia/internal/200 at phisys.tele.phi.co.ukbridge(sofia/gateway/phisys-2circles/01869321110)
>>>>>>> 2011-08-02 11:32:42.073667 [ERR] mod_sofia.c:3940 Invalid Gateway
>>>>>>> 2011-08-02 11:32:42.073667 [NOTICE] mod_sofia.c:4282 Close Channel
>>>>>>> N/A [CS_NEW]
>>>>>>> 2011-08-02 11:32:42.076523 [DEBUG] switch_core_state_machine.c:452 ()
>>>>>>> Running State Change CS_DESTROY
>>>>>>> 2011-08-02 11:32:42.079574 [DEBUG] switch_core_state_machine.c:462
>>>>>>> (N/A) State DESTROY
>>>>>>> 2011-08-02 11:32:42.079574 [DEBUG] mod_sofia.c:362 N/A SOFIA DESTROY
>>>>>>> 2011-08-02 11:32:42.081946 [DEBUG] switch_core_state_machine.c:462
>>>>>>> (N/A) State DESTROY going to sleep
>>>>>>> 2011-08-02 11:32:42.084166 [ERR] switch_ivr_originate.c:2640 Cannot
>>>>>>> create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]
>>>>>>> 2011-08-02 11:32:42.085649 [DEBUG] switch_ivr_originate.c:3506
>>>>>>> Originate Resulted in Error Cause: 28 [INVALID_NUMBER_FORMAT]
>>>>>>> 2011-08-02 11:32:42.087275 [INFO] mod_dptools.c:2623 Originate
>>>>>>> Failed.  Cause: INVALID_NUMBER_FORMAT
>>>>>>> 2011-08-02 11:32:42.088652 [DEBUG] switch_channel.c:2559
>>>>>>> (sofia/internal/200 at phisys.tele.phi.co.uk) Callstate Change RINGING
>>>>>>> -> HANGUP
>>>>>>> 2011-08-02 11:32:42.093248 [NOTICE] mod_dptools.c:2686 Hangup
>>>>>>> sofia/internal/200 at phisys.tele.phi.co.uk [CS_EXECUTE]
>>>>>>> [INVALID_NUMBER_FORMAT]
>>>>>>> 2011-08-02 11:32:42.096925 [DEBUG] switch_channel.c:2575 Send signal
>>>>>>> sofia/internal/200 at phisys.tele.phi.co.uk [KILL]
>>>>>>>
>>>>>>>
>>>>>>> On 1 August 2011 20:50, Michael Collins <msc at freeswitch.org> wrote:
>>>>>>>
>>>>>>>> Okay, so what happens when you dial out? Sorry, it's been a few days
>>>>>>>> and I don't recall where we left off. Be sure to include console log w/
>>>>>>>> siptrace on pastebin.freeswitch.org.
>>>>>>>>
>>>>>>>> -MC
>>>>>>>>
>>>>>>>>
>>>>>>>> On Mon, Aug 1, 2011 at 12:35 PM, lloyd thomas <lloydie.t at gmail.com>wrote:
>>>>>>>>
>>>>>>>>> I think they have my IP on a white list.
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On 1 August 2011 16:24, Michael Collins <msc at freeswitch.org>wrote:
>>>>>>>>>
>>>>>>>>>> Do they challenge you (digest auth) or do they have your IP
>>>>>>>>>> address on a white list? That's a critical piece of information that only
>>>>>>>>>> your provider can supply.
>>>>>>>>>>
>>>>>>>>>> -MC
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> On Fri, Jul 29, 2011 at 9:31 PM, lloyd thomas <
>>>>>>>>>> lloydie.t at gmail.com> wrote:
>>>>>>>>>>
>>>>>>>>>>> OK Inbound working with:
>>>>>>>>>>>
>>>>>>>>>>>   <!--2circles -->
>>>>>>>>>>>     <list name="domains" default="allow">
>>>>>>>>>>>       <node type="allow" cidr="80.40.150.150/32"/>
>>>>>>>>>>>     </list>
>>>>>>>>>>>
>>>>>>>>>>> Just need to sort outbound.
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> On 30 July 2011 04:59, lloyd thomas <lloydie.t at gmail.com> wrote:
>>>>>>>>>>>
>>>>>>>>>>>> Hi, dialling in produces the following error.
>>>>>>>>>>>>
>>>>>>>>>>>> 2011-07-30 04:56:07.818936 [DEBUG] sofia.c:6517 IP 80.40.150.150
>>>>>>>>>>>> Rejected by acl "domains". Falling back to Digest auth.
>>>>>>>>>>>> 2011-07-30 04:56:07.826367 [WARNING] sofia_reg.c:1246 SIP auth
>>>>>>>>>>>> challenge (INVITE) on sofia profile 'internal' for
>>>>>>>>>>>> [01869******@172.16.XXX.XXX] from ip 80.40.150.150
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> On 30 July 2011 04:34, lloyd thomas <lloydie.t at gmail.com>wrote:
>>>>>>>>>>>>
>>>>>>>>>>>>> I am registering with a them. I could not find suitable example
>>>>>>>>>>>>> in http://wiki.freeswitch.org/wiki/SIP_Provider_Examples which
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> On 29 July 2011 21:57, Michael Collins <msc at freeswitch.org>wrote:
>>>>>>>>>>>>>
>>>>>>>>>>>>>> Are you registering with the provider or are they registering
>>>>>>>>>>>>>> with you? If they register with you then a user example is appropriate. If
>>>>>>>>>>>>>> you are registering with them then all you need is a gateway configured.
>>>>>>>>>>>>>> -MC
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> On Fri, Jul 29, 2011 at 1:40 PM, lloyd thomas <
>>>>>>>>>>>>>> lloydie.t at gmail.com> wrote:
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Sorry, example is not clear to me.
>>>>>>>>>>>>>>> I don't understand why a user config is relevant to sip
>>>>>>>>>>>>>>> registration for a provider.
>>>>>>>>>>>>>>> An example will help me more. Maybe CIDR attribute in a
>>>>>>>>>>>>>>> sip_profile gateway could help.
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> On 29 July 2011 19:55, Steven Ayre <steveayre at gmail.com>wrote:
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> Look at the cidr attribute in the user directory to
>>>>>>>>>>>>>>>> authenticate by IP:
>>>>>>>>>>>>>>>> http://wiki.freeswitch.org/wiki/Acl#Users
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> -Steve
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> On 29 July 2011 19:38, lloyd thomas <lloydie.t at gmail.com>wrote:
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> *Hi I need a little help setting up a SIP registration for
>>>>>>>>>>>>>>>>> a provider that does not use auth.*
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> *All I have is info below.*
>>>>>>>>>>>>>>>>> **
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> *
>>>>>>>>>>>>>>>>> *
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> SBC/Proxy IP:        80.40.150.150:5060
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> Authentication:      Trusted IP ? 88.221.85.33
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> Assigned DDI:        01869******, 01869******
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> DTMF Method:         RFC2833
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> Status:              Live
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> No. of trunks:       2x
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> Session Timer:       1800
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> Profile*:*             Generic (35060)
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> Apparently the following is used for *
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> [vibe]
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> type = friend
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> host = 80.40.150.150
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>>
>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110803/40e0acb8/attachment-0001.html 

From anthony.minessale at gmail.com  Wed Aug  3 20:45:17 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 3 Aug 2011 11:45:17 -0500
Subject: [Freeswitch-users] EXCHANGE_ROUTING_ERROR after ~70 transfer to
 dialplan extebsion
In-Reply-To: <0F5E8341E87D4818894C7CA1B13A24B2@abisoft.biz>
References: <576190BDB2B847A180F79DF6B311A49C@abisoft.biz>
	<CAKzWOxVeG7HTNmUsH0e6iRxgAd45u8WX=arDaq6B9B6gfgOoAA@mail.gmail.com>
	<0F5E8341E87D4818894C7CA1B13A24B2@abisoft.biz>
Message-ID: <CAKbxfG9MT1W-H=vefZtfmaBgUFi+2FQLM3Q=qeHAHdt-6qyVfQ@mail.gmail.com>

add this to inbound ext:

<action application="set" data="max_forwards=200"/>

change 200 to whatever you want.



On Wed, Aug 3, 2011 at 4:11 AM, Kozak Vladimir <vkozak at abisoft.spb.ru> wrote:
> It isn't looping. It's implimentation IVR menu. Our external system send
> commands to FS (transfer to suitable extension and other) for play files,
> say text, get dtmf, ...
> And it's possible a lot of transfers.
> I try increase max_forwards header. Exist something else?
>
>
> ----- Original Message -----
>
> From: Michael Collins
> To: FreeSWITCH Users Help
> Sent: Tuesday, August 02, 2011 11:13 PM
> Subject: Re: [Freeswitch-users] EXCHANGE_ROUTING_ERROR after ~70 transfer to
> dialplan extebsion
> If you have 70 hops through the dialplan then it's probably because you have
> a routing loop. The 70 hops acts as a "circuit breaker" to keep the call
> from looping forever.
> The next step for you is to determine why your call is looping through the
> dialplan so many times. is that truly needed?
> -MC
>
> 2011/8/2 Kozak Vladimir <vkozak at abisoft.spb.ru>
>>
>> Hi all.
>> FreeSWITCH Version 1.0.head (git-9ff8f53 2011-05-03 12-13-52 -0400)
>>
>> I have problem with transfer channel to FS extension.
>> I send from external system?to FS command as
>> ?? "SendMsg " + uniqueId + "\n" +
>> ???"call-command: execute\n" +
>> ???"execute-app-name: transfer\n" +
>> ???"execute-app-arg: " + destination + "\n\n";
>>
>> after ~70 correct transfer actions FS send BYE to my phone and send
>> HANGUP_EVENT with cause EXCHANGE_ROUTING_ERROR to my system. (FS log
>> attached)
>>
>> from FS loggs:
>> 2011-08-09 04:48:38.525293 [DEBUG] mod_sofia.c:457 Channel
>> sofia/internal/1000 at vkozak.starpoundtech.net hanging up, cause:
>> EXCHANGE_ROUTING_ERROR
>> 2011-08-09 04:48:38.525293 [DEBUG] mod_sofia.c:500 Sending BYE to
>> sofia/internal/1000 at vkozak.starpoundtech.net
>>
>> what for does limitation of transfer operations exist?
>> how can I avoid this issue?
>> is't possible to increase number of hops for transfer command?
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> ________________________________
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From avi at avimarcus.net  Wed Aug  3 20:57:11 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 3 Aug 2011 19:57:11 +0300
Subject: [Freeswitch-users] EXCHANGE_ROUTING_ERROR after ~70 transfer to
 dialplan extebsion
In-Reply-To: <CAKbxfG9MT1W-H=vefZtfmaBgUFi+2FQLM3Q=qeHAHdt-6qyVfQ@mail.gmail.com>
References: <576190BDB2B847A180F79DF6B311A49C@abisoft.biz>
	<CAKzWOxVeG7HTNmUsH0e6iRxgAd45u8WX=arDaq6B9B6gfgOoAA@mail.gmail.com>
	<0F5E8341E87D4818894C7CA1B13A24B2@abisoft.biz>
	<CAKbxfG9MT1W-H=vefZtfmaBgUFi+2FQLM3Q=qeHAHdt-6qyVfQ@mail.gmail.com>
Message-ID: <CALNZuwkveksPzq_+eSvbtL4xuX4PZsaqZdZcSxmXeobFLshq3w@mail.gmail.com>

Sorry, just started looking at this thread. It sounds quite
inefficient to have every IVR call keep hitting the dialplan for each
part of the IVR. e.g. for my calling card, they.. only transfer to the
dialplan when they actually make a call. If they get hung up on, the
IVR gets triggered (but not transferred to it) and they can hangup
with **.. which just triggers the IVR, again not hitting the dialplan.
This might work well at the volume you are currently doing, but it
sounds quite inefficient.
-Avi



On Wed, Aug 3, 2011 at 7:45 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> add this to inbound ext:
>
> <action application="set" data="max_forwards=200"/>
>
> change 200 to whatever you want.
>
>
>
> On Wed, Aug 3, 2011 at 4:11 AM, Kozak Vladimir <vkozak at abisoft.spb.ru> wrote:
>> It isn't looping. It's implimentation IVR menu. Our external system send
>> commands to FS (transfer to suitable extension and other) for play files,
>> say text, get dtmf, ...
>> And it's possible a lot of transfers.
>> I try increase max_forwards header. Exist something else?
>>
>>
>> ----- Original Message -----
>>
>> From: Michael Collins
>> To: FreeSWITCH Users Help
>> Sent: Tuesday, August 02, 2011 11:13 PM
>> Subject: Re: [Freeswitch-users] EXCHANGE_ROUTING_ERROR after ~70 transfer to
>> dialplan extebsion
>> If you have 70 hops through the dialplan then it's probably because you have
>> a routing loop. The 70 hops acts as a "circuit breaker" to keep the call
>> from looping forever.
>> The next step for you is to determine why your call is looping through the
>> dialplan so many times. is that truly needed?
>> -MC
>>
>> 2011/8/2 Kozak Vladimir <vkozak at abisoft.spb.ru>
>>>
>>> Hi all.
>>> FreeSWITCH Version 1.0.head (git-9ff8f53 2011-05-03 12-13-52 -0400)
>>>
>>> I have problem with transfer channel to FS extension.
>>> I send from external system?to FS command as
>>> ?? "SendMsg " + uniqueId + "\n" +
>>> ???"call-command: execute\n" +
>>> ???"execute-app-name: transfer\n" +
>>> ???"execute-app-arg: " + destination + "\n\n";
>>>
>>> after ~70 correct transfer actions FS send BYE to my phone and send
>>> HANGUP_EVENT with cause EXCHANGE_ROUTING_ERROR to my system. (FS log
>>> attached)
>>>
>>> from FS loggs:
>>> 2011-08-09 04:48:38.525293 [DEBUG] mod_sofia.c:457 Channel
>>> sofia/internal/1000 at vkozak.starpoundtech.net hanging up, cause:
>>> EXCHANGE_ROUTING_ERROR
>>> 2011-08-09 04:48:38.525293 [DEBUG] mod_sofia.c:500 Sending BYE to
>>> sofia/internal/1000 at vkozak.starpoundtech.net
>>>
>>> what for does limitation of transfer operations exist?
>>> how can I avoid this issue?
>>> is't possible to increase number of hops for transfer command?
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> ________________________________
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From ijurado at econcept.es  Wed Aug  3 21:24:59 2011
From: ijurado at econcept.es (Isaac Jurado)
Date: Wed, 3 Aug 2011 19:24:59 +0200
Subject: [Freeswitch-users] Hangup hooks on B legs
In-Reply-To: <CALNZuwm7Zh1FZpeavRECM6VjPUERsJODi=gRxWsXboR4hKEiLg@mail.gmail.com>
References: <CABj_V-mOmm05ycYJZRPaeePeVzB4LWdTfzjNXwM3O=Kjmrn5tQ@mail.gmail.com>
	<CALNZuwm7Zh1FZpeavRECM6VjPUERsJODi=gRxWsXboR4hKEiLg@mail.gmail.com>
Message-ID: <CABj_V-mubvz2cpCY1=ynLrAkDv088ZVURhBJ+1BiYe4XWhUyTA@mail.gmail.com>

On Wed, Aug 3, 2011 at 6:09 PM, Avi Marcus <avi at avimarcus.net> wrote:
>
> Yes, you can export it, "nolocal" to activate only on the B leg, e.g.:
>
> <action application="export" data="nolocal:api_hangup_hook=lua file.lua"/>

It works!  Thanks a lot.  Now I have a small problem, the channels seem
to be destroyed or unaccessible when the hangup hook is executed.  I
tested a dialplan with the following line:

<action application="set" data="session_in_hangup_hook=true"/>
<action application="export" data="api_hangup_hook=uuid_exists ${uuid}"/>

And the FS log reports that none of the legs exists:

2011-08-03 19:01:30.609887 [DEBUG] switch_core_state_machine.c:492
Hangup Command with no Session
uuid_exists(2bd2ce25-99fe-41fe-bcfe-d8224a9d6f18):
false
[...]
2011-08-03 19:01:30.645615 [DEBUG] switch_core_state_machine.c:492
Hangup Command with Session
uuid_exists(2bd2ce25-99fe-41fe-bcfe-d8224a9d6f18):
false

This shows that the export works, as the hook is executed twice.
Although the ${uuid} is expanded only once.  But it's curious as the
channel uuid where the hook is beign executed has not been destroyed
yet.

Cheers.

-- 
Isaac Jurado
Internet Busines Solutions eConcept


From avi at avimarcus.net  Wed Aug  3 21:39:50 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 3 Aug 2011 20:39:50 +0300
Subject: [Freeswitch-users] Hangup hooks on B legs
In-Reply-To: <CABj_V-mubvz2cpCY1=ynLrAkDv088ZVURhBJ+1BiYe4XWhUyTA@mail.gmail.com>
References: <CABj_V-mOmm05ycYJZRPaeePeVzB4LWdTfzjNXwM3O=Kjmrn5tQ@mail.gmail.com>
	<CALNZuwm7Zh1FZpeavRECM6VjPUERsJODi=gRxWsXboR4hKEiLg@mail.gmail.com>
	<CABj_V-mubvz2cpCY1=ynLrAkDv088ZVURhBJ+1BiYe4XWhUyTA@mail.gmail.com>
Message-ID: <CALNZuwkCW=ZiC9b8g9UCAkqFvFfWDyA_BywGLa_nJtvXv-2Y_Q@mail.gmail.com>

I'm not sure exactly what you are trying to do, but yes, the ${uuid}
is expanded at the time of the "hunt" in the dialplan, before the
second leg was even created.
-Avi Marcus



On Wed, Aug 3, 2011 at 8:24 PM, Isaac Jurado <ijurado at econcept.es> wrote:
> On Wed, Aug 3, 2011 at 6:09 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>
>> Yes, you can export it, "nolocal" to activate only on the B leg, e.g.:
>>
>> <action application="export" data="nolocal:api_hangup_hook=lua file.lua"/>
>
> It works! ?Thanks a lot. ?Now I have a small problem, the channels seem
> to be destroyed or unaccessible when the hangup hook is executed. ?I
> tested a dialplan with the following line:
>
> <action application="set" data="session_in_hangup_hook=true"/>
> <action application="export" data="api_hangup_hook=uuid_exists ${uuid}"/>
>
> And the FS log reports that none of the legs exists:
>
> 2011-08-03 19:01:30.609887 [DEBUG] switch_core_state_machine.c:492
> Hangup Command with no Session
> uuid_exists(2bd2ce25-99fe-41fe-bcfe-d8224a9d6f18):
> false
> [...]
> 2011-08-03 19:01:30.645615 [DEBUG] switch_core_state_machine.c:492
> Hangup Command with Session
> uuid_exists(2bd2ce25-99fe-41fe-bcfe-d8224a9d6f18):
> false
>
> This shows that the export works, as the hook is executed twice.
> Although the ${uuid} is expanded only once. ?But it's curious as the
> channel uuid where the hook is beign executed has not been destroyed
> yet.
>
> Cheers.
>
> --
> Isaac Jurado
> Internet Busines Solutions eConcept
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From ijurado at econcept.es  Wed Aug  3 21:47:46 2011
From: ijurado at econcept.es (Isaac Jurado)
Date: Wed, 3 Aug 2011 19:47:46 +0200
Subject: [Freeswitch-users] Hangup hooks on B legs
In-Reply-To: <CALNZuwkCW=ZiC9b8g9UCAkqFvFfWDyA_BywGLa_nJtvXv-2Y_Q@mail.gmail.com>
References: <CABj_V-mOmm05ycYJZRPaeePeVzB4LWdTfzjNXwM3O=Kjmrn5tQ@mail.gmail.com>
	<CALNZuwm7Zh1FZpeavRECM6VjPUERsJODi=gRxWsXboR4hKEiLg@mail.gmail.com>
	<CABj_V-mubvz2cpCY1=ynLrAkDv088ZVURhBJ+1BiYe4XWhUyTA@mail.gmail.com>
	<CALNZuwkCW=ZiC9b8g9UCAkqFvFfWDyA_BywGLa_nJtvXv-2Y_Q@mail.gmail.com>
Message-ID: <CABj_V-=uB_X77ZUrcYDU_Rx0yJLLwhzMAU3xtfoiOy=PeEwOdA@mail.gmail.com>

On Wed, Aug 3, 2011 at 7:39 PM, Avi Marcus <avi at avimarcus.net> wrote:
>
> I'm not sure exactly what you are trying to do, but yes, the ${uuid}
> is expanded at the time of the "hunt" in the dialplan, before the
> second leg was even created.

My original intention was to set a channel variable from the
api_hangup_hook:

<action application="export" data="nolocal:api_hangup_hook=uuid_setvar
${uuid} fooooo baaaaar"/>

But as the expanded ${uuid} does not exist, the uuid_setvar command
complains.

Cheers.

-- 
Isaac Jurado
Internet Busines Solutions eConcept


From ijurado at econcept.es  Wed Aug  3 21:52:25 2011
From: ijurado at econcept.es (Isaac Jurado)
Date: Wed, 3 Aug 2011 19:52:25 +0200
Subject: [Freeswitch-users] CDR ordering?
Message-ID: <CABj_V-mg5UwBqUHKogXC8RBnDh5M1ixMMdA4xL2=S=PmrjCeVQ@mail.gmail.com>

Hello again,

In a simple call, our development server seems to generate the B-leg CDR
(XML fomat) before the A-leg CDR consistently.  Ironically, our
test server seems to do it the other way around.  We believe that CDR
ordering should not be taken for granted, but I wanted to make sure of
it.

Are there any ordering guarnatees for CDR generation (when b-leg logging
is enabled in mod_xml_cdr)?

Thanks and sorry for the noise.

-- 
Isaac Jurado
Internet Busines Solutions eConcept


From avi at avimarcus.net  Wed Aug  3 21:56:00 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 3 Aug 2011 20:56:00 +0300
Subject: [Freeswitch-users] Hangup hooks on B legs
In-Reply-To: <CABj_V-=uB_X77ZUrcYDU_Rx0yJLLwhzMAU3xtfoiOy=PeEwOdA@mail.gmail.com>
References: <CABj_V-mOmm05ycYJZRPaeePeVzB4LWdTfzjNXwM3O=Kjmrn5tQ@mail.gmail.com>
	<CALNZuwm7Zh1FZpeavRECM6VjPUERsJODi=gRxWsXboR4hKEiLg@mail.gmail.com>
	<CABj_V-mubvz2cpCY1=ynLrAkDv088ZVURhBJ+1BiYe4XWhUyTA@mail.gmail.com>
	<CALNZuwkCW=ZiC9b8g9UCAkqFvFfWDyA_BywGLa_nJtvXv-2Y_Q@mail.gmail.com>
	<CABj_V-=uB_X77ZUrcYDU_Rx0yJLLwhzMAU3xtfoiOy=PeEwOdA@mail.gmail.com>
Message-ID: <CALNZuwkszOa6mhUA7UuL7NM9ZHDKSL7NcLxA2p9G0MP6-h_GQQ@mail.gmail.com>

If the channel is hung up, then it's hung up.. that's what the hangup
hook is, afaik.
If you look at the wiki page for this:
http://wiki.freeswitch.org/wiki/Variable_session_in_hangup_hook
it only says that you can "access" the variable, not that the channel
still exists or that you can set them.
Why do you need to use a hangup hook to set a variable? can't you set
it at the start? Or import B leg into A leg's CDRs?

-Avi Marcus

718-989-9485 (USA)
054-844-3271 (Israel Kosher)
077-228-5055 (Israel Landline)
020-3519-3606?(UK)



On Wed, Aug 3, 2011 at 8:47 PM, Isaac Jurado <ijurado at econcept.es> wrote:
> On Wed, Aug 3, 2011 at 7:39 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>
>> I'm not sure exactly what you are trying to do, but yes, the ${uuid}
>> is expanded at the time of the "hunt" in the dialplan, before the
>> second leg was even created.
>
> My original intention was to set a channel variable from the
> api_hangup_hook:
>
> <action application="export" data="nolocal:api_hangup_hook=uuid_setvar
> ${uuid} fooooo baaaaar"/>
>
> But as the expanded ${uuid} does not exist, the uuid_setvar command
> complains.
>
> Cheers.
>
> --
> Isaac Jurado
> Internet Busines Solutions eConcept
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From avi at avimarcus.net  Wed Aug  3 21:57:21 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 3 Aug 2011 20:57:21 +0300
Subject: [Freeswitch-users] CDR ordering?
In-Reply-To: <CABj_V-mg5UwBqUHKogXC8RBnDh5M1ixMMdA4xL2=S=PmrjCeVQ@mail.gmail.com>
References: <CABj_V-mg5UwBqUHKogXC8RBnDh5M1ixMMdA4xL2=S=PmrjCeVQ@mail.gmail.com>
Message-ID: <CALNZuw=ATa3SxEfikK+ZBthH1HC0AP+gVPXhascOk+H0TN7ROQ@mail.gmail.com>

You can turn on  <param name="prefix-a-leg" value="true"/> in your
xml_cdr.xml and then you'll know which leg you are dealing with...

-Avi


On Wed, Aug 3, 2011 at 8:52 PM, Isaac Jurado <ijurado at econcept.es> wrote:
> Hello again,
>
> In a simple call, our development server seems to generate the B-leg CDR
> (XML fomat) before the A-leg CDR consistently. ?Ironically, our
> test server seems to do it the other way around. ?We believe that CDR
> ordering should not be taken for granted, but I wanted to make sure of
> it.
>
> Are there any ordering guarnatees for CDR generation (when b-leg logging
> is enabled in mod_xml_cdr)?
>
> Thanks and sorry for the noise.
>
> --
> Isaac Jurado
> Internet Busines Solutions eConcept
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From ijurado at econcept.es  Wed Aug  3 23:14:34 2011
From: ijurado at econcept.es (Isaac Jurado)
Date: Wed, 3 Aug 2011 21:14:34 +0200
Subject: [Freeswitch-users] Hangup hooks on B legs
In-Reply-To: <CALNZuwkszOa6mhUA7UuL7NM9ZHDKSL7NcLxA2p9G0MP6-h_GQQ@mail.gmail.com>
References: <CABj_V-mOmm05ycYJZRPaeePeVzB4LWdTfzjNXwM3O=Kjmrn5tQ@mail.gmail.com>
	<CALNZuwm7Zh1FZpeavRECM6VjPUERsJODi=gRxWsXboR4hKEiLg@mail.gmail.com>
	<CABj_V-mubvz2cpCY1=ynLrAkDv088ZVURhBJ+1BiYe4XWhUyTA@mail.gmail.com>
	<CALNZuwkCW=ZiC9b8g9UCAkqFvFfWDyA_BywGLa_nJtvXv-2Y_Q@mail.gmail.com>
	<CABj_V-=uB_X77ZUrcYDU_Rx0yJLLwhzMAU3xtfoiOy=PeEwOdA@mail.gmail.com>
	<CALNZuwkszOa6mhUA7UuL7NM9ZHDKSL7NcLxA2p9G0MP6-h_GQQ@mail.gmail.com>
Message-ID: <CABj_V-nr4gP9B5uRY+9hRTwtd2GEy1on-BBXH9wdG1qJdyJXKQ@mail.gmail.com>

On Wed, Aug 3, 2011 at 7:56 PM, Avi Marcus <avi at avimarcus.net> wrote:
>
> If the channel is hung up, then it's hung up.. that's what the hangup
> hook is, afaik.  If you look at the wiki page for this:
> http://wiki.freeswitch.org/wiki/Variable_session_in_hangup_hook
> it only says that you can "access" the variable, not that the channel
> still exists or that you can set them.

It looks that the session object is only available through bindings like
mod_lua.  I'll have to check

> Why do you need to use a hangup hook to set a variable? can't you set
> it at the start? Or import B leg into A leg's CDRs?

It's a bit tricker than that.  We are trying to implement a stateless
CDR processing and billing mechanism that works also across call
transfers.

As I said, we will probably end up defining a Lua hook.

Thanks again for your quick response.

Cheers.

-- 
Isaac Jurado
Internet Busines Solutions eConcept


From msc at freeswitch.org  Wed Aug  3 23:36:17 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 3 Aug 2011 12:36:17 -0700
Subject: [Freeswitch-users] Hangup hooks on B legs
In-Reply-To: <CABj_V-nr4gP9B5uRY+9hRTwtd2GEy1on-BBXH9wdG1qJdyJXKQ@mail.gmail.com>
References: <CABj_V-mOmm05ycYJZRPaeePeVzB4LWdTfzjNXwM3O=Kjmrn5tQ@mail.gmail.com>
	<CALNZuwm7Zh1FZpeavRECM6VjPUERsJODi=gRxWsXboR4hKEiLg@mail.gmail.com>
	<CABj_V-mubvz2cpCY1=ynLrAkDv088ZVURhBJ+1BiYe4XWhUyTA@mail.gmail.com>
	<CALNZuwkCW=ZiC9b8g9UCAkqFvFfWDyA_BywGLa_nJtvXv-2Y_Q@mail.gmail.com>
	<CABj_V-=uB_X77ZUrcYDU_Rx0yJLLwhzMAU3xtfoiOy=PeEwOdA@mail.gmail.com>
	<CALNZuwkszOa6mhUA7UuL7NM9ZHDKSL7NcLxA2p9G0MP6-h_GQQ@mail.gmail.com>
	<CABj_V-nr4gP9B5uRY+9hRTwtd2GEy1on-BBXH9wdG1qJdyJXKQ@mail.gmail.com>
Message-ID: <CAKzWOxXnKTsrScynot_jH1r76Cwjn7TkPgeYEcFv3YRQ7ZqM7w@mail.gmail.com>

Just remember that once the call ends, the "session" object you have is
read-only. You can still do interesting things, but you cannot alter CDRs
after the call ends, at least not in a session hangup hook. You are better
off with XML CDRs and post-call logic. With all of the screwy transfer
scenarios you have to account for you will absolutely need to do some
post-processing logic. Sounds like lots of "fun". :P

-MC

On Wed, Aug 3, 2011 at 12:14 PM, Isaac Jurado <ijurado at econcept.es> wrote:

> On Wed, Aug 3, 2011 at 7:56 PM, Avi Marcus <avi at avimarcus.net> wrote:
> >
> > If the channel is hung up, then it's hung up.. that's what the hangup
> > hook is, afaik.  If you look at the wiki page for this:
> > http://wiki.freeswitch.org/wiki/Variable_session_in_hangup_hook
> > it only says that you can "access" the variable, not that the channel
> > still exists or that you can set them.
>
> It looks that the session object is only available through bindings like
> mod_lua.  I'll have to check
>
> > Why do you need to use a hangup hook to set a variable? can't you set
> > it at the start? Or import B leg into A leg's CDRs?
>
> It's a bit tricker than that.  We are trying to implement a stateless
> CDR processing and billing mechanism that works also across call
> transfers.
>
> As I said, we will probably end up defining a Lua hook.
>
> Thanks again for your quick response.
>
> Cheers.
>
> --
> Isaac Jurado
> Internet Busines Solutions eConcept
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110803/4fbb7918/attachment.html 

From ijurado at econcept.es  Thu Aug  4 00:00:57 2011
From: ijurado at econcept.es (Isaac Jurado)
Date: Wed, 3 Aug 2011 22:00:57 +0200
Subject: [Freeswitch-users] Hangup hooks on B legs
In-Reply-To: <CAKzWOxXnKTsrScynot_jH1r76Cwjn7TkPgeYEcFv3YRQ7ZqM7w@mail.gmail.com>
References: <CABj_V-mOmm05ycYJZRPaeePeVzB4LWdTfzjNXwM3O=Kjmrn5tQ@mail.gmail.com>
	<CALNZuwm7Zh1FZpeavRECM6VjPUERsJODi=gRxWsXboR4hKEiLg@mail.gmail.com>
	<CABj_V-mubvz2cpCY1=ynLrAkDv088ZVURhBJ+1BiYe4XWhUyTA@mail.gmail.com>
	<CALNZuwkCW=ZiC9b8g9UCAkqFvFfWDyA_BywGLa_nJtvXv-2Y_Q@mail.gmail.com>
	<CABj_V-=uB_X77ZUrcYDU_Rx0yJLLwhzMAU3xtfoiOy=PeEwOdA@mail.gmail.com>
	<CALNZuwkszOa6mhUA7UuL7NM9ZHDKSL7NcLxA2p9G0MP6-h_GQQ@mail.gmail.com>
	<CABj_V-nr4gP9B5uRY+9hRTwtd2GEy1on-BBXH9wdG1qJdyJXKQ@mail.gmail.com>
	<CAKzWOxXnKTsrScynot_jH1r76Cwjn7TkPgeYEcFv3YRQ7ZqM7w@mail.gmail.com>
Message-ID: <CABj_V-kY4TasHg=AYAmf+T1b4OD1vczw+oh224bQBiQhVVCYZw@mail.gmail.com>

On Wed, Aug 3, 2011 at 9:36 PM, Michael Collins <msc at freeswitch.org> wrote:
>
> Just remember that once the call ends, the "session" object you have
> is read-only.

Good to know.

> You can still do interesting things, but you cannot alter CDRs after
> the call ends, at least not in a session hangup hook.

These days we are trying to modify other channels variables from the
hangup hook (channels like the ones involved in whatever kind of
transfer).

> You are better off with XML CDRs and post-call logic. With all of the
> screwy transfer scenarios you have to account for you will absolutely
> need to do some post-processing logic. Sounds like lots of "fun". :P

Fun indeed.  Our ideal target would be to have each CDR self-contained.
In general, we only want to bill the A-leg CDRs as they are the
originators.  However, some B-leg CDRs need to be billed if they were
created by a transfer and some A-leg CDRs can be discarded if they
participated in an attended transfer.

As you rightfully mentioned, when transfers are chained is where the
obscure scenarios arise.

If our experiments provide a working proof of concept, we have two
choices left:

    1. Subscribe to CHANNEL_STATE events in order to catch transfer
       directly from event socket.  But we would be mixing mod_xml_cdr,
       mod_xml_curl and event socket altogether, so we want to avoid
       that.

    2. Forget about stateless CDRs and pray (or inspect the source code)
       for mod_xml_cdr generating originator CDRs last, always.

If anybody is interested, I can keep him/her informed.

Cheers.

-- 
Isaac Jurado
Internet Busines Solutions eConcept


From ijurado at econcept.es  Thu Aug  4 00:02:02 2011
From: ijurado at econcept.es (Isaac Jurado)
Date: Wed, 3 Aug 2011 22:02:02 +0200
Subject: [Freeswitch-users] Hangup hooks on B legs
In-Reply-To: <CABj_V-kY4TasHg=AYAmf+T1b4OD1vczw+oh224bQBiQhVVCYZw@mail.gmail.com>
References: <CABj_V-mOmm05ycYJZRPaeePeVzB4LWdTfzjNXwM3O=Kjmrn5tQ@mail.gmail.com>
	<CALNZuwm7Zh1FZpeavRECM6VjPUERsJODi=gRxWsXboR4hKEiLg@mail.gmail.com>
	<CABj_V-mubvz2cpCY1=ynLrAkDv088ZVURhBJ+1BiYe4XWhUyTA@mail.gmail.com>
	<CALNZuwkCW=ZiC9b8g9UCAkqFvFfWDyA_BywGLa_nJtvXv-2Y_Q@mail.gmail.com>
	<CABj_V-=uB_X77ZUrcYDU_Rx0yJLLwhzMAU3xtfoiOy=PeEwOdA@mail.gmail.com>
	<CALNZuwkszOa6mhUA7UuL7NM9ZHDKSL7NcLxA2p9G0MP6-h_GQQ@mail.gmail.com>
	<CABj_V-nr4gP9B5uRY+9hRTwtd2GEy1on-BBXH9wdG1qJdyJXKQ@mail.gmail.com>
	<CAKzWOxXnKTsrScynot_jH1r76Cwjn7TkPgeYEcFv3YRQ7ZqM7w@mail.gmail.com>
	<CABj_V-kY4TasHg=AYAmf+T1b4OD1vczw+oh224bQBiQhVVCYZw@mail.gmail.com>
Message-ID: <CABj_V-n7QZcbfNgz8xg8K-kPTmhKKOU11j1ee8=6w8uaj+=vGg@mail.gmail.com>

On Wed, Aug 3, 2011 at 10:00 PM, Isaac Jurado <ijurado at econcept.es> wrote:
>
> If our experiments provide a working proof of concept, we have two
> choices left:

That should read: ?if our experiments DON'T provide?

Regards.

-- 
Isaac Jurado
Internet Busines Solutions eConcept


From msc at freeswitch.org  Thu Aug  4 00:38:04 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 3 Aug 2011 13:38:04 -0700
Subject: [Freeswitch-users] Hangup hooks on B legs
In-Reply-To: <CABj_V-n7QZcbfNgz8xg8K-kPTmhKKOU11j1ee8=6w8uaj+=vGg@mail.gmail.com>
References: <CABj_V-mOmm05ycYJZRPaeePeVzB4LWdTfzjNXwM3O=Kjmrn5tQ@mail.gmail.com>
	<CALNZuwm7Zh1FZpeavRECM6VjPUERsJODi=gRxWsXboR4hKEiLg@mail.gmail.com>
	<CABj_V-mubvz2cpCY1=ynLrAkDv088ZVURhBJ+1BiYe4XWhUyTA@mail.gmail.com>
	<CALNZuwkCW=ZiC9b8g9UCAkqFvFfWDyA_BywGLa_nJtvXv-2Y_Q@mail.gmail.com>
	<CABj_V-=uB_X77ZUrcYDU_Rx0yJLLwhzMAU3xtfoiOy=PeEwOdA@mail.gmail.com>
	<CALNZuwkszOa6mhUA7UuL7NM9ZHDKSL7NcLxA2p9G0MP6-h_GQQ@mail.gmail.com>
	<CABj_V-nr4gP9B5uRY+9hRTwtd2GEy1on-BBXH9wdG1qJdyJXKQ@mail.gmail.com>
	<CAKzWOxXnKTsrScynot_jH1r76Cwjn7TkPgeYEcFv3YRQ7ZqM7w@mail.gmail.com>
	<CABj_V-kY4TasHg=AYAmf+T1b4OD1vczw+oh224bQBiQhVVCYZw@mail.gmail.com>
	<CABj_V-n7QZcbfNgz8xg8K-kPTmhKKOU11j1ee8=6w8uaj+=vGg@mail.gmail.com>
Message-ID: <CAKzWOxW5q2D_hPLdndgTi674ASjB00pDudhXfaUeKnO-06v3NA@mail.gmail.com>

One other thing you might want to do is look at the callflows at the end of
each xml cdr. they paint a picture of "what really happened" on the call and
might give you the information that you need.
-MC

On Wed, Aug 3, 2011 at 1:02 PM, Isaac Jurado <ijurado at econcept.es> wrote:

> On Wed, Aug 3, 2011 at 10:00 PM, Isaac Jurado <ijurado at econcept.es> wrote:
> >
> > If our experiments provide a working proof of concept, we have two
> > choices left:
>
> That should read: ?if our experiments DON'T provide?
>
> Regards.
>
> --
> Isaac Jurado
> Internet Busines Solutions eConcept
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110803/a80a953b/attachment.html 

From spencer at 5ninesolutions.com  Thu Aug  4 04:54:01 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Wed, 3 Aug 2011 20:54:01 -0400
Subject: [Freeswitch-users] mod_com_g729 not cleaning up channels
Message-ID: <BBF16F90-E376-4B8D-A8C7-82BCEB99A219@5ninesolutions.com>

Hello all,
After a recent update to current git head, I'm noticing the following behavior:

freeswitch at internal> show channels

0 total.

freeswitch at internal> g729_used
0:1

freeswitch at internal> g729_info
Permitted G729 channels: 4
Encoders in use: 0
Decoders in use: 1

Is there any reason a decoder would still be in use when there are no channels in use?  I'm using mod_com_g729 194

Thanks,
Spencer




From anthony.minessale at gmail.com  Thu Aug  4 08:44:27 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 3 Aug 2011 23:44:27 -0500
Subject: [Freeswitch-users] Receiving junk ESL events
In-Reply-To: <1312355407424-6647776.post@n2.nabble.com>
References: <1312355407424-6647776.post@n2.nabble.com>
Message-ID: <CAKbxfG9CO6dKCVXs1G37C-DQsApytY-f_5se9sEKLFz0m1MOpw@mail.gmail.com>

the events are async so you can't count on the next event returned to
be a reply to one you sent.
you need to use the api command and uuid_kill to do the hangup

char cmd[128];
snprintf(cmd, sizeof(cmd), "api uuid_kill %s\n\n", some_uuid);

esl_send_recv(&handle, cmd);

this function actually guarantees a reply

On Wed, Aug 3, 2011 at 2:10 AM, dma <mays.david at gmail.com> wrote:
> Hello All,
>
> I have some problem with receiving and handling ESL events.
>
> My code is like:
>
> void handle_eslswitch_event_plain(esl_handle_t *handle, esl_event_t *event)
> {
> ? ? ? ?... ...
> ? ? ? ?TRACE_DEBUG("ESL event [%d] %s: UUID: %s",
> ? ? ? ? ? ? ? ?event->event_id, fs_get_event_header(event, "Event-Name"),
> uniqueid);
> ? ? ? ? print_event(event);
> ? ? ? ?... ...
> }
>
> void handle_event(esl_handle_t *handle, esl_event_t *last_event)
> {
> ? ? ? ?... ...
> ? ? ? ?if (!strcasecmp(type, "text/event-plain")) {
> ? ? ? ? ? ?handle_eslswitch_event_plain(handle, handle->last_ievent);
> ? ? ? ?}
> ? ? ? ?... ...
> }
>
> int main()
> {
> ? ? ? ?... ...
> ? ? ? ?//receive from event socket
> ? ? ? ?if (handle.last_event) handle_event(&handle, handle.last_event);
> ? ? ? ?... ...
>
> Usually I receive correct events. But I occasionally receive incorrect
> event. See below the 1st is the normal event and the 2nd is the wrong event:
>
> <2011-07-29 14:25:29> [DEBUG] ESL event [8] CHANNEL_HANGUP_COMPLETE: UUID:
> f320e4a6-db23-46d0-8d89-9957eccbd4c9
> <2011-07-29 14:25:29> [DEBUG] RECV EVENT
> Event-Name: CHANNEL_HANGUP_COMPLETE
> Core-UUID: 26b77cba-8fdd-486d-90ec-6844bca58c72
> FreeSWITCH-Hostname: fs01
> FreeSWITCH-IPv4: 10.1.1.46
> FreeSWITCH-IPv6: ::1
> Event-Date-Local: 2011-07-29 14:25:29
>
> However, usually after failure in executing "hangup" and getting "-ERR ..."
> in the event->last_sr_reply, I have wrong event, but not always (an -ERR
> returned from executing "hangup" doesn't always result in a wrong event).
> Here it is:
>
> <2011-07-29 14:25:29> [NOTICE] -ERR invalid session id
> [f320e4a6-db23-46d0-8d89-9957eccbd4c9]
> <2011-07-28 15:55:23> [DEBUG] ESL event [0] : UUID:
> <2011-07-28 15:55:23> [DEBUG] RECV EVENT
> Content-Length: 6485
> Content-Type: text/event-plain
>
> Event-Name: CHANNEL_HANGUP
> Core-UUID: 26b77cba-8fdd-486d-90ec-6844bca58c72
> FreeSWITCH-Hostname: fs01
> FreeSWITCH-IPv4: 10.1.1.46
> FreeSWITCH-IPv6: %3A%3A1
> Event-Date-Local: 2011-07-28%2015%3A55%3A23
>
> In the above case, both Event ID and UUID are invalid.
>
> I think i need more information regarding how to use the data elements in
> ESL event, especially how to use the following:
>
> ? ?char last_reply[1024];
> ? ?/*! Las command reply when called with esl_send_recv */
> ? ?char last_sr_reply[1024];
> ? ?/*! Last event received. Only populated when **save_event is NULL */
> ? ?esl_event_t *last_event;
> ? ?/*! Last event received when called by esl_send_recv */
> ? ?esl_event_t *last_sr_event;
> ? ?/*! This will hold already processed events queued by esl_recv_event */
> ? ?esl_event_t *race_event;
> ? ?/*! Events that have content-type == text/plain and a body */
> ? ?esl_event_t *last_ievent;
>
> My questions, what event should I check if I am interested in only
> channel/call related events, and how? Do I check handle.last_event, or
> handle.last_ievent, or what?
>
> Else, when checking command reply, do I check last_sr_reply, or
> last_sr_event->body?
>
> Please kindly advice.
>
> Thanks,
> D.Ma
>
>
>
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Receiving-junk-ESL-events-tp6647776p6647776.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From dhairya.blogs at gmail.com  Thu Aug  4 09:18:22 2011
From: dhairya.blogs at gmail.com (Dhairya Vora)
Date: Thu, 4 Aug 2011 10:48:22 +0530
Subject: [Freeswitch-users] freeswitch 1.0.7 required
Message-ID: <CA+P_fE_etZKPCY+y7dioV5CBiK2XtPPRPfQNbG8dz8=qaJLKXw@mail.gmail.com>

I am installing freeswitch. I read that latest version is 1.0.7 but in git
it is not available. They say that it is at
*http://latest.freeswitch.org/*but I am unable to open that url.
Either server is down, or url is changed.
Any other location to download freeswitch 1.0.7 (or the latest version) ?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110804/b9c00f9e/attachment.html 

From krice at freeswitch.org  Thu Aug  4 09:21:35 2011
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 04 Aug 2011 00:21:35 -0500
Subject: [Freeswitch-users] freeswitch 1.0.7 required
In-Reply-To: <CA+P_fE_etZKPCY+y7dioV5CBiK2XtPPRPfQNbG8dz8=qaJLKXw@mail.gmail.com>
Message-ID: <CA5F968F.1A0ED%krice@freeswitch.org>

Just get git head... That?s where you want to be anyway... Yes its stable...
But as with any new deployment you should test it to make sure it meets your
needs



On 8/4/11 12:18 AM, "Dhairya Vora" <dhairya.blogs at gmail.com> wrote:

> I am installing freeswitch. I read that latest version is 1.0.7 but in git it
> is not available. They say that it is at http://latest.freeswitch.org/ but I
> am unable to open that url. Either server is down, or url is changed. Any
> other location to download freeswitch 1.0.7 (or the latest version) ?
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110804/7aab2249/attachment.html 

From dhairya.blogs at gmail.com  Thu Aug  4 09:33:26 2011
From: dhairya.blogs at gmail.com (Dhairya Vora)
Date: Thu, 4 Aug 2011 11:03:26 +0530
Subject: [Freeswitch-users] freeswitch 1.0.7 required
In-Reply-To: <CA5F968F.1A0ED%krice@freeswitch.org>
References: <CA+P_fE_etZKPCY+y7dioV5CBiK2XtPPRPfQNbG8dz8=qaJLKXw@mail.gmail.com>
	<CA5F968F.1A0ED%krice@freeswitch.org>
Message-ID: <CA+P_fE_UgehuXmdtj8DOtoHYToRrUQg7tjpkCswe0-SgPbD10Q@mail.gmail.com>

Here (http://wiki.freeswitch.org/wiki/Download_FreeSWITCH) they say that in
Git, 1.0.7 is not available.

(FYI: here (
http://lists.freeswitch.org/pipermail/freeswitch-users/2011-January/067446.html)
they say that "git does not give odd number version." Really??)

By the way, I installed using git and I thought that it is 1.0.6. Now it
shows 1.0.head. just see the output.
****************************************************************************************************
freeswitch at localhost.localdomain> version

FreeSWITCH Version 1.0.head (git-4b1bb61 2011-08-01 15-43-07 -0500)
****************************************************************************************************



On Thu, Aug 4, 2011 at 10:51 AM, Ken Rice <krice at freeswitch.org> wrote:

>  Just get git head... That?s where you want to be anyway... Yes its
> stable... But as with any new deployment you should test it to make sure it
> meets your needs
>
>
>
>
> On 8/4/11 12:18 AM, "Dhairya Vora" <dhairya.blogs at gmail.com> wrote:
>
> I am installing freeswitch. I read that latest version is 1.0.7 but in git
> it is not available. They say that it is at *http://latest.freeswitch.org/
> * but I am unable to open that url. Either server is down, or url is
> changed. Any other location to download freeswitch 1.0.7 (or the latest
> version) ?
>
> ------------------------------
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110804/fc165d0a/attachment.html 

From krice at freeswitch.org  Thu Aug  4 09:38:06 2011
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 04 Aug 2011 00:38:06 -0500
Subject: [Freeswitch-users] freeswitch 1.0.7 required
In-Reply-To: <CA+P_fE_UgehuXmdtj8DOtoHYToRrUQg7tjpkCswe0-SgPbD10Q@mail.gmail.com>
Message-ID: <CA5F9A6E.1A0F0%krice@freeswitch.org>

1.0.X , X is a snapshot of git at some point in time... Its a release
version... The problem is people don?t test when the main developers say
they are getting ready for a new release, so they test as best as they can
an cut a release...

20 minutes later, a dozen bugs are reported...  Because people expect
everyone else to test... The way to fix this was decided that there will be
OLD version (and by old I mean really old) and there will be head for the
most part... That forces people to test... And 99.99% of the time if you do
fine something broken ?make current? fixes the issue...


And yes 1.0.head is what comes from git... Notice that git-HEXNUMBER...
That?s the actual give hash version...
Have a nice day

K


On 8/4/11 12:33 AM, "Dhairya Vora" <dhairya.blogs at gmail.com> wrote:

> Here (http://wiki.freeswitch.org/wiki/Download_FreeSWITCH) they say that in
> Git, 1.0.7 is not available.
> 
> (FYI: here 
> (http://lists.freeswitch.org/pipermail/freeswitch-users/2011-January/067446.ht
> ml) they say that "git does not give odd number version." Really??)
> 
> By the way, I installed using git and I thought that it is 1.0.6. Now it shows
> 1.0.head. just see the output.
> ******************************************************************************
> **********************
> freeswitch at localhost.localdomain> version
> 
> FreeSWITCH Version 1.0.head (git-4b1bb61 2011-08-01 15-43-07 -0500)
> ******************************************************************************
> **********************
> 
> 
> 
> On Thu, Aug 4, 2011 at 10:51 AM, Ken Rice <krice at freeswitch.org> wrote:
>> Just get git head... That?s where you want to be anyway... Yes its stable...
>> But as with any new deployment you should test it to make sure it meets your
>> needs
>> 
>> 
>> 
>> 
>> On 8/4/11 12:18 AM, "Dhairya Vora" <dhairya.blogs at gmail.com
>> <http://dhairya.blogs at gmail.com> > wrote:
>> 
>>> I am installing freeswitch. I read that latest version is 1.0.7 but in git
>>> it is not available. They say that it is at http://latest.freeswitch.org/
>>> but I am unable to open that url. Either server is down, or url is changed.
>>> Any other location to download freeswitch 1.0.7 (or the latest version) ?
>>> 
>>> 
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> <http://FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110804/93215dca/attachment.html 

From bino at indoakses-online.com  Thu Aug  4 10:30:33 2011
From: bino at indoakses-online.com (bino oetomo)
Date: Thu, 04 Aug 2011 13:30:33 +0700
Subject: [Freeswitch-users] How to for video call ?
In-Reply-To: <4E38CA09.8050804@indoakses-online.com>
References: <4E3655DF.4060207@indoakses-online.com>	<CAHKqMb60DcNm8P8FztXyJOhmf0h6wQEMgZw_ami_vbWho2CP8Q@mail.gmail.com>	<4E376A92.9060201@indoakses-online.com>	<CALXCt0ouywkiXe-cit_q4=JmVCk9nwMdOq7Z-EM5ZG-6kcHHmw@mail.gmail.com>
	<4E38CA09.8050804@indoakses-online.com>
Message-ID: <4E3A3C89.1050405@indoakses-online.com>

Dear All
C/q Giovani and Nandy

Thx for all your enlightment.
Now it's run using fusionPBX.


I Found there is at least 2 factor that I need to make sure :
1. Domains : it's done
2. Codec : for My FreeSwitch system .. it need that both party used same 
v-codec.

Is there some way to :
1. Make freeswitch do transcoding between client ... so that each client 
didn't need to set same codec betwen them. Or ..
2. set freeswitch to force clients to use same codec (i.e : H263-1998)

Sincerely
-bino-



From gcd at i.ph  Thu Aug  4 10:43:40 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Thu, 4 Aug 2011 14:43:40 +0800
Subject: [Freeswitch-users] How to for video call ?
In-Reply-To: <4E3A3C89.1050405@indoakses-online.com>
References: <4E3655DF.4060207@indoakses-online.com>
	<CAHKqMb60DcNm8P8FztXyJOhmf0h6wQEMgZw_ami_vbWho2CP8Q@mail.gmail.com>
	<4E376A92.9060201@indoakses-online.com>
	<CALXCt0ouywkiXe-cit_q4=JmVCk9nwMdOq7Z-EM5ZG-6kcHHmw@mail.gmail.com>
	<4E38CA09.8050804@indoakses-online.com>
	<4E3A3C89.1050405@indoakses-online.com>
Message-ID: <CAHKqMb6JO3x93PtUbGfrrZ_LKDgxwLSWKC11C6jkQfMx0GC2jg@mail.gmail.com>

audio transcoding is possible but not for video.
http://wiki.freeswitch.org/wiki/Codecs#Pass-through_video_codecs


On Thu, Aug 4, 2011 at 2:30 PM, bino oetomo <bino at indoakses-online.com>wrote:

> Dear All
> C/q Giovani and Nandy
>
> Thx for all your enlightment.
> Now it's run using fusionPBX.
>
>
> I Found there is at least 2 factor that I need to make sure :
> 1. Domains : it's done
> 2. Codec : for My FreeSwitch system .. it need that both party used same
> v-codec.
>
> Is there some way to :
> 1. Make freeswitch do transcoding between client ... so that each client
> didn't need to set same codec betwen them. Or ..
> 2. set freeswitch to force clients to use same codec (i.e : H263-1998)
>
> Sincerely
> -bino-
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110804/a25055ce/attachment.html 

From dujinfang at gmail.com  Thu Aug  4 11:29:06 2011
From: dujinfang at gmail.com (Seven Du)
Date: Thu, 4 Aug 2011 15:29:06 +0800
Subject: [Freeswitch-users] Question on caller id
Message-ID: <6D2A7E4AD25D43AE94995EAC9B45BD86@gmail.com>

Hi,

Just found a weird problem. I originate user/1001 &echo, and the caller id in ESL (Caller-Caller-ID-Number) shows "internal/sip:1001 at 192.168.7.7:41886;rinstance=1b34eba11b046e53" while I expect "1001".

How can I get 1001 as caller id?

SIP trace:

http://pastebin.freeswitch.org/16970


Registrations:
=================================================================
Call-ID: NzhhYTExOWRiMjY0ZGY3Y2MxNDg3ZjJjMTA4ZGZlNTM.
User:  1001 at 192.168.7.7
Contact: "user" <sip:1001 at 192.168.7.7:41886;rinstance=1b34eba11b046e53>
Agent: Bria 3 release 3.1 stamp 58312
Status:  Registered(UDP)(unknown) EXP(2011-08-04 16:01:38) EXPSECS(2324)
Host:  seven-air.local
IP:  192.168.7.7
Port:  41886
Auth-User: 1001
Auth-Realm:  192.168.7.7
MWI-Account: 1001 at 192.168.7.7

Total items returned: 1
=================================================================

Thanks.

-- 
Seven Du
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj: http://www.freeswitch.org.cn
Sent with Sparrow

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110804/a7268a2f/attachment.html 

From mays.david at gmail.com  Thu Aug  4 11:32:44 2011
From: mays.david at gmail.com (David Ma)
Date: Thu, 4 Aug 2011 15:32:44 +0800
Subject: [Freeswitch-users] Receiving junk ESL events
In-Reply-To: <CAKbxfG9CO6dKCVXs1G37C-DQsApytY-f_5se9sEKLFz0m1MOpw@mail.gmail.com>
References: <1312355407424-6647776.post@n2.nabble.com>
	<CAKbxfG9CO6dKCVXs1G37C-DQsApytY-f_5se9sEKLFz0m1MOpw@mail.gmail.com>
Message-ID: <CAMLBmRKK5jPiUizKOuE_56Z7tLE7eop46j3N4=_AZsKPUtvNRA@mail.gmail.com>

Hi Anthony,

Thanks a lot for the solution. I shall try uuid_kill.

I am currently using "sendmsg" to "hangup" a UUID call, like this:

        snprintf(cmd, sizeof(cmd), "sendmsg %s\ncall-command:
hangup\nhangup-cause: ....

I send the message using esl_send_recv() and check handle.last_sr_reply
immediately after the function returns. Now I realize this is wrong because
"sendmsg" is async and "api uuid_kill" can guarantee a reply.

Thanks again,
D.Ma

On Thu, Aug 4, 2011 at 12:44 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> the events are async so you can't count on the next event returned to
> be a reply to one you sent.
> you need to use the api command and uuid_kill to do the hangup
>
> char cmd[128];
> snprintf(cmd, sizeof(cmd), "api uuid_kill %s\n\n", some_uuid);
>
> esl_send_recv(&handle, cmd);
>
> this function actually guarantees a reply
>
> On Wed, Aug 3, 2011 at 2:10 AM, dma <mays.david at gmail.com> wrote:
> > Hello All,
> >
> > I have some problem with receiving and handling ESL events.
> >
> > My code is like:
> >
> > void handle_eslswitch_event_plain(esl_handle_t *handle, esl_event_t
> *event)
> > {
> >        ... ...
> >        TRACE_DEBUG("ESL event [%d] %s: UUID: %s",
> >                event->event_id, fs_get_event_header(event, "Event-Name"),
> > uniqueid);
> >         print_event(event);
> >        ... ...
> > }
> >
> > void handle_event(esl_handle_t *handle, esl_event_t *last_event)
> > {
> >        ... ...
> >        if (!strcasecmp(type, "text/event-plain")) {
> >            handle_eslswitch_event_plain(handle, handle->last_ievent);
> >        }
> >        ... ...
> > }
> >
> > int main()
> > {
> >        ... ...
> >        //receive from event socket
> >        if (handle.last_event) handle_event(&handle, handle.last_event);
> >        ... ...
> >
> > Usually I receive correct events. But I occasionally receive incorrect
> > event. See below the 1st is the normal event and the 2nd is the wrong
> event:
> >
> > <2011-07-29 14:25:29> [DEBUG] ESL event [8] CHANNEL_HANGUP_COMPLETE:
> UUID:
> > f320e4a6-db23-46d0-8d89-9957eccbd4c9
> > <2011-07-29 14:25:29> [DEBUG] RECV EVENT
> > Event-Name: CHANNEL_HANGUP_COMPLETE
> > Core-UUID: 26b77cba-8fdd-486d-90ec-6844bca58c72
> > FreeSWITCH-Hostname: fs01
> > FreeSWITCH-IPv4: 10.1.1.46
> > FreeSWITCH-IPv6: ::1
> > Event-Date-Local: 2011-07-29 14:25:29
> >
> > However, usually after failure in executing "hangup" and getting "-ERR
> ..."
> > in the event->last_sr_reply, I have wrong event, but not always (an -ERR
> > returned from executing "hangup" doesn't always result in a wrong event).
> > Here it is:
> >
> > <2011-07-29 14:25:29> [NOTICE] -ERR invalid session id
> > [f320e4a6-db23-46d0-8d89-9957eccbd4c9]
> > <2011-07-28 15:55:23> [DEBUG] ESL event [0] : UUID:
> > <2011-07-28 15:55:23> [DEBUG] RECV EVENT
> > Content-Length: 6485
> > Content-Type: text/event-plain
> >
> > Event-Name: CHANNEL_HANGUP
> > Core-UUID: 26b77cba-8fdd-486d-90ec-6844bca58c72
> > FreeSWITCH-Hostname: fs01
> > FreeSWITCH-IPv4: 10.1.1.46
> > FreeSWITCH-IPv6: %3A%3A1
> > Event-Date-Local: 2011-07-28%2015%3A55%3A23
> >
> > In the above case, both Event ID and UUID are invalid.
> >
> > I think i need more information regarding how to use the data elements in
> > ESL event, especially how to use the following:
> >
> >    char last_reply[1024];
> >    /*! Las command reply when called with esl_send_recv */
> >    char last_sr_reply[1024];
> >    /*! Last event received. Only populated when **save_event is NULL */
> >    esl_event_t *last_event;
> >    /*! Last event received when called by esl_send_recv */
> >    esl_event_t *last_sr_event;
> >    /*! This will hold already processed events queued by esl_recv_event
> */
> >    esl_event_t *race_event;
> >    /*! Events that have content-type == text/plain and a body */
> >    esl_event_t *last_ievent;
> >
> > My questions, what event should I check if I am interested in only
> > channel/call related events, and how? Do I check handle.last_event, or
> > handle.last_ievent, or what?
> >
> > Else, when checking command reply, do I check last_sr_reply, or
> > last_sr_event->body?
> >
> > Please kindly advice.
> >
> > Thanks,
> > D.Ma
> >
> >
> >
> > --
> > View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Receiving-junk-ESL-events-tp6647776p6647776.html
> > Sent from the freeswitch-users mailing list archive at Nabble.com.
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110804/04b17aa2/attachment.html 

From dujinfang at gmail.com  Thu Aug  4 13:01:45 2011
From: dujinfang at gmail.com (Seven Du)
Date: Thu, 4 Aug 2011 17:01:45 +0800
Subject: [Freeswitch-users] Question on caller id
In-Reply-To: <6D2A7E4AD25D43AE94995EAC9B45BD86@gmail.com>
References: <6D2A7E4AD25D43AE94995EAC9B45BD86@gmail.com>
Message-ID: <69CEDDCB6CA64348A52994F29A24C09F@gmail.com>

I made a new trace against git HEAD and reported a jira: 

http://jira.freeswitch.org/browse/FS-3483
-- 
Seven Du
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj: http://www.freeswitch.org.cn
Sent with Sparrow (http://www.sparrowmailapp.com)
On Thursday, August 4, 2011 at 3:29 PM, Seven Du wrote: 
> Hi,
> 
> Just found a weird problem. I originate user/1001 &echo, and the caller id in ESL (Caller-Caller-ID-Number) shows "internal/sip:1001 at 192.168.7.7 (mailto:1001 at 192.168.7.7):41886;rinstance=1b34eba11b046e53" while I expect "1001".
> 
> How can I get 1001 as caller id?
> 
> SIP trace:
> 
> http://pastebin.freeswitch.org/16970
> 
> 
> Registrations:
> =================================================================
> Call-ID: NzhhYTExOWRiMjY0ZGY3Y2MxNDg3ZjJjMTA4ZGZlNTM.
> User: 1001 at 192.168.7.7 (mailto:1001 at 192.168.7.7)
> Contact: "user" <sip:1001 at 192.168.7.7 (mailto:1001 at 192.168.7.7):41886;rinstance=1b34eba11b046e53>
> Agent: Bria 3 release 3.1 stamp 58312
> Status:  Registered(UDP)(unknown) EXP(2011-08-04 16:01:38) EXPSECS(2324)
> Host:  seven-air.local
> IP:  192.168.7.7
> Port:  41886
> Auth-User: 1001
> Auth-Realm:  192.168.7.7
> MWI-Account: 1001 at 192.168.7.7 (mailto:1001 at 192.168.7.7)
> 
> Total items returned: 1
> =================================================================
> 
> Thanks.
> 
> -- 
> Seven Du
> About: http://about.me/dujinfang
> Blog: http://www.dujinfang.com
> Proj: http://www.freeswitch.org.cn
> Sent with Sparrow
> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110804/594cbd15/attachment-0001.html 

From ijurado at econcept.es  Thu Aug  4 13:03:54 2011
From: ijurado at econcept.es (Isaac Jurado)
Date: Thu, 4 Aug 2011 11:03:54 +0200
Subject: [Freeswitch-users] Hangup hooks on B legs
In-Reply-To: <CALNZuwkszOa6mhUA7UuL7NM9ZHDKSL7NcLxA2p9G0MP6-h_GQQ@mail.gmail.com>
References: <CABj_V-mOmm05ycYJZRPaeePeVzB4LWdTfzjNXwM3O=Kjmrn5tQ@mail.gmail.com>
	<CALNZuwm7Zh1FZpeavRECM6VjPUERsJODi=gRxWsXboR4hKEiLg@mail.gmail.com>
	<CABj_V-mubvz2cpCY1=ynLrAkDv088ZVURhBJ+1BiYe4XWhUyTA@mail.gmail.com>
	<CALNZuwkCW=ZiC9b8g9UCAkqFvFfWDyA_BywGLa_nJtvXv-2Y_Q@mail.gmail.com>
	<CABj_V-=uB_X77ZUrcYDU_Rx0yJLLwhzMAU3xtfoiOy=PeEwOdA@mail.gmail.com>
	<CALNZuwkszOa6mhUA7UuL7NM9ZHDKSL7NcLxA2p9G0MP6-h_GQQ@mail.gmail.com>
Message-ID: <CABj_V-mfm79xO6imjmdvMidOqkiwYNxLPTCjJNXcscWTdB0B2g@mail.gmail.com>

On Wed, Aug 3, 2011 at 7:56 PM, Avi Marcus <avi at avimarcus.net> wrote:
>
> If the channel is hung up, then it's hung up.. that's what the hangup
> hook is, afaik.
> If you look at the wiki page for this:
> http://wiki.freeswitch.org/wiki/Variable_session_in_hangup_hook
> it only says that you can "access" the variable, not that the channel
> still exists or that you can set them.

Ok.  I still have some problems with this.  My dialplan contains the
following:

<action application="export" data="session_in_hangup_hook=true"/>
<action application="export" data="api_hangup_hook=luarun
transfer_check.lua ${uuid}"/>

The transfer_check.lua script is the following:

session = freeswitch.Session(argv[1])
local cause = session:hangupCause()

if cause == "ATTENDED_TRANSFER" or cause == "BLIND_TRANSFER"
then
    api = freeswitch.API()
    api.execute("log",  "NOTICE Transfer detected, billsec is " ..
session:getVariable("billsec"))
end

The problem is that, as already discussed, the channel no longer exists
so the UUID is not referring to anything.

Basically, my question is: How do you obtain the session object in a
hangup hook?

Cheers.

-- 
Isaac Jurado
Internet Busines Solutions eConcept


From nicevoip at googlemail.com  Thu Aug  4 13:48:00 2011
From: nicevoip at googlemail.com (Nice Voip)
Date: Thu, 4 Aug 2011 11:48:00 +0200
Subject: [Freeswitch-users] Problem with XML CDR
Message-ID: <CA+7EGbmk6WhgC8tCfFtSqkrQDosd4j87x7x1Bei0J+Km=HWHpw@mail.gmail.com>

Dear All,

This problem is very hard to reproduce and i really don't know when it would
happen, i don't have sip log or other traces but only the CDR file and its
looks like this:


<caller_id_name>TELESAT TRAJKOVI?
      <ani>+3xxxxxxx</ani>          ? MILE</caller_id_name>
      <aniii></aniii>

and sometimes

<oritinate_early_media> is writted <originate_uarly_media> (insead of early
its "uarly")

i noted down this issue on: FreeSWITCH Version 1.0.head (git-1d3417a
2011-06-07 17-35-49 -0400)

Thanks.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110804/c5eab0ac/attachment.html 

From avi at avimarcus.net  Thu Aug  4 13:56:50 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 4 Aug 2011 12:56:50 +0300
Subject: [Freeswitch-users] Problem with XML CDR
In-Reply-To: <CA+7EGbmk6WhgC8tCfFtSqkrQDosd4j87x7x1Bei0J+Km=HWHpw@mail.gmail.com>
References: <CA+7EGbmk6WhgC8tCfFtSqkrQDosd4j87x7x1Bei0J+Km=HWHpw@mail.gmail.com>
Message-ID: <CALNZuwnK9g83RoGF4CnM0Ou8qAabYPB-nWQ7_uAuYL_Cr=1zLw@mail.gmail.com>

The usual response is.. can you update from your nearly two month old
version and reproduce the same issue?
-Avi


On Thu, Aug 4, 2011 at 12:48 PM, Nice Voip <nicevoip at googlemail.com> wrote:

> Dear All,
>
> This problem is very hard to reproduce and i really don't know when it
> would happen, i don't have sip log or other traces but only the CDR file and
> its looks like this:
>
>
> <caller_id_name>TELESAT TRAJKOVI?
>       <ani>+3xxxxxxx</ani>          ? MILE</caller_id_name>
>       <aniii></aniii>
>
> and sometimes
>
> <oritinate_early_media> is writted <originate_uarly_media> (insead of early
> its "uarly")
>
> i noted down this issue on: FreeSWITCH Version 1.0.head (git-1d3417a
> 2011-06-07 17-35-49 -0400)
>
> Thanks.
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110804/444f26ce/attachment.html 

From nicevoip at googlemail.com  Thu Aug  4 14:03:18 2011
From: nicevoip at googlemail.com (Nice Voip)
Date: Thu, 4 Aug 2011 12:03:18 +0200
Subject: [Freeswitch-users] Problem with XML CDR
In-Reply-To: <CALNZuwnK9g83RoGF4CnM0Ou8qAabYPB-nWQ7_uAuYL_Cr=1zLw@mail.gmail.com>
References: <CA+7EGbmk6WhgC8tCfFtSqkrQDosd4j87x7x1Bei0J+Km=HWHpw@mail.gmail.com>
	<CALNZuwnK9g83RoGF4CnM0Ou8qAabYPB-nWQ7_uAuYL_Cr=1zLw@mail.gmail.com>
Message-ID: <CA+7EGb=nj_XKFkPMijtD_MSj9XurFG0THDZVq+mr+AExcqrEaA@mail.gmail.com>

Hmm i've latest version too, but there is not much traffic, and also on this
two month older version its rarely happen, in any case i'll try to move to
latest version and will try to reproduce.... but when it will be reproduced
my version will not be latest anymore :)



On Thu, Aug 4, 2011 at 11:56 AM, Avi Marcus <avi at avimarcus.net> wrote:

> The usual response is.. can you update from your nearly two month old
> version and reproduce the same issue?
> -Avi
>
>
> On Thu, Aug 4, 2011 at 12:48 PM, Nice Voip <nicevoip at googlemail.com>wrote:
>
>> Dear All,
>>
>> This problem is very hard to reproduce and i really don't know when it
>> would happen, i don't have sip log or other traces but only the CDR file and
>> its looks like this:
>>
>>
>> <caller_id_name>TELESAT TRAJKOVI?
>>       <ani>+3xxxxxxx</ani>          ? MILE</caller_id_name>
>>       <aniii></aniii>
>>
>> and sometimes
>>
>> <oritinate_early_media> is writted <originate_uarly_media> (insead of
>> early its "uarly")
>>
>> i noted down this issue on: FreeSWITCH Version 1.0.head (git-1d3417a
>> 2011-06-07 17-35-49 -0400)
>>
>> Thanks.
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110804/08c90e3c/attachment.html 

From dhairya.blogs at gmail.com  Thu Aug  4 14:39:14 2011
From: dhairya.blogs at gmail.com (Dhairya Vora)
Date: Thu, 4 Aug 2011 16:09:14 +0530
Subject: [Freeswitch-users] freeswitch setup error at making in
	lang/cmu_us_kal ...
Message-ID: <CA+P_fE8tM-iTg-Y0_x_bP+-iNTZZvB6+tYY2z_==2X2qVJEgBg@mail.gmail.com>

I am making a fresh installation of freeswitch on a dedicated server (CentOS
5.6, 64 bit). While installing, the setup stops at "making in
lang/cmu_us_kal ..."

DETAILED:
making install mod_flite
making in ...
making in include ...
making in src ...
making in src/audio ...
making in src/utils ...
making in src/regex ...
making in src/hrg ...
making in src/stats ...
making in src/speech ...
making in src/lexicon ...
making in src/synth ...
making in src/wavesynth ...
making in src/cg ...
making in lang ...
making in lang/cmulex ...
making in lang/usenglish ...
making in lang/cmu_us_kal ...

Is this issue related to less RAM or virtual memory ? Or is it something
else?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110804/84cd5808/attachment.html 

From dhairya.blogs at gmail.com  Thu Aug  4 14:43:22 2011
From: dhairya.blogs at gmail.com (Dhairya Vora)
Date: Thu, 4 Aug 2011 16:13:22 +0530
Subject: [Freeswitch-users] freeswitch setup error at making in
	lang/cmu_us_kal ...
In-Reply-To: <CA+P_fE8tM-iTg-Y0_x_bP+-iNTZZvB6+tYY2z_==2X2qVJEgBg@mail.gmail.com>
References: <CA+P_fE8tM-iTg-Y0_x_bP+-iNTZZvB6+tYY2z_==2X2qVJEgBg@mail.gmail.com>
Message-ID: <CA+P_fE9-kc+5TY-PCe0yvss_0dDRBNYnpNvpHtPuooh0V+tnng@mail.gmail.com>

It does not proceed, does not give any error...

On Thu, Aug 4, 2011 at 4:09 PM, Dhairya Vora <dhairya.blogs at gmail.com>wrote:

> I am making a fresh installation of freeswitch on a dedicated server
> (CentOS 5.6, 64 bit). While installing, the setup stops at "making in
> lang/cmu_us_kal ..."
>
> DETAILED:
> making install mod_flite
> making in ...
> making in include ...
> making in src ...
> making in src/audio ...
> making in src/utils ...
> making in src/regex ...
> making in src/hrg ...
> making in src/stats ...
> making in src/speech ...
> making in src/lexicon ...
> making in src/synth ...
> making in src/wavesynth ...
> making in src/cg ...
> making in lang ...
> making in lang/cmulex ...
> making in lang/usenglish ...
> making in lang/cmu_us_kal ...
>
> Is this issue related to less RAM or virtual memory ? Or is it something
> else?
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110804/8c7cc6d9/attachment.html 

From rahulkrishna222 at gmail.com  Thu Aug  4 15:09:56 2011
From: rahulkrishna222 at gmail.com (rahulkrishna)
Date: Thu, 4 Aug 2011 04:09:56 -0700 (PDT)
Subject: [Freeswitch-users] Freeswitch Hangup Event Tracking
In-Reply-To: <CAAYf2LuJhRtq-x=p_Mn1FxyF++fMAJb3iuMbw=DT6r907Zk5CQ@mail.gmail.com>
References: <CAALn-hO8+t2P-=cvFSvfw-EZxje+KuejQeiM2r-Vx8pgxoYpvA@mail.gmail.com>
	<CAAYf2LuJhRtq-x=p_Mn1FxyF++fMAJb3iuMbw=DT6r907Zk5CQ@mail.gmail.com>
Message-ID: <1312456196179-6652289.post@n2.nabble.com>

Hi., can you please tell me that . how can i track CHANNEL_HANGUP Event . 
and in which dailplan i need to write . . please reply

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-Hangup-Event-Tracking-tp6648025p6652289.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From rahulkrishna222 at gmail.com  Thu Aug  4 15:18:23 2011
From: rahulkrishna222 at gmail.com (rahulkrishna)
Date: Thu, 4 Aug 2011 04:18:23 -0700 (PDT)
Subject: [Freeswitch-users] Freeswitch Hangup Event Tracking
In-Reply-To: <CAKzWOxUR9xLLK_njx7Rtrknxd=fOcKmVBnWERjFXySE3uSVjGQ@mail.gmail.com>
References: <CAALn-hO8+t2P-=cvFSvfw-EZxje+KuejQeiM2r-Vx8pgxoYpvA@mail.gmail.com>
	<CAKzWOxUR9xLLK_njx7Rtrknxd=fOcKmVBnWERjFXySE3uSVjGQ@mail.gmail.com>
Message-ID: <1312456703621-6652301.post@n2.nabble.com>

Hi thanks for your reply. 
What am actually need is when the user hangup the call .i want to keep alive
that channel because i want to execute a program that is done in mod_java..
and i don't know how to execute this application on api_hangup_hook event.
please reply..

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-Hangup-Event-Tracking-tp6648025p6652301.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From rahulkrishna222 at gmail.com  Thu Aug  4 15:46:19 2011
From: rahulkrishna222 at gmail.com (rahulkrishna)
Date: Thu, 4 Aug 2011 04:46:19 -0700 (PDT)
Subject: [Freeswitch-users] api_hangup_hook and hupall
In-Reply-To: <AANLkTinYafxtbPyjzNlix1aqjwE6lbjs0HWN5U9afTuo@mail.gmail.com>
References: <AANLkTinYafxtbPyjzNlix1aqjwE6lbjs0HWN5U9afTuo@mail.gmail.com>
Message-ID: <1312458379804-6652362.post@n2.nabble.com>

Hai can you please tell me how can run a mod_java application on call HangUp
using  api_hangup_hook .
Please give an example to.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/api-hangup-hook-and-hupall-tp5159879p6652362.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From steveayre at gmail.com  Thu Aug  4 17:05:30 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 4 Aug 2011 14:05:30 +0100
Subject: [Freeswitch-users] Hangup hooks on B legs
In-Reply-To: <CABj_V-mfm79xO6imjmdvMidOqkiwYNxLPTCjJNXcscWTdB0B2g@mail.gmail.com>
References: <CABj_V-mOmm05ycYJZRPaeePeVzB4LWdTfzjNXwM3O=Kjmrn5tQ@mail.gmail.com>
	<CALNZuwm7Zh1FZpeavRECM6VjPUERsJODi=gRxWsXboR4hKEiLg@mail.gmail.com>
	<CABj_V-mubvz2cpCY1=ynLrAkDv088ZVURhBJ+1BiYe4XWhUyTA@mail.gmail.com>
	<CALNZuwkCW=ZiC9b8g9UCAkqFvFfWDyA_BywGLa_nJtvXv-2Y_Q@mail.gmail.com>
	<CABj_V-=uB_X77ZUrcYDU_Rx0yJLLwhzMAU3xtfoiOy=PeEwOdA@mail.gmail.com>
	<CALNZuwkszOa6mhUA7UuL7NM9ZHDKSL7NcLxA2p9G0MP6-h_GQQ@mail.gmail.com>
	<CABj_V-mfm79xO6imjmdvMidOqkiwYNxLPTCjJNXcscWTdB0B2g@mail.gmail.com>
Message-ID: <CAFiqYunySoaNVp6Fb2Sg6YDLATLx7wh9yBW=RXA5Jiqyb_VyZA@mail.gmail.com>

Is this any use?

http://wiki.freeswitch.org/wiki/Mod_lua#Special_Case:_env_object

-Steve




On 4 August 2011 10:03, Isaac Jurado <ijurado at econcept.es> wrote:

> On Wed, Aug 3, 2011 at 7:56 PM, Avi Marcus <avi at avimarcus.net> wrote:
> >
> > If the channel is hung up, then it's hung up.. that's what the hangup
> > hook is, afaik.
> > If you look at the wiki page for this:
> > http://wiki.freeswitch.org/wiki/Variable_session_in_hangup_hook
> > it only says that you can "access" the variable, not that the channel
> > still exists or that you can set them.
>
> Ok.  I still have some problems with this.  My dialplan contains the
> following:
>
> <action application="export" data="session_in_hangup_hook=true"/>
> <action application="export" data="api_hangup_hook=luarun
> transfer_check.lua ${uuid}"/>
>
> The transfer_check.lua script is the following:
>
> session = freeswitch.Session(argv[1])
> local cause = session:hangupCause()
>
> if cause == "ATTENDED_TRANSFER" or cause == "BLIND_TRANSFER"
> then
>    api = freeswitch.API()
>    api.execute("log",  "NOTICE Transfer detected, billsec is " ..
> session:getVariable("billsec"))
> end
>
> The problem is that, as already discussed, the channel no longer exists
> so the UUID is not referring to anything.
>
> Basically, my question is: How do you obtain the session object in a
> hangup hook?
>
> Cheers.
>
> --
> Isaac Jurado
> Internet Busines Solutions eConcept
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110804/6d530742/attachment.html 

From ijurado at econcept.es  Thu Aug  4 17:18:43 2011
From: ijurado at econcept.es (Isaac Jurado)
Date: Thu, 4 Aug 2011 15:18:43 +0200
Subject: [Freeswitch-users] Hangup hooks on B legs
In-Reply-To: <CAFiqYunySoaNVp6Fb2Sg6YDLATLx7wh9yBW=RXA5Jiqyb_VyZA@mail.gmail.com>
References: <CABj_V-mOmm05ycYJZRPaeePeVzB4LWdTfzjNXwM3O=Kjmrn5tQ@mail.gmail.com>
	<CALNZuwm7Zh1FZpeavRECM6VjPUERsJODi=gRxWsXboR4hKEiLg@mail.gmail.com>
	<CABj_V-mubvz2cpCY1=ynLrAkDv088ZVURhBJ+1BiYe4XWhUyTA@mail.gmail.com>
	<CALNZuwkCW=ZiC9b8g9UCAkqFvFfWDyA_BywGLa_nJtvXv-2Y_Q@mail.gmail.com>
	<CABj_V-=uB_X77ZUrcYDU_Rx0yJLLwhzMAU3xtfoiOy=PeEwOdA@mail.gmail.com>
	<CALNZuwkszOa6mhUA7UuL7NM9ZHDKSL7NcLxA2p9G0MP6-h_GQQ@mail.gmail.com>
	<CABj_V-mfm79xO6imjmdvMidOqkiwYNxLPTCjJNXcscWTdB0B2g@mail.gmail.com>
	<CAFiqYunySoaNVp6Fb2Sg6YDLATLx7wh9yBW=RXA5Jiqyb_VyZA@mail.gmail.com>
Message-ID: <CABj_V-=-Mx3OmBeKfrJWjU_yF3aAgN9REjxmP7oR0hReQvzSSQ@mail.gmail.com>

On Thu, Aug 4, 2011 at 3:05 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
> Is this any use?
>
> http://wiki.freeswitch.org/wiki/Mod_lua#Special_Case:_env_object

Holly crap, I'm blind!  I've visited the mod_lua page at least fifty
times and I didn't see that.  I feel embarrassed.

Thank you so much.

-- 
Isaac Jurado
Internet Busines Solutions eConcept


From roger.castaldo at gmail.com  Thu Aug  4 17:26:01 2011
From: roger.castaldo at gmail.com (Roger Castaldo)
Date: Thu, 4 Aug 2011 09:26:01 -0400
Subject: [Freeswitch-users] Freeswitch Hangup Event Tracking
In-Reply-To: <1312456196179-6652289.post@n2.nabble.com>
References: <CAALn-hO8+t2P-=cvFSvfw-EZxje+KuejQeiM2r-Vx8pgxoYpvA@mail.gmail.com>
	<CAAYf2LuJhRtq-x=p_Mn1FxyF++fMAJb3iuMbw=DT6r907Zk5CQ@mail.gmail.com>
	<1312456196179-6652289.post@n2.nabble.com>
Message-ID: <CAAYf2LvJgJn49Nsrt7MXk0NGKgWxby7W5GaDoMBqxZWDS-CR9w@mail.gmail.com>

You need to enable mod_eventsocket and confgure a program to connect to the
socket that will then be available.  There are several available samples in
several languages.  All should be discussed within the wiki.

On Thu, Aug 4, 2011 at 7:09 AM, rahulkrishna <rahulkrishna222 at gmail.com>wrote:

> Hi., can you please tell me that . how can i track CHANNEL_HANGUP Event .
> and in which dailplan i need to write . . please reply
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-Hangup-Event-Tracking-tp6648025p6652289.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110804/2fdafc81/attachment.html 

From curriegrad2004 at gmail.com  Thu Aug  4 20:20:27 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Thu, 4 Aug 2011 09:20:27 -0700
Subject: [Freeswitch-users] freeswitch 1.0.7 required
In-Reply-To: <CA5F9A6E.1A0F0%krice@freeswitch.org>
References: <CA+P_fE_UgehuXmdtj8DOtoHYToRrUQg7tjpkCswe0-SgPbD10Q@mail.gmail.com>
	<CA5F9A6E.1A0F0%krice@freeswitch.org>
Message-ID: <CAMWiMy_61z8UhxL6wG+rzVQ9jdhhKdH9m9zUHtUKdQ_bghCG-w@mail.gmail.com>

1.0.7 was more like a joke release. Of course us seasoned FS users
know better ;)

On Wed, Aug 3, 2011 at 10:38 PM, Ken Rice <krice at freeswitch.org> wrote:
> 1.0.X , X is a snapshot of git at some point in time... Its a release
> version... The problem is people don?t test when the main developers say
> they are getting ready for a new release, so they test as best as they can
> an cut a release...
>
> 20 minutes later, a dozen bugs are reported... ?Because people expect
> everyone else to test... The way to fix this was decided that there will be
> OLD version (and by old I mean really old) and there will be head for the
> most part... That forces people to test... And 99.99% of the time if you do
> fine something broken ?make current? fixes the issue...
>
>
> And yes 1.0.head is what comes from git... Notice that git-HEXNUMBER...
> That?s the actual give hash version...
> Have a nice day
>
> K
>
>
> On 8/4/11 12:33 AM, "Dhairya Vora" <dhairya.blogs at gmail.com> wrote:
>
> Here (http://wiki.freeswitch.org/wiki/Download_FreeSWITCH) they say that in
> Git, 1.0.7 is not available.
>
> (FYI: here
> (http://lists.freeswitch.org/pipermail/freeswitch-users/2011-January/067446.html)
> they say that "git does not give odd number version." Really??)
>
> By the way, I installed using git and I thought that it is 1.0.6. Now it
> shows 1.0.head. just see the output.
> ****************************************************************************************************
> freeswitch at localhost.localdomain> version
>
> FreeSWITCH Version 1.0.head (git-4b1bb61 2011-08-01 15-43-07 -0500)
> ****************************************************************************************************
>
>
>
> On Thu, Aug 4, 2011 at 10:51 AM, Ken Rice <krice at freeswitch.org> wrote:
>
> Just get git head... That?s where you want to be anyway... Yes its stable...
> But as with any new deployment you should test it to make sure it meets your
> needs
>
>
>
>
> On 8/4/11 12:18 AM, "Dhairya Vora" <dhairya.blogs at gmail.com
> <http://dhairya.blogs at gmail.com> > wrote:
>
> I am installing freeswitch. I read that latest version is 1.0.7 but in git
> it is not available. They say that it is at http://latest.freeswitch.org/
> but I am unable to open that url. Either server is down, or url is changed.
> Any other location to download freeswitch 1.0.7 (or the latest version) ?
>
> ________________________________
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> <http://FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> ________________________________
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From marketing at cluecon.com  Thu Aug  4 21:33:40 2011
From: marketing at cluecon.com (Michael Collins)
Date: Thu, 4 Aug 2011 10:33:40 -0700
Subject: [Freeswitch-users] Last Minute ClueCon Items: Hotel, Parking,
	Directions, Speakers
Message-ID: <CAKzWOxVkpSEOH55cW9OLzvAzVPCUmwLrt=0LRgbG=D6mBtekjg@mail.gmail.com>

ClueCon is only a few days away! Here are a few items for you to keep in
mind:

*HOTEL*

The Talbott Hotel has sold out of the $148/night rooms. If you book by
Sunday you can still get a reasonable rate of $169/night. Thank you to Brian
West for going above and beyond in order to give us another alternative.

*PARKING*

Parking in downtown Chicago is both expensive and challenging. Dr. Moshe
Yudkowsky, one of our regular presenters, is quite familiar with the area
and offers us this the follow information:

The three methods I use to park are:

(1) Don't park downtown. Several CTA stops have nearby parking - for
example, not far from my office there's a CTA-owned parking structure that
charges $1/day to park right at the El train stop. Park your car outside
downtown, take the train, and avoid traffic.

For those that like to cycle, the bike path along the lake is excellent and
convenient.

(2) Parkwhiz.com <http://parkwhiz.com/> is a service that lets you make
reservations online and pay in advance. Here's a search on the hotel's
address for August 10th: <
http://www.parkwhiz.com/search/?destination=20+EAST+CHESTNUT+ST%2C+chicago&form2=1&start_date=08%2F10%2F2011&start_time=08%3A30&end_time=17%3A00&end_date=08%2F10%2F2011
>

(3) Faspark.com <http://faspark.com/> is a startup that lets you search for
free or paid parking near your destination. As you might imagine, a few
quick tests showed very little chance of finding a spot in mid-day. I have
the Android app and it seems to work tolerably well.

If you are inclined to do the "park and ride" then this CTA web site will be
most helpful: http://www.transitchicago.com/parking/

*DIRECTIONS*

For those flying into Midway (MDW) or O'Hare (ORD) you have a few choices
for getting from the airport to the hotel (and back again). You can take a
taxi or a shuttle, but those are about $30 each way (give or take) which is
not a great value. Personally, I've taken the train the past three times
I've flown in to Chicago and found it to be very economical (approx $3 each
way). Additionally, it's kind of interesting to check out the scenery as you
ride into town.

>From ORD -
Take Blue line to Jackson station, free transfer to Red line
Take Red line toward Howard, exit at Chicago station
Exit station, take State St. north
Sofitel: turn right on Chestnut St.
Talbott: turn right on Delware St.

>From MDW -
Take Orange line to State station, free transfer to Red line
Take Red line toward Howard, exit at Chicago station
Exit station, take State St. north
Sofitel: turn right on Chestnut St.
Talbott: turn right on Delware St.

Sofitel is located at 20 E Chestnut St
The Talbott is located at 20 E Delaware St

CTA train maps are available online and for download here:
http://www.transitchicago.com/maps/. If you have any questions please feel
free to email us at this address and we'll be happy to help.

*SPEAKERS*

If you are speaking this year then please review cluecon.com/schedule - this
is the most up-to-date information on who is speaking and when. Also, please
email us your presentation files so that we can put them on the presentation
laptop. We support PDF, PowerPoint, and Keynote. If you have special
requirements then let us know and we'll assist. We thank all those who are
generously donating their time and energy in giving presentations this year.

We are eagerly awaiting ClueCon 2011! Looking forward to seeing you all next
week.
Cheers!
-- 
Michael S Collins
ClueCon Team
http://www.cluecon.com
877-7-4ACLUE
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110804/19339034/attachment-0001.html 

From micha.knop at googlemail.com  Thu Aug  4 12:42:41 2011
From: micha.knop at googlemail.com (Micha Knop)
Date: Thu, 4 Aug 2011 10:42:41 +0200
Subject: [Freeswitch-users] DNS SRV records
Message-ID: <CAB3X6Qi0YbYz7RWpSxdbrMypH8rav4+JdCpb+8MheqmPUMNhqw@mail.gmail.com>

Hello!

For a better usability (I don't want to expect my users to deal with
IP addresses) and for a better manageability (I don't want to inform
my users about IP address changes) I'd like to set DNS SRV records.
But which port should I announce? The internal or the external port?
What about soft phones on notebooks which are sometimes on my internal
network and sometimes outside? Do they have to have two profiles?  Or
do I have to work with bind views?

Thanks for any hint!

Cheers,

Micha


From 7eicher at gmx.de  Thu Aug  4 14:14:39 2011
From: 7eicher at gmx.de (Markus Siebeneicher)
Date: Thu, 04 Aug 2011 10:14:39 +0000
Subject: [Freeswitch-users] Does FreeSWITCH supports Googles VP8 (WebM) Codec
Message-ID: <4E3A710F.3000003@gmx.de>

Hello folks from FreeSWITCH,

i am using FreeSWITCH with the conference system BigBlueButton, made by 
people from canada.

It works pretty fine, so far. Due to the need that we would like to 
support more than 500 users in a single video/audio session, we are in 
interest to use the best codecs.

So i listen to VP8 from Google. Does FreeSWITCH support it anyway? I 
couldn't find something in the FAQ's. I listen something about ffmeg, is 
it in this abstraction module?

Thanks for a short reply and thanks for the good work!

Kind regards,

Markus


From micha.knop at googlemail.com  Thu Aug  4 19:38:38 2011
From: micha.knop at googlemail.com (micha)
Date: Thu, 4 Aug 2011 08:38:38 -0700 (PDT)
Subject: [Freeswitch-users] ACL: inside out
Message-ID: <1312472318582-6653218.post@n2.nabble.com>

Hello!

How do I define what is inside and what is outside my network? My FreeSWITCH
server and my devices that reside on my internal network have public IP
addresses but in separate subnets.

I tried the following in acl.conf.xml:

   <list name="lan" default="deny">
      <node type="allow" cidr="XXX.XXX.XXX.0/22"/>
    </list>
	
But my internal clients are still listed as external registrations.


Where is my error?

Cheers,

Micha

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/ACL-inside-out-tp6653218p6653218.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Thu Aug  4 22:54:49 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 4 Aug 2011 11:54:49 -0700
Subject: [Freeswitch-users] api_hangup_hook and hupall
In-Reply-To: <1312458379804-6652362.post@n2.nabble.com>
References: <AANLkTinYafxtbPyjzNlix1aqjwE6lbjs0HWN5U9afTuo@mail.gmail.com>
	<1312458379804-6652362.post@n2.nabble.com>
Message-ID: <CAKzWOxWhAVhGhaGsCJQiCXO=M7P_W_nZh2cT6nBtDiZouMO+tg@mail.gmail.com>

Looking into mod_java.c it seems that it implements a "java" dialplan
application but not a "javarun" API like the other language modules. You
might want to contact the mod_java author and ask for advice:
damjandotjov at gmail.com

Or you could use Lua for your hangup hook as it is lighter and faster than
any of the embedded languages.

-MC

On Thu, Aug 4, 2011 at 4:46 AM, rahulkrishna <rahulkrishna222 at gmail.com>wrote:

> Hai can you please tell me how can run a mod_java application on call
> HangUp
> using  api_hangup_hook .
> Please give an example to.
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/api-hangup-hook-and-hupall-tp5159879p6652362.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110804/92dbebfc/attachment.html 

From michael.ricordeau at gmail.com  Thu Aug  4 23:24:58 2011
From: michael.ricordeau at gmail.com (Michael Ricordeau)
Date: Thu, 4 Aug 2011 21:24:58 +0200
Subject: [Freeswitch-users] outboundsocket mode,
	bridge and bind_digit_action doesn't work
Message-ID: <20110804212458.4462c72d@gmail.com>

Hi,

I think I'm doing something wrong but I don't find a way to fix my problem :

when executing bind_digit_action in an outbound socket and doing a bridge,
digit action binding is not executed (no log in Freeswitch)

when doing same thing in XML dialplan it works (I found log in Freeswitch).

I have checked logs and in both cases, digit realm is set :
Digit parser DPTOOLS: binding 00/test/0 callback: 0xb6ba1b00 data: 0x82eafe8



For outbound socket, here what I'm doing with netcat :


nc -l -v 8084


and the commands I passed to outbound socket :


connect

divert_events on

sendmsg
call-command: execute
execute-app-name: answer

sendmsg
call-command: execute
execute-app-name: bind_digit_action
execute-app-arg: test,00,exec:log,NOTICE TEST

sendmsg
call-command: execute
execute-app-name: bridge
execute-app-arg: user/1000




With XML Dialplan below, same binding/bridge works !!!!! :
    
    <extension name="test">
        <condition field="destination_number" expression="^0000$">
            <action application="answer"/>
            <action application="bind_digit_action" data="test,00,exec:log,NOTICE TEST"/>
            <action application="bridge" data="user/1000"/>
        </condition>
    </extension>



Expected result is when A leg presses "00", a log notice "TEST" is printed in the logger but only works with XML dialplan.
In outbound socket mode, I can hear the digits pressed on B leg but binding is not executed (no log notice "TEST")


Thanks


Micha?l











From steveayre at gmail.com  Fri Aug  5 01:39:52 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 4 Aug 2011 22:39:52 +0100
Subject: [Freeswitch-users] outboundsocket mode,
	bridge and bind_digit_action doesn't work
In-Reply-To: <20110804212458.4462c72d@gmail.com>
References: <20110804212458.4462c72d@gmail.com>
Message-ID: <86A3A865-CD9E-41F5-9BC6-E7BF4892BCE7@gmail.com>

Try using digit_action_set_realm to set the current realm to test before the bridge:

> sendmsg
> call-command: execute
> execute-app-name: digit_action_realm_set
> execute-app-arg: test

Steve on iPhone


On 4 Aug 2011, at 20:24, Michael Ricordeau <michael.ricordeau at gmail.com> wrote:

> Hi,
> 
> I think I'm doing something wrong but I don't find a way to fix my problem :
> 
> when executing bind_digit_action in an outbound socket and doing a bridge,
> digit action binding is not executed (no log in Freeswitch)
> 
> when doing same thing in XML dialplan it works (I found log in Freeswitch).
> 
> I have checked logs and in both cases, digit realm is set :
> Digit parser DPTOOLS: binding 00/test/0 callback: 0xb6ba1b00 data: 0x82eafe8
> 
> 
> 
> For outbound socket, here what I'm doing with netcat :
> 
> 
> nc -l -v 8084
> 
> 
> and the commands I passed to outbound socket :
> 
> 
> connect
> 
> divert_events on
> 
> sendmsg
> call-command: execute
> execute-app-name: answer
> 
> sendmsg
> call-command: execute
> execute-app-name: bind_digit_action
> execute-app-arg: test,00,exec:log,NOTICE TEST
> 
> sendmsg
> call-command: execute
> execute-app-name: bridge
> execute-app-arg: user/1000
> 
> 
> 
> 
> With XML Dialplan below, same binding/bridge works !!!!! :
> 
>    <extension name="test">
>        <condition field="destination_number" expression="^0000$">
>            <action application="answer"/>
>            <action application="bind_digit_action" data="test,00,exec:log,NOTICE TEST"/>
>            <action application="bridge" data="user/1000"/>
>        </condition>
>    </extension>
> 
> 
> 
> Expected result is when A leg presses "00", a log notice "TEST" is printed in the logger but only works with XML dialplan.
> In outbound socket mode, I can hear the digits pressed on B leg but binding is not executed (no log notice "TEST")
> 
> 
> Thanks
> 
> 
> Micha?l
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110804/845826ba/attachment.html 

From michael.ricordeau at gmail.com  Fri Aug  5 02:30:37 2011
From: michael.ricordeau at gmail.com (Michael Ricordeau)
Date: Fri, 5 Aug 2011 00:30:37 +0200
Subject: [Freeswitch-users] outboundsocket mode,
 bridge and bind_digit_action doesn't work
In-Reply-To: <86A3A865-CD9E-41F5-9BC6-E7BF4892BCE7@gmail.com>
References: <20110804212458.4462c72d@gmail.com>
	<86A3A865-CD9E-41F5-9BC6-E7BF4892BCE7@gmail.com>
Message-ID: <20110805003037.1ea31502@gmail.com>

Just tested with digit_action_set_realm and doesnt work.

If I set bind_digit_action and presses '00' before doing the bridge it works.

During the bridge, doesn't work and after the bridge works again.


Here is a pastebin : http://pastebin.freeswitch.org/16989


I found some difference in FS logs
 
For outbound eventsocket, we have :
2011-08-05 01:26:07.403250 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF 0:800
2011-08-05 01:26:07.403250 [DEBUG] switch_ivr_bridge.c:391 Send signal sofia/internal/sip:1001 at 81.220.86.183:59445 [BREAK]
2011-08-05 01:26:07.423252 [DEBUG] switch_rtp.c:2328 Send start packet for [0] ts=3214267546 dur=160/160/800 seq=11501
2011-08-05 01:26:07.463253 [DEBUG] switch_rtp.c:2264 Send middle packet for [0] ts=3214267546 dur=320/320/800 seq=11502
2011-08-05 01:26:07.503242 [DEBUG] switch_rtp.c:2264 Send middle packet for [0] ts=3214267546 dur=480/480/800 seq=11503
2011-08-05 01:26:07.543244 [DEBUG] switch_rtp.c:2264 Send middle packet for [0] ts=3214267546 dur=640/640/800 seq=11504
2011-08-05 01:26:07.583245 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF 0:800
 
 
And for XML Dialplan :
2011-08-05 01:26:51.053768 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF 0:800
2011-08-05 01:26:51.553793 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF 0:800
2011-08-05 01:26:51.553793 [DEBUG] mod_dptools.c:151 sofia/internal/1000 at 46.102.242.62 Digit match binding [exec:log][NOTICE TEST]



Le Thu, 4 Aug 2011 22:39:52 +0100,
Steven Ayre <steveayre at gmail.com> a ?crit :

> Try using digit_action_set_realm to set the current realm to test before the bridge:
> 
> > sendmsg
> > call-command: execute
> > execute-app-name: digit_action_realm_set
> > execute-app-arg: test
> 
> Steve on iPhone
> 
> 
> On 4 Aug 2011, at 20:24, Michael Ricordeau <michael.ricordeau at gmail.com> wrote:
> 
> > Hi,
> > 
> > I think I'm doing something wrong but I don't find a way to fix my problem :
> > 
> > when executing bind_digit_action in an outbound socket and doing a bridge,
> > digit action binding is not executed (no log in Freeswitch)
> > 
> > when doing same thing in XML dialplan it works (I found log in Freeswitch).
> > 
> > I have checked logs and in both cases, digit realm is set :
> > Digit parser DPTOOLS: binding 00/test/0 callback: 0xb6ba1b00 data: 0x82eafe8
> > 
> > 
> > 
> > For outbound socket, here what I'm doing with netcat :
> > 
> > 
> > nc -l -v 8084
> > 
> > 
> > and the commands I passed to outbound socket :
> > 
> > 
> > connect
> > 
> > divert_events on
> > 
> > sendmsg
> > call-command: execute
> > execute-app-name: answer
> > 
> > sendmsg
> > call-command: execute
> > execute-app-name: bind_digit_action
> > execute-app-arg: test,00,exec:log,NOTICE TEST
> > 
> > sendmsg
> > call-command: execute
> > execute-app-name: bridge
> > execute-app-arg: user/1000
> > 
> > 
> > 
> > 
> > With XML Dialplan below, same binding/bridge works !!!!! :
> > 
> >    <extension name="test">
> >        <condition field="destination_number" expression="^0000$">
> >            <action application="answer"/>
> >            <action application="bind_digit_action" data="test,00,exec:log,NOTICE TEST"/>
> >            <action application="bridge" data="user/1000"/>
> >        </condition>
> >    </extension>
> > 
> > 
> > 
> > Expected result is when A leg presses "00", a log notice "TEST" is printed in the logger but only works with XML dialplan.
> > In outbound socket mode, I can hear the digits pressed on B leg but binding is not executed (no log notice "TEST")
> > 
> > 
> > Thanks
> > 
> > 
> > Micha?l
> > 
> > 
> > 
> > 
> > 
> > 
> > 
> > 
> > 
> > 
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org


From dujinfang at gmail.com  Fri Aug  5 03:57:24 2011
From: dujinfang at gmail.com (Seven Du)
Date: Fri, 5 Aug 2011 07:57:24 +0800
Subject: [Freeswitch-users] Does FreeSWITCH supports Googles VP8 (WebM)
 Codec
In-Reply-To: <4E3A710F.3000003@gmx.de>
References: <4E3A710F.3000003@gmx.de>
Message-ID: <26A4B29329D049A4AA15FA1B375BC7CA@gmail.com>

All video codecs is passthrough currently in FS, so it doesn't matter of performance in FS even it might matter in other parts of BBB.

I guess you could easily write a passthrough module for VP8 just like mod_h26x.
-- 
Seven Du
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj: http://www.freeswitch.org.cn
Sent with Sparrow (http://www.sparrowmailapp.com)

On Thursday, August 4, 2011 at 6:14 PM, Markus Siebeneicher wrote: 
> Hello folks from FreeSWITCH,
> 
> i am using FreeSWITCH with the conference system BigBlueButton, made by 
> people from canada.
> 
> It works pretty fine, so far. Due to the need that we would like to 
> support more than 500 users in a single video/audio session, we are in 
> interest to use the best codecs.
> 
> So i listen to VP8 from Google. Does FreeSWITCH support it anyway? I 
> couldn't find something in the FAQ's. I listen something about ffmeg, is 
> it in this abstraction module?
> 
> Thanks for a short reply and thanks for the good work!
> 
> Kind regards,
> 
> Markus
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org)
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110805/84f2c428/attachment.html 

From msc at freeswitch.org  Fri Aug  5 04:13:40 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 4 Aug 2011 17:13:40 -0700
Subject: [Freeswitch-users] Hangup hooks on B legs
In-Reply-To: <CABj_V-=-Mx3OmBeKfrJWjU_yF3aAgN9REjxmP7oR0hReQvzSSQ@mail.gmail.com>
References: <CABj_V-mOmm05ycYJZRPaeePeVzB4LWdTfzjNXwM3O=Kjmrn5tQ@mail.gmail.com>
	<CALNZuwm7Zh1FZpeavRECM6VjPUERsJODi=gRxWsXboR4hKEiLg@mail.gmail.com>
	<CABj_V-mubvz2cpCY1=ynLrAkDv088ZVURhBJ+1BiYe4XWhUyTA@mail.gmail.com>
	<CALNZuwkCW=ZiC9b8g9UCAkqFvFfWDyA_BywGLa_nJtvXv-2Y_Q@mail.gmail.com>
	<CABj_V-=uB_X77ZUrcYDU_Rx0yJLLwhzMAU3xtfoiOy=PeEwOdA@mail.gmail.com>
	<CALNZuwkszOa6mhUA7UuL7NM9ZHDKSL7NcLxA2p9G0MP6-h_GQQ@mail.gmail.com>
	<CABj_V-mfm79xO6imjmdvMidOqkiwYNxLPTCjJNXcscWTdB0B2g@mail.gmail.com>
	<CAFiqYunySoaNVp6Fb2Sg6YDLATLx7wh9yBW=RXA5Jiqyb_VyZA@mail.gmail.com>
	<CABj_V-=-Mx3OmBeKfrJWjU_yF3aAgN9REjxmP7oR0hReQvzSSQ@mail.gmail.com>
Message-ID: <CAKzWOxVwhUzjn32J4TRkvAtxhrOObbgoPb46NYk3D-4mra4dBg@mail.gmail.com>

On Thu, Aug 4, 2011 at 6:18 AM, Isaac Jurado <ijurado at econcept.es> wrote:

> On Thu, Aug 4, 2011 at 3:05 PM, Steven Ayre <steveayre at gmail.com> wrote:
> >
> > Is this any use?
> >
> > http://wiki.freeswitch.org/wiki/Mod_lua#Special_Case:_env_object
>
> Holly crap, I'm blind!  I've visited the mod_lua page at least fifty
> times and I didn't see that.  I feel embarrassed.
>
>
To help connect the dots I added some links to/from that Lua wiki page and
channel_in_hangup_hook and also api_hangup_hook and reporting_hangup_hook.

-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110804/d60dc1f9/attachment.html 

From msc at freeswitch.org  Fri Aug  5 04:14:45 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 4 Aug 2011 17:14:45 -0700
Subject: [Freeswitch-users] Hangup hooks on B legs
In-Reply-To: <CAKzWOxVwhUzjn32J4TRkvAtxhrOObbgoPb46NYk3D-4mra4dBg@mail.gmail.com>
References: <CABj_V-mOmm05ycYJZRPaeePeVzB4LWdTfzjNXwM3O=Kjmrn5tQ@mail.gmail.com>
	<CALNZuwm7Zh1FZpeavRECM6VjPUERsJODi=gRxWsXboR4hKEiLg@mail.gmail.com>
	<CABj_V-mubvz2cpCY1=ynLrAkDv088ZVURhBJ+1BiYe4XWhUyTA@mail.gmail.com>
	<CALNZuwkCW=ZiC9b8g9UCAkqFvFfWDyA_BywGLa_nJtvXv-2Y_Q@mail.gmail.com>
	<CABj_V-=uB_X77ZUrcYDU_Rx0yJLLwhzMAU3xtfoiOy=PeEwOdA@mail.gmail.com>
	<CALNZuwkszOa6mhUA7UuL7NM9ZHDKSL7NcLxA2p9G0MP6-h_GQQ@mail.gmail.com>
	<CABj_V-mfm79xO6imjmdvMidOqkiwYNxLPTCjJNXcscWTdB0B2g@mail.gmail.com>
	<CAFiqYunySoaNVp6Fb2Sg6YDLATLx7wh9yBW=RXA5Jiqyb_VyZA@mail.gmail.com>
	<CABj_V-=-Mx3OmBeKfrJWjU_yF3aAgN9REjxmP7oR0hReQvzSSQ@mail.gmail.com>
	<CAKzWOxVwhUzjn32J4TRkvAtxhrOObbgoPb46NYk3D-4mra4dBg@mail.gmail.com>
Message-ID: <CAKzWOxW9BCKxoC1jofzdKyFpNxis1Bc9nFge00gx42VEkbT8tw@mail.gmail.com>

>
>
> To help connect the dots I added some links to/from that Lua wiki page and
> channel_in_hangup_hook and also api_hangup_hook and reporting_hangup_hook.
>
> -MC
>
Sorry, typo: that was supposed to be "api_reporting_hook" which I need to
document anyway. ;)
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110804/bc005821/attachment.html 

From jmesquita at freeswitch.org  Fri Aug  5 04:25:40 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Thu, 4 Aug 2011 21:25:40 -0300
Subject: [Freeswitch-users] Does FreeSWITCH supports Googles VP8 (WebM)
	Codec
In-Reply-To: <26A4B29329D049A4AA15FA1B375BC7CA@gmail.com>
References: <4E3A710F.3000003@gmx.de>
	<26A4B29329D049A4AA15FA1B375BC7CA@gmail.com>
Message-ID: <CABH8QX03ishdEiHfb8EGXcH4GHCKwhaRA-9SRZ2Mg-Aj4d5caA@mail.gmail.com>

>From what I could see, a mod_vp8 would be cool because its license is
BSD-like and not like h264 and others that have patent restricting use.

The problem is that, afaik there is no endpoints using vp8 nowadays so it
would be restricted to some internet application or custom application
making use of vp8...

Maybe the experts could shed some light on the matter, I don't even know if
VP8 would be suitable for phone calls...

Regards,
Jo?o Mesquita



On Thu, Aug 4, 2011 at 8:57 PM, Seven Du <dujinfang at gmail.com> wrote:

>  All video codecs is passthrough currently in FS, so it doesn't matter of
> performance in FS even it might matter in other parts of BBB.
>
> I guess you could easily write a passthrough module for VP8 just like
> mod_h26x.
> --
> Seven Du
> About: http://about.me/dujinfang
> Blog: http://www.dujinfang.com
> Proj:  http://www.freeswitch.org.cn
> Sent with Sparrow <http://www.sparrowmailapp.com>
>
>
> On Thursday, August 4, 2011 at 6:14 PM, Markus Siebeneicher wrote:
>
> Hello folks from FreeSWITCH,
>
> i am using FreeSWITCH with the conference system BigBlueButton, made by
> people from canada.
>
> It works pretty fine, so far. Due to the need that we would like to
> support more than 500 users in a single video/audio session, we are in
> interest to use the best codecs.
>
> So i listen to VP8 from Google. Does FreeSWITCH support it anyway? I
> couldn't find something in the FAQ's. I listen something about ffmeg, is
> it in this abstraction module?
>
> Thanks for a short reply and thanks for the good work!
>
> Kind regards,
>
> Markus
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110804/cad06ca3/attachment.html 

From steveu at coppice.org  Fri Aug  5 06:17:14 2011
From: steveu at coppice.org (Steve Underwood)
Date: Fri, 05 Aug 2011 10:17:14 +0800
Subject: [Freeswitch-users] Does FreeSWITCH supports Googles VP8 (WebM)
 Codec
In-Reply-To: <CABH8QX03ishdEiHfb8EGXcH4GHCKwhaRA-9SRZ2Mg-Aj4d5caA@mail.gmail.com>
References: <4E3A710F.3000003@gmx.de>	<26A4B29329D049A4AA15FA1B375BC7CA@gmail.com>
	<CABH8QX03ishdEiHfb8EGXcH4GHCKwhaRA-9SRZ2Mg-Aj4d5caA@mail.gmail.com>
Message-ID: <4E3B52AA.70805@coppice.org>

On 08/05/2011 08:25 AM, Jo?o Mesquita wrote:
> >From what I could see, a mod_vp8 would be cool because its license is 
> BSD-like and not like h264 and others that have patent restricting use.
>
> The problem is that, afaik there is no endpoints using vp8 nowadays so 
> it would be restricted to some internet application or custom 
> application making use of vp8...
>
> Maybe the experts could shed some light on the matter, I don't even 
> know if VP8 would be suitable for phone calls...
>
> Regards,
> Jo?o Mesquita
>
I read the Skype is about to start using VP8. This seems pretty odd, 
considering who their new owners are, but maybe the decision was made 
long ago. If that's right, the availability of endpoints will change 
very rapidly.... as long as you can interwork with them. It seems VP8 
was designed to be video conference friendly, but I have no idea just 
how friendly. Google seem to have done serious work on speeding up 
encode on a PC.

Steve



From simon0922 at gmail.com  Fri Aug  5 07:21:24 2011
From: simon0922 at gmail.com (Simon Leck)
Date: Fri, 5 Aug 2011 11:21:24 +0800
Subject: [Freeswitch-users] Disable Allow Events in Fs
Message-ID: <000901cc531e$c2f06340$48d129c0$@gmail.com>

Hi Everybody,

 

I am trying to disable allow events in FS. Can some give me some guideline
on how to do it. 

 

Thanks in advanced.

 

Many thanks

Simon

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110805/d20937b6/attachment-0001.html 

From tculjaga at gmail.com  Fri Aug  5 11:18:03 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 5 Aug 2011 09:18:03 +0200
Subject: [Freeswitch-users] Mod_rad_auth issue for FS working with
 FreeRadius server
In-Reply-To: <CAOTmbMzu2qDYh2T24JSpCYn1xH2J4iGiK7UkUB4Kq=-KdcDWGQ@mail.gmail.com>
References: <CAOTmbMzvd5OHL8z=_8vu6Orp2_prGxaw1gmXYCM2=2+pgzs_WA@mail.gmail.com>
	<CAJoKKdX01Qc8ibs25PUELi8GrzMDNOrHb_bQvYdq3ZqA4XLy4w@mail.gmail.com>
	<CAOTmbMxL3pLOuw655fyT5VzAON_N0PrDN+-i-++=oZoYU6+Fmg@mail.gmail.com>
	<CAOTmbMyvyDA-CRpaMC=WRGVkiyTG=Rs+n4hyeQWdyTZpqysSzw@mail.gmail.com>
	<CAJoKKdXSwF02S9tBkorGwOjj57AtkRpU0c177qXASEpodKVk=w@mail.gmail.com>
	<CAOTmbMzu2qDYh2T24JSpCYn1xH2J4iGiK7UkUB4Kq=-KdcDWGQ@mail.gmail.com>
Message-ID: <CAJoKKdU66jK3f+mOEV90Y3zrmzpTSdWnwGsWimZxir42khiJ-w@mail.gmail.com>

add to rad_auth.conf.xml

<param name="NAS-Port-Type" id="61" value="0" pec="0" expr="0"
direction="in"/>
<param name="Login-User" id="6" value="1" pec="0" expr="0" direction="in"/>



as for Auth Type im not sure if you need it ... this is up to your server.
According to dictionary file you need to set it as follows:

<param name="Auth-Type" id="1000" value="?" pec="0" expr="0"
direction="in"/>

the value (set as ?) is one of the folowing. Again, not sure what is
required by your server.

VALUE           Auth-Type               Local                   0
VALUE           Auth-Type               System                  1
VALUE           Auth-Type               SecurID                 2
VALUE           Auth-Type               Crypt-Local             3
VALUE           Auth-Type               Reject                  4

#
#       Cistron extensions
#
VALUE           Auth-Type               Pam                     253
VALUE           Auth-Type               Accept                  254



regards,
Tihomir.


On Wed, Aug 3, 2011 at 6:32 AM, fieldpeak <fieldpeak at gmail.com> wrote:

> Hi Tihomir,
>
> Sorry, i missed your mail in gmail before, just now saw it, and after using
> your dictionary.all, the dictionary issue was resolved, very appreciated for
> your kindly help! however, it did not fully functional yet,
>
> Attached are configuration files that i used, when i dial 601 to trigger to
> auth, the freeradius server shows log below, the supecious log is the value
> User-Password, it should be '1111' that i've set in the mysql db of
> freeradisu server for the user 1001 .
>
> i searched in google, for "known good" password issue, i suggest change
> user-password to cleartext-password, however, i did not find where it is.
> and also the Auth-Type, where to configure it...
>
> Freeradius server log:
>
> rad_recv: Access-Request packet from host 127.0.0.1 port 52684, id=49,
> length=111
>         User-Name = "1001"
>         User-Password = "?\210\365@\263\t\306\343\243iT?\311C\t\002"
>         Called-Station-Id = "888"
>         h323-conf-id = "749d2b5a-16ad-48e4-af58-24011949d1b5"
>         Calling-Station-Id = "1001"
>         NAS-Port = 0
>         NAS-IP-Address = 127.0.0.1
> # Executing section authorize from file
> /usr/local/etc/raddb/sites-enabled/default
> +- entering group authorize {...}
> ++[preprocess] returns ok
> [auth_log]      expand:
> /usr/local/var/log/radius/radacct/%{Client-IP-Address}/auth-detail-%Y%m%d ->
> /usr/local/var/log/radius/radacct/127.0.0.1/auth-detail-20110803
> [auth_log]
> /usr/local/var/log/radius/radacct/%{Client-IP-Address}/auth-detail-%Y%m%d
> expands to /usr/local/var/log/radius/radacct/
> 127.0.0.1/auth-detail-20110803
> [auth_log]      expand: %t -> Wed Aug  3 12:06:33 2011
> ++[auth_log] returns ok
> ++[chap] returns noop
> ++[mschap] returns noop
> ++[digest] returns noop
> [suffix] No '@' in User-Name = "1001", looking up realm NULL
> [suffix] No such realm "NULL"
> ++[suffix] returns noop
> [eap] No EAP-Message, not doing EAP
> ++[eap] returns noop
> ++[unix] returns notfound
> ++[files] returns noop
> [sql]   expand: %{User-Name} -> 1001
> [sql] sql_set_user escaped user --> '1001'
> rlm_sql (sql): Reserving sql socket id: 4
> [sql]   expand: SELECT id, username, attribute, value, op           FROM
> radcheck           WHERE username = '%{SQL-User-Name}'           ORDER BY id
> -> SELECT id, username, attribute, value, op           FROM
> radcheck           WHERE username = '1001'           ORDER BY id
> [sql]   expand: SELECT groupname           FROM radusergroup
> WHERE username = '%{SQL-User-Name}'           ORDER BY priority -> SELECT
> groupname           FROM radusergroup           WHERE username =
> '1001'           ORDER BY priority
> rlm_sql (sql): Released sql socket id: 4
> [sql] User 1001 not found
> ++[sql] returns notfound
> ++[expiration] returns noop
> ++[logintime] returns noop
> [pap] WARNING! No "known good" password found for the user.  Authentication
> may fail because of this.
> ++[pap] returns noop
> ERROR: No authenticate method (Auth-Type) found for the request: Rejecting
> the user
> Failed to authenticate the user.
>   WARNING: Unprintable characters in the password.        Double-check the
> shared secret on the server and the NAS!
> Using Post-Auth-Type Reject
> # Executing group from file /usr/local/etc/raddb/sites-enabled/default
> +- entering group REJECT {...}
> [attr_filter.access_reject]     expand: %{User-Name} -> 1001
>  attr_filter: Matched entry DEFAULT at line 11
> ++[attr_filter.access_reject] returns updated
> Delaying reject of request 8 for 1 seconds
> Going to the next request
> Waking up in 0.9 seconds.
> Sending delayed reject for request 8
> Sending Access-Reject of id 49 to 127.0.0.1 port 52684
> Waking up in 4.9 seconds.
> Cleaning up request 8 ID 49 with timestamp +7674
> Ready to process requests.
> WARNING! No "known good" password found for the user
>
> Regards,
> Charles
>
>
> 2011/8/3 Tihomir Culjaga <tculjaga at gmail.com>
>
>> did u use the dictionary i have attached ?
>>
>>
>> On Tue, Aug 2, 2011 at 10:08 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>>
>>> i tried change to 'h323-conf-id' to 'h323-call-origin' in
>>> 02_unitest_rad-ANI-auth.xml, rad_auth.conf.xml, however, it still prompt
>>> '[ERR] mod_rad_auth.c:428 Unknown attribute: key:h323-conf-id, not found
>>> in dictionary', so where the mod_rad_auth read out the 'h323-conf-id'? very
>>> very strange, which dictionary it was using...
>>>
>>> Regards,
>>> Charles
>>>
>>>
>>> 2011/8/2 fieldpeak <fieldpeak at gmail.com>
>>>
>>>> Hi Tihomir,
>>>>
>>>> Finally the answer coming, i see the hope, thanks for your reply, :)
>>>>
>>>> As your advise, i only use one attribute(h323-conf-id) in my dialplan,
>>>> and only one attribute(h323-conf-id) in rad_auth.conf.xml, and using the
>>>> attached dictionary (from ciso) which contains this attribute, however, it
>>>> still prompt 'unknown attribute', so i suspected if it was reading
>>>> /usr/local/etc/radiusclient/dictionary, so i copy the same dictionary to
>>>> /usr/local/freeswitch/radius/, it did not any help at all... very strange...
>>>>
>>>> Log:
>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set default_realm
>>>> := .
>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set radius_timeout
>>>> := 3.
>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set radius_retries
>>>> := 2.
>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set
>>>> radius_deadtime := 0.
>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set bindaddr := *.
>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:371 ... radius:
>>>> User-Name: 38516060333
>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:380 ... radius:
>>>> User-Password: 003282
>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:396 ... radius:
>>>> Called-station-Id: 16094191500
>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:413 Handle attribute:
>>>> h323-conf-id
>>>> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:428 Unknown attribute:
>>>> key:h323-conf-id, not found in dictionary
>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:538 abort sending
>>>> radius packet.
>>>> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:546 An error occured
>>>> during RADIUS Authentication(RC=-1)
>>>> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:702 An error occured
>>>> during radius authorization.
>>>>
>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  AUTH_RESULT=)
>>>>
>>>>
>>>>
>>>>   <extension name="unitest_rad-ANI-auth">
>>>>     <condition field="destination_number" expression="^601$">
>>>>       <!-- <action application="log" data="INFO  Before Auth "/> -->
>>>>
>>>>       <action inline="true" application="set" data="CALLID=h323-conf-id
>>>> =${uuid}"/>
>>>>
>>>>       <action inline="true" application="set" data="USERNAME=1001"/>
>>>>       <action inline="true" application="set" data="PASSWD=1111"/>
>>>>
>>>>
>>>>       <action application="sleep" data="2000"/>
>>>>       <action application="auth_function" data="in ${DIALED_NUMBER}, in
>>>> ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>
>>>>
>>>>     </condition>
>>>>   </extension>
>>>>
>>>>
>>>>
>>>> <configuration name="rad_auth.conf" description="radius authentification
>>>> module">
>>>>   <settings>
>>>>
>>>>   </settings>
>>>>
>>>>   <client>
>>>>     <param name="authserver" value="127.0.0.1:1812:gateway"/>
>>>>     <param name="dictionary"
>>>> value="/usr/local/etc/radiusclient/dictionary"/>
>>>>     <param name="seqfile" value="/var/run/radius.seq"/>
>>>>     <param name="mapfile"
>>>> value="/usr/local/etc/radiusclient/port-id-map"/>
>>>>     <param name="default_realm" value=""/>
>>>>     <param name="radius_timeout" value="3"/>
>>>>     <param name="radius_retries" value="2"/>
>>>>     <param name="radius_deadtime" value="0"/>
>>>>     <param name="bindaddr" value="*"/>
>>>>   </client>
>>>>
>>>>   <vsas>
>>>>
>>>>
>>>>     <param name="h323-conf-id" id="24" value="CALLID" pec="9" expr="1"
>>>> direction="in"/>
>>>>
>>>>   </vsas>
>>>>  </configuration>
>>>>
>>>>
>>>>
>>>> 2011/8/2 Tihomir Culjaga <tculjaga at gmail.com>
>>>>
>>>>> hi,
>>>>>
>>>>> dictionary.all is just the name of a file containing all attributes i
>>>>> needed at that time.
>>>>>
>>>>> you can include other dictionaries by putting #INCLUDE <pathname> at
>>>>> the end of the dictionary file you reference in rad_auth.conf.xml.
>>>>> if the INCLUDE doesn't work, just append dictionary.cisco to your
>>>>> dictionary file... and make your own file.
>>>>>
>>>>>
>>>>> check inline comments down below...
>>>>>
>>>>>
>>>>> T.
>>>>>
>>>>>
>>>>> On Sun, Jul 31, 2011 at 10:46 AM, fieldpeak <fieldpeak at gmail.com>wrote:
>>>>>
>>>>>> Hello Gurus,
>>>>>>
>>>>>> i met a issue when using
>>>>>> mod_rad_auth(http://wiki.freeswitch.org/wiki/Mod_rad_auth) to works
>>>>>> with freeradius server+mysql for AAA, the details is below, Could
>>>>>> anyone give any hints, Thanks in advance.
>>>>>>
>>>>>> i setup a dial plan "unitest_rad-ANI-auth" as wiki above, however,
>>>>>> when i dialed 601 to trigger the dial plan, the console show errors,
>>>>>> it looks "h323-conf-id" is not in the directory, then i tried to add
>>>>>> this attribute to the dictionary, however, it does not help, in the
>>>>>> wiki, it mentioned the rad_auth.conf.xml contains <param
>>>>>> name="dictionary"
>>>>>> value="/usr/local/etc/radiusclient/dictionary.all"/>, however i did
>>>>>> not find the file "dictionary.all" at that directory, so i use
>>>>>> dictionary. BTW, the freeradius server + mysql works well.
>>>>>>
>>>>>
>>>>> i just appended the information needed into dictionary.all file...
>>>>> (vendor and attribute definition).
>>>>>
>>>>>
>>>>>
>>>>>>
>>>>>> console errors:
>>>>>>
>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 auth_function(in , in
>>>>>> 38516060333, in 003282, out AUTH_RESULT)
>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:301 allocate initial
>>>>>> structure.
>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:313 initialzed
>>>>>> configuration.
>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set authserver
>>>>>> := 127.0.0.1:1812:gateway.
>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set dictionary
>>>>>> := /usr/local/etc/radiusclient/dictionary.
>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set seqfile :=
>>>>>> /var/run/radius.seq.
>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set mapfile :=
>>>>>> /usr/local/etc/radiusclient/port-id-map.
>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>> default_realm := .
>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>> radius_timeout := 3.
>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>> radius_retries := 2.
>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>> radius_deadtime := 0.
>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set bindaddr :=
>>>>>> *.
>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:371 ... radius:
>>>>>> User-Name: 38516060333
>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:380 ... radius:
>>>>>> User-Password: 003282
>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:391 ... radius:
>>>>>> Called-station-Id is empty, ignoring...
>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:413 Handle
>>>>>> attribute: h323-conf-id
>>>>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:428 Unknown attribute:
>>>>>> key:h323-conf-id, not found in dictionary
>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:538 abort sending
>>>>>> radius packet.
>>>>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:546 An error occured
>>>>>> during RADIUS Authentication(RC=-1)
>>>>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:702 An error occured
>>>>>> during radius authorization.
>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  AUTH_RESULT=)
>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  AUTH_RESULT=
>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  billing_model=)
>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  billing_model=
>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  credit_amount=)
>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  credit_amount=
>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  currency=)
>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  currency=
>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO
>>>>>>  preffered_lang=)
>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  preffered_lang=
>>>>>>
>>>>>> added below in the dictionary(/usr/local/etc/radiusclient/dictionary):
>>>>>>
>>>>>> ATTRIBUTE       h323-conf-id            1008    string
>>>>>>
>>>>>
>>>>> you need the vendor definition as well
>>>>>
>>>>>
>>>>>>
>>>>>>
>>>>>> dial plan:
>>>>>> <extension name="unitest_rad-ANI-auth">
>>>>>>    <condition field="destination_number" expression="^601$">
>>>>>>      <action application="log" data="INFO  Before Auth "/>
>>>>>>
>>>>>>      <action inline="true" application="set"
>>>>>> data="CALLID=h323-conf-id=${uuid}"/>
>>>>>>      <action inline="true" application="set"
>>>>>> data="SERVICENUM=h323-prompt-id=${destination_number}"/>
>>>>>>      <action inline="true" application="set"
>>>>>> data="TRANSACTIONID=h323-ivr-out=transactionID:1234"/>
>>>>>>  <!--      <action inline="true" application="set"
>>>>>> data="CALLINGNUMBER=${caller_id_number}"/> -->
>>>>>>      <action inline="true" application="set"
>>>>>> data="CALLINGNUMBER=38516060333"/>
>>>>>>      <action inline="true" application="set"
>>>>>> data="USERNAME=38516060333"/>
>>>>>>  <!--      <action inline="true" application="set"
>>>>>> data="USERNAME=209354"/> -->
>>>>>>      <action inline="true" application="set" data="PASSWD=003282"/>
>>>>>>  <!--      <action inline="true" application="set"
>>>>>> data="DIALED_NUMBER=16094191500"/>  -->
>>>>>>
>>>>>>      <action application="sleep" data="2000"/>
>>>>>>      <action application="auth_function" data="in ${DIALED_NUMBER},
>>>>>> in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>
>>>>>>
>>>>>>
>>>>>>      <action application="log" data="INFO
>>>>>>  AUTH_RESULT=${AUTH_RESULT}"/>
>>>>>>      <action application="log" data="INFO
>>>>>>  billing_model=${billing_model}"/>
>>>>>>      <action application="log" data="INFO
>>>>>>  credit_amount=${credit_amount}"/>
>>>>>>      <action application="log" data="INFO  currency=${currency}"/>
>>>>>>      <action application="log" data="INFO
>>>>>>  preffered_lang=${preffered_lang}"/>
>>>>>>      <action application="log" data="INFO
>>>>>>  credit_time=${credit_time}"/>
>>>>>>      <action application="log" data="INFO
>>>>>> h323_ivr_duration=${h323_ivr_duration}"/>
>>>>>>      <action application="log" data="INFO
>>>>>>  return_code=${return_code}"/>
>>>>>>      <!-- <action application="execute_extension" data="AUTH XML
>>>>>> default"/> -->
>>>>>>    </condition>
>>>>>>  </extension>
>>>>>>
>>>>>>  radius_cdr.conf.xml:
>>>>>>  <configuration name="radius_cdr.conf" description="RADIUS CDR
>>>>>> Configuration">
>>>>>>
>>>>>>        <settings>
>>>>>>
>>>>>>                <!-- location of the radius dictionary files -->
>>>>>>
>>>>>>                <param name="dictionary"
>>>>>> value="/usr/local/freeswitch/conf/radius/dictionary"/>
>>>>>>
>>>>>>
>>>>> your dictionary file need to contain all the attributes you are trying
>>>>> to use or to include other dictionaries (In this case dictionary.cisco) from
>>>>> the dictionary file you are referencing here.
>>>>>
>>>>>
>>>>>>                <!-- number of retries for each server -->
>>>>>>
>>>>>>                <param name="radius_retries" value="3"/>
>>>>>>
>>>>>>                <!-- number of seconds to wait between retries -->
>>>>>>
>>>>>>                <param name="radius_timeout" value="5"/>
>>>>>>
>>>>>>                <!-- accounting servers, up to 8 allowed -->
>>>>>>
>>>>>>                <!-- value is "host:port:secret", port is optional -->
>>>>>>
>>>>>>                <!-- use IP ADDRESSES, not hostnames -->
>>>>>>
>>>>>>                <param name="acct_server" value="127.0.0.1:1813
>>>>>> :testing123"/>
>>>>>>
>>>>>>
>>>>>>        </settings>
>>>>>>
>>>>>> </configuration>
>>>>>>
>>>>>>  the FS version:
>>>>>>  FreeSWITCH Version 1.0.head (git-492bc6b 2011-07-23 12-53-04 -0400)
>>>>>>
>>>>>>  Regards,
>>>>>>  Charles
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>>
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110805/6464bc46/attachment-0001.html 

From fieldpeak at gmail.com  Fri Aug  5 12:12:46 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Fri, 5 Aug 2011 16:12:46 +0800
Subject: [Freeswitch-users] Mod_rad_auth issue for FS working with
 FreeRadius server
In-Reply-To: <CAJoKKdU66jK3f+mOEV90Y3zrmzpTSdWnwGsWimZxir42khiJ-w@mail.gmail.com>
References: <CAOTmbMzvd5OHL8z=_8vu6Orp2_prGxaw1gmXYCM2=2+pgzs_WA@mail.gmail.com>
	<CAJoKKdX01Qc8ibs25PUELi8GrzMDNOrHb_bQvYdq3ZqA4XLy4w@mail.gmail.com>
	<CAOTmbMxL3pLOuw655fyT5VzAON_N0PrDN+-i-++=oZoYU6+Fmg@mail.gmail.com>
	<CAOTmbMyvyDA-CRpaMC=WRGVkiyTG=Rs+n4hyeQWdyTZpqysSzw@mail.gmail.com>
	<CAJoKKdXSwF02S9tBkorGwOjj57AtkRpU0c177qXASEpodKVk=w@mail.gmail.com>
	<CAOTmbMzu2qDYh2T24JSpCYn1xH2J4iGiK7UkUB4Kq=-KdcDWGQ@mail.gmail.com>
	<CAJoKKdU66jK3f+mOEV90Y3zrmzpTSdWnwGsWimZxir42khiJ-w@mail.gmail.com>
Message-ID: <CAOTmbMxW=EDr3-Ayo2R_MsCCTUZ4PnDkkRUuR2FisH9tK5xkWA@mail.gmail.com>

Hi Tihomir,

Thanks for your advise, i've added below to rad_auth.conf.xml (vsas
section), as well as tried auth-type to 0(local) and 1(system), however, the
issue still exist.

<param name="NAS-Port-Type" id="61" value="0" pec="0" expr="0"
direction="in"/>
<param name="Login-User" id="6" value="1" pec="0" expr="0" direction="in"/>
<param name="Auth-Type" id="1000" value="0" pec="0" expr="0"
direction="in"/>

FreeRadius output:

Found Auth-Type = PAP
# Executing group from file /usr/local/etc/raddb/sites-enabled/default
+- entering group PAP {...}
[pap] login attempt with password "Q?????? ??????p???F?+??a"
[pap] Using clear text password "1111"
[pap] Passwords don't match
++[pap] returns reject
Failed to authenticate the user.
  WARNING: Unprintable characters in the password.        Double-check
the shared secret on the server and the NAS!

Using Post-Auth-Type Reject
# Executing group from file /usr/local/etc/raddb/sites-enabled/default
+- entering group REJECT {...}
[attr_filter.access_reject]     expand: %{User-Name} -> 1001
 attr_filter: Matched entry DEFAULT at line 11
++[attr_filter.access_reject] returns updated
Delaying reject of request 38 for 1 seconds

Regards,
Charles

2011/8/5 Tihomir Culjaga <tculjaga at gmail.com>

> add to rad_auth.conf.xml
>
> <param name="NAS-Port-Type" id="61" value="0" pec="0" expr="0"
> direction="in"/>
> <param name="Login-User" id="6" value="1" pec="0" expr="0" direction="in"/>
>
>
>
> as for Auth Type im not sure if you need it ... this is up to your server.
> According to dictionary file you need to set it as follows:
>
> <param name="Auth-Type" id="1000" value="?" pec="0" expr="0"
> direction="in"/>
>
> the value (set as ?) is one of the folowing. Again, not sure what is
> required by your server.
>
> VALUE           Auth-Type               Local                   0
> VALUE           Auth-Type               System                  1
> VALUE           Auth-Type               SecurID                 2
> VALUE           Auth-Type               Crypt-Local             3
> VALUE           Auth-Type               Reject                  4
>
> #
> #       Cistron extensions
> #
> VALUE           Auth-Type               Pam                     253
> VALUE           Auth-Type               Accept                  254
>
>
>
> regards,
> Tihomir.
>
>
>
> On Wed, Aug 3, 2011 at 6:32 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>
>> Hi Tihomir,
>>
>> Sorry, i missed your mail in gmail before, just now saw it, and after
>> using your dictionary.all, the dictionary issue was resolved, very
>> appreciated for your kindly help! however, it did not fully functional yet,
>>
>> Attached are configuration files that i used, when i dial 601 to trigger
>> to auth, the freeradius server shows log below, the supecious log is the
>> value User-Password, it should be '1111' that i've set in the mysql db of
>> freeradisu server for the user 1001 .
>>
>> i searched in google, for "known good" password issue, i suggest change
>> user-password to cleartext-password, however, i did not find where it is.
>> and also the Auth-Type, where to configure it...
>>
>> Freeradius server log:
>>
>> rad_recv: Access-Request packet from host 127.0.0.1 port 52684, id=49,
>> length=111
>>         User-Name = "1001"
>>         User-Password = "?\210\365@\263\t\306\343\243iT?\311C\t\002"
>>         Called-Station-Id = "888"
>>         h323-conf-id = "749d2b5a-16ad-48e4-af58-24011949d1b5"
>>         Calling-Station-Id = "1001"
>>         NAS-Port = 0
>>         NAS-IP-Address = 127.0.0.1
>> # Executing section authorize from file
>> /usr/local/etc/raddb/sites-enabled/default
>> +- entering group authorize {...}
>> ++[preprocess] returns ok
>> [auth_log]      expand:
>> /usr/local/var/log/radius/radacct/%{Client-IP-Address}/auth-detail-%Y%m%d ->
>> /usr/local/var/log/radius/radacct/127.0.0.1/auth-detail-20110803
>> [auth_log]
>> /usr/local/var/log/radius/radacct/%{Client-IP-Address}/auth-detail-%Y%m%d
>> expands to /usr/local/var/log/radius/radacct/
>> 127.0.0.1/auth-detail-20110803
>> [auth_log]      expand: %t -> Wed Aug  3 12:06:33 2011
>> ++[auth_log] returns ok
>> ++[chap] returns noop
>> ++[mschap] returns noop
>> ++[digest] returns noop
>> [suffix] No '@' in User-Name = "1001", looking up realm NULL
>> [suffix] No such realm "NULL"
>> ++[suffix] returns noop
>> [eap] No EAP-Message, not doing EAP
>> ++[eap] returns noop
>> ++[unix] returns notfound
>> ++[files] returns noop
>> [sql]   expand: %{User-Name} -> 1001
>> [sql] sql_set_user escaped user --> '1001'
>> rlm_sql (sql): Reserving sql socket id: 4
>> [sql]   expand: SELECT id, username, attribute, value, op           FROM
>> radcheck           WHERE username = '%{SQL-User-Name}'           ORDER BY id
>> -> SELECT id, username, attribute, value, op           FROM
>> radcheck           WHERE username = '1001'           ORDER BY id
>> [sql]   expand: SELECT groupname           FROM radusergroup
>> WHERE username = '%{SQL-User-Name}'           ORDER BY priority -> SELECT
>> groupname           FROM radusergroup           WHERE username =
>> '1001'           ORDER BY priority
>> rlm_sql (sql): Released sql socket id: 4
>> [sql] User 1001 not found
>> ++[sql] returns notfound
>> ++[expiration] returns noop
>> ++[logintime] returns noop
>> [pap] WARNING! No "known good" password found for the user.
>> Authentication may fail because of this.
>> ++[pap] returns noop
>> ERROR: No authenticate method (Auth-Type) found for the request: Rejecting
>> the user
>> Failed to authenticate the user.
>>   WARNING: Unprintable characters in the password.        Double-check the
>> shared secret on the server and the NAS!
>> Using Post-Auth-Type Reject
>> # Executing group from file /usr/local/etc/raddb/sites-enabled/default
>> +- entering group REJECT {...}
>> [attr_filter.access_reject]     expand: %{User-Name} -> 1001
>>  attr_filter: Matched entry DEFAULT at line 11
>> ++[attr_filter.access_reject] returns updated
>> Delaying reject of request 8 for 1 seconds
>> Going to the next request
>> Waking up in 0.9 seconds.
>> Sending delayed reject for request 8
>> Sending Access-Reject of id 49 to 127.0.0.1 port 52684
>> Waking up in 4.9 seconds.
>> Cleaning up request 8 ID 49 with timestamp +7674
>> Ready to process requests.
>> WARNING! No "known good" password found for the user
>>
>> Regards,
>> Charles
>>
>>
>> 2011/8/3 Tihomir Culjaga <tculjaga at gmail.com>
>>
>>> did u use the dictionary i have attached ?
>>>
>>>
>>> On Tue, Aug 2, 2011 at 10:08 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>>>
>>>> i tried change to 'h323-conf-id' to 'h323-call-origin' in
>>>> 02_unitest_rad-ANI-auth.xml, rad_auth.conf.xml, however, it still prompt
>>>> '[ERR] mod_rad_auth.c:428 Unknown attribute: key:h323-conf-id, not
>>>> found in dictionary', so where the mod_rad_auth read out the 'h323-conf-id'?
>>>> very very strange, which dictionary it was using...
>>>>
>>>> Regards,
>>>> Charles
>>>>
>>>>
>>>> 2011/8/2 fieldpeak <fieldpeak at gmail.com>
>>>>
>>>>> Hi Tihomir,
>>>>>
>>>>> Finally the answer coming, i see the hope, thanks for your reply, :)
>>>>>
>>>>> As your advise, i only use one attribute(h323-conf-id) in my dialplan,
>>>>> and only one attribute(h323-conf-id) in rad_auth.conf.xml, and using the
>>>>> attached dictionary (from ciso) which contains this attribute, however, it
>>>>> still prompt 'unknown attribute', so i suspected if it was reading
>>>>> /usr/local/etc/radiusclient/dictionary, so i copy the same dictionary to
>>>>> /usr/local/freeswitch/radius/, it did not any help at all... very strange...
>>>>>
>>>>> Log:
>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set default_realm
>>>>> := .
>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set
>>>>> radius_timeout := 3.
>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set
>>>>> radius_retries := 2.
>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set
>>>>> radius_deadtime := 0.
>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set bindaddr :=
>>>>> *.
>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:371 ... radius:
>>>>> User-Name: 38516060333
>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:380 ... radius:
>>>>> User-Password: 003282
>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:396 ... radius:
>>>>> Called-station-Id: 16094191500
>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:413 Handle attribute:
>>>>> h323-conf-id
>>>>> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:428 Unknown attribute:
>>>>> key:h323-conf-id, not found in dictionary
>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:538 abort sending
>>>>> radius packet.
>>>>> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:546 An error occured
>>>>> during RADIUS Authentication(RC=-1)
>>>>> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:702 An error occured
>>>>> during radius authorization.
>>>>>
>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  AUTH_RESULT=)
>>>>>
>>>>>
>>>>>
>>>>>   <extension name="unitest_rad-ANI-auth">
>>>>>     <condition field="destination_number" expression="^601$">
>>>>>       <!-- <action application="log" data="INFO  Before Auth "/> -->
>>>>>
>>>>>       <action inline="true" application="set" data="CALLID=
>>>>> h323-conf-id=${uuid}"/>
>>>>>
>>>>>       <action inline="true" application="set" data="USERNAME=1001"/>
>>>>>       <action inline="true" application="set" data="PASSWD=1111"/>
>>>>>
>>>>>
>>>>>       <action application="sleep" data="2000"/>
>>>>>       <action application="auth_function" data="in ${DIALED_NUMBER}, in
>>>>> ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>
>>>>>
>>>>>     </condition>
>>>>>   </extension>
>>>>>
>>>>>
>>>>>
>>>>> <configuration name="rad_auth.conf" description="radius
>>>>> authentification module">
>>>>>   <settings>
>>>>>
>>>>>   </settings>
>>>>>
>>>>>   <client>
>>>>>     <param name="authserver" value="127.0.0.1:1812:gateway"/>
>>>>>     <param name="dictionary"
>>>>> value="/usr/local/etc/radiusclient/dictionary"/>
>>>>>     <param name="seqfile" value="/var/run/radius.seq"/>
>>>>>     <param name="mapfile"
>>>>> value="/usr/local/etc/radiusclient/port-id-map"/>
>>>>>     <param name="default_realm" value=""/>
>>>>>     <param name="radius_timeout" value="3"/>
>>>>>     <param name="radius_retries" value="2"/>
>>>>>     <param name="radius_deadtime" value="0"/>
>>>>>     <param name="bindaddr" value="*"/>
>>>>>   </client>
>>>>>
>>>>>   <vsas>
>>>>>
>>>>>
>>>>>     <param name="h323-conf-id" id="24" value="CALLID" pec="9" expr="1"
>>>>> direction="in"/>
>>>>>
>>>>>   </vsas>
>>>>>  </configuration>
>>>>>
>>>>>
>>>>>
>>>>> 2011/8/2 Tihomir Culjaga <tculjaga at gmail.com>
>>>>>
>>>>>> hi,
>>>>>>
>>>>>> dictionary.all is just the name of a file containing all attributes i
>>>>>> needed at that time.
>>>>>>
>>>>>> you can include other dictionaries by putting #INCLUDE <pathname> at
>>>>>> the end of the dictionary file you reference in rad_auth.conf.xml.
>>>>>> if the INCLUDE doesn't work, just append dictionary.cisco to your
>>>>>> dictionary file... and make your own file.
>>>>>>
>>>>>>
>>>>>> check inline comments down below...
>>>>>>
>>>>>>
>>>>>> T.
>>>>>>
>>>>>>
>>>>>> On Sun, Jul 31, 2011 at 10:46 AM, fieldpeak <fieldpeak at gmail.com>wrote:
>>>>>>
>>>>>>> Hello Gurus,
>>>>>>>
>>>>>>> i met a issue when using
>>>>>>> mod_rad_auth(http://wiki.freeswitch.org/wiki/Mod_rad_auth) to works
>>>>>>> with freeradius server+mysql for AAA, the details is below, Could
>>>>>>> anyone give any hints, Thanks in advance.
>>>>>>>
>>>>>>> i setup a dial plan "unitest_rad-ANI-auth" as wiki above, however,
>>>>>>> when i dialed 601 to trigger the dial plan, the console show errors,
>>>>>>> it looks "h323-conf-id" is not in the directory, then i tried to add
>>>>>>> this attribute to the dictionary, however, it does not help, in the
>>>>>>> wiki, it mentioned the rad_auth.conf.xml contains <param
>>>>>>> name="dictionary"
>>>>>>> value="/usr/local/etc/radiusclient/dictionary.all"/>, however i did
>>>>>>> not find the file "dictionary.all" at that directory, so i use
>>>>>>> dictionary. BTW, the freeradius server + mysql works well.
>>>>>>>
>>>>>>
>>>>>> i just appended the information needed into dictionary.all file...
>>>>>> (vendor and attribute definition).
>>>>>>
>>>>>>
>>>>>>
>>>>>>>
>>>>>>> console errors:
>>>>>>>
>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 auth_function(in , in
>>>>>>> 38516060333, in 003282, out AUTH_RESULT)
>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:301 allocate
>>>>>>> initial
>>>>>>> structure.
>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:313 initialzed
>>>>>>> configuration.
>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set authserver
>>>>>>> := 127.0.0.1:1812:gateway.
>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set dictionary
>>>>>>> := /usr/local/etc/radiusclient/dictionary.
>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set seqfile :=
>>>>>>> /var/run/radius.seq.
>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set mapfile :=
>>>>>>> /usr/local/etc/radiusclient/port-id-map.
>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>> default_realm := .
>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>> radius_timeout := 3.
>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>> radius_retries := 2.
>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>> radius_deadtime := 0.
>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set bindaddr :=
>>>>>>> *.
>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:371 ... radius:
>>>>>>> User-Name: 38516060333
>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:380 ... radius:
>>>>>>> User-Password: 003282
>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:391 ... radius:
>>>>>>> Called-station-Id is empty, ignoring...
>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:413 Handle
>>>>>>> attribute: h323-conf-id
>>>>>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:428 Unknown
>>>>>>> attribute:
>>>>>>> key:h323-conf-id, not found in dictionary
>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:538 abort sending
>>>>>>> radius packet.
>>>>>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:546 An error occured
>>>>>>> during RADIUS Authentication(RC=-1)
>>>>>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:702 An error occured
>>>>>>> during radius authorization.
>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  AUTH_RESULT=)
>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  AUTH_RESULT=
>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO
>>>>>>>  billing_model=)
>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  billing_model=
>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO
>>>>>>>  credit_amount=)
>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  credit_amount=
>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  currency=)
>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  currency=
>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO
>>>>>>>  preffered_lang=)
>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  preffered_lang=
>>>>>>>
>>>>>>> added below in the
>>>>>>> dictionary(/usr/local/etc/radiusclient/dictionary):
>>>>>>>
>>>>>>> ATTRIBUTE       h323-conf-id            1008    string
>>>>>>>
>>>>>>
>>>>>> you need the vendor definition as well
>>>>>>
>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> dial plan:
>>>>>>> <extension name="unitest_rad-ANI-auth">
>>>>>>>    <condition field="destination_number" expression="^601$">
>>>>>>>      <action application="log" data="INFO  Before Auth "/>
>>>>>>>
>>>>>>>      <action inline="true" application="set"
>>>>>>> data="CALLID=h323-conf-id=${uuid}"/>
>>>>>>>      <action inline="true" application="set"
>>>>>>> data="SERVICENUM=h323-prompt-id=${destination_number}"/>
>>>>>>>      <action inline="true" application="set"
>>>>>>> data="TRANSACTIONID=h323-ivr-out=transactionID:1234"/>
>>>>>>>  <!--      <action inline="true" application="set"
>>>>>>> data="CALLINGNUMBER=${caller_id_number}"/> -->
>>>>>>>      <action inline="true" application="set"
>>>>>>> data="CALLINGNUMBER=38516060333"/>
>>>>>>>      <action inline="true" application="set"
>>>>>>> data="USERNAME=38516060333"/>
>>>>>>>  <!--      <action inline="true" application="set"
>>>>>>> data="USERNAME=209354"/> -->
>>>>>>>      <action inline="true" application="set" data="PASSWD=003282"/>
>>>>>>>  <!--      <action inline="true" application="set"
>>>>>>> data="DIALED_NUMBER=16094191500"/>  -->
>>>>>>>
>>>>>>>      <action application="sleep" data="2000"/>
>>>>>>>      <action application="auth_function" data="in ${DIALED_NUMBER},
>>>>>>> in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>
>>>>>>>
>>>>>>>
>>>>>>>      <action application="log" data="INFO
>>>>>>>  AUTH_RESULT=${AUTH_RESULT}"/>
>>>>>>>      <action application="log" data="INFO
>>>>>>>  billing_model=${billing_model}"/>
>>>>>>>      <action application="log" data="INFO
>>>>>>>  credit_amount=${credit_amount}"/>
>>>>>>>      <action application="log" data="INFO  currency=${currency}"/>
>>>>>>>      <action application="log" data="INFO
>>>>>>>  preffered_lang=${preffered_lang}"/>
>>>>>>>      <action application="log" data="INFO
>>>>>>>  credit_time=${credit_time}"/>
>>>>>>>      <action application="log" data="INFO
>>>>>>> h323_ivr_duration=${h323_ivr_duration}"/>
>>>>>>>      <action application="log" data="INFO
>>>>>>>  return_code=${return_code}"/>
>>>>>>>      <!-- <action application="execute_extension" data="AUTH XML
>>>>>>> default"/> -->
>>>>>>>    </condition>
>>>>>>>  </extension>
>>>>>>>
>>>>>>>  radius_cdr.conf.xml:
>>>>>>>  <configuration name="radius_cdr.conf" description="RADIUS CDR
>>>>>>> Configuration">
>>>>>>>
>>>>>>>        <settings>
>>>>>>>
>>>>>>>                <!-- location of the radius dictionary files -->
>>>>>>>
>>>>>>>                <param name="dictionary"
>>>>>>> value="/usr/local/freeswitch/conf/radius/dictionary"/>
>>>>>>>
>>>>>>>
>>>>>> your dictionary file need to contain all the attributes you are trying
>>>>>> to use or to include other dictionaries (In this case dictionary.cisco) from
>>>>>> the dictionary file you are referencing here.
>>>>>>
>>>>>>
>>>>>>>                <!-- number of retries for each server -->
>>>>>>>
>>>>>>>                <param name="radius_retries" value="3"/>
>>>>>>>
>>>>>>>                <!-- number of seconds to wait between retries -->
>>>>>>>
>>>>>>>                <param name="radius_timeout" value="5"/>
>>>>>>>
>>>>>>>                <!-- accounting servers, up to 8 allowed -->
>>>>>>>
>>>>>>>                <!-- value is "host:port:secret", port is optional -->
>>>>>>>
>>>>>>>                <!-- use IP ADDRESSES, not hostnames -->
>>>>>>>
>>>>>>>                <param name="acct_server" value="127.0.0.1:1813
>>>>>>> :testing123"/>
>>>>>>>
>>>>>>>
>>>>>>>        </settings>
>>>>>>>
>>>>>>> </configuration>
>>>>>>>
>>>>>>>  the FS version:
>>>>>>>  FreeSWITCH Version 1.0.head (git-492bc6b 2011-07-23 12-53-04 -0400)
>>>>>>>
>>>>>>>  Regards,
>>>>>>>  Charles
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>>
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110805/4ecff2e4/attachment-0001.html 

From valery.kalinin at gmail.com  Fri Aug  5 13:18:08 2011
From: valery.kalinin at gmail.com (Valery Kalinin)
Date: Fri, 5 Aug 2011 15:18:08 +0600
Subject: [Freeswitch-users] My new web utility for FreeSWITCH
Message-ID: <CAF=swh9Np+_gyhG+9fsiBM0b0OPb8GcZ6uoQh2bkb1S2=1Ri0Q@mail.gmail.com>

FreeSWITCH channel viewer

Web-based PHP utility for online view FreeSWITCH channels.
All calls are displayed _without_ refreshing the page.
Program uses comet-style transport, without using ajax & flash.
Tested on: Internet Explorer 8.0, FireFox 4.0.1
Some columns are made thin. Click on him for explode.
Any column collapse/explode on click.

Download & installation guide:
https://sites.google.com/site/freeswitched/home/channel-viewer

How this work (data = list of calls):

client (browser)??? server (PHP script)
---------------??? ??? -------------------------------------------------
send data??? --->??? get list of calls (call FreeSWITCH)
? ? ? ? ? ? ? ? ? ?? ??? ??? compare data
apply changes??? <---??? send only changes of data
? ? ? ? ? ? ? ? ? ? ? ?? ?? ??? ??? cicle while php script timeout
(max_execution_time):
? ? ? ? ? ? ? ? ? ? ? ?? ?? ??? ??? - get list of calls (call
FreeSWITCH) - each 1 sec
? ? ? ? ? ? ? ? ? ? ? ?? ?? ??? ??? - compare data
apply changes??? <---?? - send only changes of data
restart script??? <---??? after php script timeout send command to restart
send data??? --->??? ...repeat...

All these tricks are made for traffic minimization.

Tests and feature requests are welcome.

Thanks.


From ijurado at econcept.es  Fri Aug  5 13:54:09 2011
From: ijurado at econcept.es (Isaac Jurado)
Date: Fri, 5 Aug 2011 11:54:09 +0200
Subject: [Freeswitch-users] Hangup hooks on B legs
In-Reply-To: <CAKzWOxVwhUzjn32J4TRkvAtxhrOObbgoPb46NYk3D-4mra4dBg@mail.gmail.com>
References: <CABj_V-mOmm05ycYJZRPaeePeVzB4LWdTfzjNXwM3O=Kjmrn5tQ@mail.gmail.com>
	<CALNZuwm7Zh1FZpeavRECM6VjPUERsJODi=gRxWsXboR4hKEiLg@mail.gmail.com>
	<CABj_V-mubvz2cpCY1=ynLrAkDv088ZVURhBJ+1BiYe4XWhUyTA@mail.gmail.com>
	<CALNZuwkCW=ZiC9b8g9UCAkqFvFfWDyA_BywGLa_nJtvXv-2Y_Q@mail.gmail.com>
	<CABj_V-=uB_X77ZUrcYDU_Rx0yJLLwhzMAU3xtfoiOy=PeEwOdA@mail.gmail.com>
	<CALNZuwkszOa6mhUA7UuL7NM9ZHDKSL7NcLxA2p9G0MP6-h_GQQ@mail.gmail.com>
	<CABj_V-mfm79xO6imjmdvMidOqkiwYNxLPTCjJNXcscWTdB0B2g@mail.gmail.com>
	<CAFiqYunySoaNVp6Fb2Sg6YDLATLx7wh9yBW=RXA5Jiqyb_VyZA@mail.gmail.com>
	<CABj_V-=-Mx3OmBeKfrJWjU_yF3aAgN9REjxmP7oR0hReQvzSSQ@mail.gmail.com>
	<CAKzWOxVwhUzjn32J4TRkvAtxhrOObbgoPb46NYk3D-4mra4dBg@mail.gmail.com>
Message-ID: <CABj_V-kr=aaS8--ucc9O1ooHW0SorBZ96QcmdZJnawvzS=qB=A@mail.gmail.com>

On Fri, Aug 5, 2011 at 2:13 AM, Michael Collins <msc at freeswitch.org> wrote:
>
> To help connect the dots I added some links to/from that Lua wiki page
> and channel_in_hangup_hook and also api_hangup_hook and
> reporting_hangup_hook.

That "channel_in_hangup_hook" sounds promising to me, although I can't
find any mention of it in the wiki:

http://wiki.freeswitch.org/wiki/Special:Search?search=channel_in_hangup_hook&go=Go

Regards.

-- 
Isaac Jurado
Internet Busines Solutions eConcept


From steveayre at gmail.com  Fri Aug  5 16:01:02 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 5 Aug 2011 13:01:02 +0100
Subject: [Freeswitch-users] Does FreeSWITCH supports Googles VP8 (WebM)
	Codec
In-Reply-To: <4E3A710F.3000003@gmx.de>
References: <4E3A710F.3000003@gmx.de>
Message-ID: <CAFiqYu=GHO+7ZrSqcKCLxiAXRPYv9=nmO8V-dX36JdnEXwj+RA@mail.gmail.com>

FS supports video, but it's limited support. All video codecs are
passthrough only, with no transcoding.

I'm not sure whether any module currently provides a passthrough VP8
implementation. If there isn't though then proxy_media (
http://wiki.freeswitch.org/wiki/Proxy_Media) will still let you support that
codec. It requires both endpoints support VP8 and at least one audio codec
though, and you'll lose some FS functionality (no accessing the audio to do
things like record, eavesdrop, inband DTMF detection).

-Steve


On 4 August 2011 11:14, Markus Siebeneicher <7eicher at gmx.de> wrote:

> Hello folks from FreeSWITCH,
>
> i am using FreeSWITCH with the conference system BigBlueButton, made by
> people from canada.
>
> It works pretty fine, so far. Due to the need that we would like to
> support more than 500 users in a single video/audio session, we are in
> interest to use the best codecs.
>
> So i listen to VP8 from Google. Does FreeSWITCH support it anyway? I
> couldn't find something in the FAQ's. I listen something about ffmeg, is
> it in this abstraction module?
>
> Thanks for a short reply and thanks for the good work!
>
> Kind regards,
>
> Markus
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110805/b12c206b/attachment.html 

From fieldpeak at gmail.com  Fri Aug  5 17:21:12 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Fri, 5 Aug 2011 21:21:12 +0800
Subject: [Freeswitch-users] Mod_rad_auth issue for FS working with
 FreeRadius server
In-Reply-To: <CAOTmbMxW=EDr3-Ayo2R_MsCCTUZ4PnDkkRUuR2FisH9tK5xkWA@mail.gmail.com>
References: <CAOTmbMzvd5OHL8z=_8vu6Orp2_prGxaw1gmXYCM2=2+pgzs_WA@mail.gmail.com>
	<CAJoKKdX01Qc8ibs25PUELi8GrzMDNOrHb_bQvYdq3ZqA4XLy4w@mail.gmail.com>
	<CAOTmbMxL3pLOuw655fyT5VzAON_N0PrDN+-i-++=oZoYU6+Fmg@mail.gmail.com>
	<CAOTmbMyvyDA-CRpaMC=WRGVkiyTG=Rs+n4hyeQWdyTZpqysSzw@mail.gmail.com>
	<CAJoKKdXSwF02S9tBkorGwOjj57AtkRpU0c177qXASEpodKVk=w@mail.gmail.com>
	<CAOTmbMzu2qDYh2T24JSpCYn1xH2J4iGiK7UkUB4Kq=-KdcDWGQ@mail.gmail.com>
	<CAJoKKdU66jK3f+mOEV90Y3zrmzpTSdWnwGsWimZxir42khiJ-w@mail.gmail.com>
	<CAOTmbMxW=EDr3-Ayo2R_MsCCTUZ4PnDkkRUuR2FisH9tK5xkWA@mail.gmail.com>
Message-ID: <CAOTmbMyDmeJKsX=B-nTM1a7aaTyHyraGNnTEqB86KJOZDFEXpg@mail.gmail.com>

Hi Tihomir,

the issue was resovled by changing the secret of NAS client.
very appreciated for your dedecated help :)

Regards,
Charels

2011/8/5 fieldpeak <fieldpeak at gmail.com>

> Hi Tihomir,
>
> Thanks for your advise, i've added below to rad_auth.conf.xml (vsas
> section), as well as tried auth-type to 0(local) and 1(system), however, the
> issue still exist.
>
>
> <param name="NAS-Port-Type" id="61" value="0" pec="0" expr="0"
> direction="in"/>
> <param name="Login-User" id="6" value="1" pec="0" expr="0" direction="in"/>
> <param name="Auth-Type" id="1000" value="0" pec="0" expr="0"
> direction="in"/>
>
> FreeRadius output:
>
> Found Auth-Type = PAP
> # Executing group from file /usr/local/etc/raddb/sites-enabled/default
> +- entering group PAP {...}
> [pap] login attempt with password "Q?????? ??????p???F?+??a"
> [pap] Using clear text password "1111"
> [pap] Passwords don't match
> ++[pap] returns reject
> Failed to authenticate the user.
>   WARNING: Unprintable characters in the password.        Double-check the shared secret on the server and the NAS!
>
> Using Post-Auth-Type Reject
> # Executing group from file /usr/local/etc/raddb/sites-enabled/default
> +- entering group REJECT {...}
> [attr_filter.access_reject]     expand: %{User-Name} -> 1001
>  attr_filter: Matched entry DEFAULT at line 11
> ++[attr_filter.access_reject] returns updated
> Delaying reject of request 38 for 1 seconds
>
> Regards,
> Charles
>
>
> 2011/8/5 Tihomir Culjaga <tculjaga at gmail.com>
>
>> add to rad_auth.conf.xml
>>
>> <param name="NAS-Port-Type" id="61" value="0" pec="0" expr="0"
>> direction="in"/>
>> <param name="Login-User" id="6" value="1" pec="0" expr="0"
>> direction="in"/>
>>
>>
>>
>> as for Auth Type im not sure if you need it ... this is up to your server.
>> According to dictionary file you need to set it as follows:
>>
>> <param name="Auth-Type" id="1000" value="?" pec="0" expr="0"
>> direction="in"/>
>>
>> the value (set as ?) is one of the folowing. Again, not sure what is
>> required by your server.
>>
>> VALUE           Auth-Type               Local                   0
>> VALUE           Auth-Type               System                  1
>> VALUE           Auth-Type               SecurID                 2
>> VALUE           Auth-Type               Crypt-Local             3
>> VALUE           Auth-Type               Reject                  4
>>
>> #
>> #       Cistron extensions
>> #
>> VALUE           Auth-Type               Pam                     253
>> VALUE           Auth-Type               Accept                  254
>>
>>
>>
>> regards,
>> Tihomir.
>>
>>
>>
>> On Wed, Aug 3, 2011 at 6:32 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>>
>>> Hi Tihomir,
>>>
>>> Sorry, i missed your mail in gmail before, just now saw it, and after
>>> using your dictionary.all, the dictionary issue was resolved, very
>>> appreciated for your kindly help! however, it did not fully functional yet,
>>>
>>> Attached are configuration files that i used, when i dial 601 to trigger
>>> to auth, the freeradius server shows log below, the supecious log is the
>>> value User-Password, it should be '1111' that i've set in the mysql db of
>>> freeradisu server for the user 1001 .
>>>
>>> i searched in google, for "known good" password issue, i suggest change
>>> user-password to cleartext-password, however, i did not find where it is.
>>> and also the Auth-Type, where to configure it...
>>>
>>> Freeradius server log:
>>>
>>> rad_recv: Access-Request packet from host 127.0.0.1 port 52684, id=49,
>>> length=111
>>>         User-Name = "1001"
>>>         User-Password = "?\210\365@\263\t\306\343\243iT?\311C\t\002"
>>>         Called-Station-Id = "888"
>>>         h323-conf-id = "749d2b5a-16ad-48e4-af58-24011949d1b5"
>>>         Calling-Station-Id = "1001"
>>>         NAS-Port = 0
>>>         NAS-IP-Address = 127.0.0.1
>>> # Executing section authorize from file
>>> /usr/local/etc/raddb/sites-enabled/default
>>> +- entering group authorize {...}
>>> ++[preprocess] returns ok
>>> [auth_log]      expand:
>>> /usr/local/var/log/radius/radacct/%{Client-IP-Address}/auth-detail-%Y%m%d ->
>>> /usr/local/var/log/radius/radacct/127.0.0.1/auth-detail-20110803
>>> [auth_log]
>>> /usr/local/var/log/radius/radacct/%{Client-IP-Address}/auth-detail-%Y%m%d
>>> expands to /usr/local/var/log/radius/radacct/
>>> 127.0.0.1/auth-detail-20110803
>>> [auth_log]      expand: %t -> Wed Aug  3 12:06:33 2011
>>> ++[auth_log] returns ok
>>> ++[chap] returns noop
>>> ++[mschap] returns noop
>>> ++[digest] returns noop
>>> [suffix] No '@' in User-Name = "1001", looking up realm NULL
>>> [suffix] No such realm "NULL"
>>> ++[suffix] returns noop
>>> [eap] No EAP-Message, not doing EAP
>>> ++[eap] returns noop
>>> ++[unix] returns notfound
>>> ++[files] returns noop
>>> [sql]   expand: %{User-Name} -> 1001
>>> [sql] sql_set_user escaped user --> '1001'
>>> rlm_sql (sql): Reserving sql socket id: 4
>>> [sql]   expand: SELECT id, username, attribute, value, op           FROM
>>> radcheck           WHERE username = '%{SQL-User-Name}'           ORDER BY id
>>> -> SELECT id, username, attribute, value, op           FROM
>>> radcheck           WHERE username = '1001'           ORDER BY id
>>> [sql]   expand: SELECT groupname           FROM radusergroup
>>> WHERE username = '%{SQL-User-Name}'           ORDER BY priority -> SELECT
>>> groupname           FROM radusergroup           WHERE username =
>>> '1001'           ORDER BY priority
>>> rlm_sql (sql): Released sql socket id: 4
>>> [sql] User 1001 not found
>>> ++[sql] returns notfound
>>> ++[expiration] returns noop
>>> ++[logintime] returns noop
>>> [pap] WARNING! No "known good" password found for the user.
>>> Authentication may fail because of this.
>>> ++[pap] returns noop
>>> ERROR: No authenticate method (Auth-Type) found for the request:
>>> Rejecting the user
>>> Failed to authenticate the user.
>>>   WARNING: Unprintable characters in the password.        Double-check
>>> the shared secret on the server and the NAS!
>>> Using Post-Auth-Type Reject
>>> # Executing group from file /usr/local/etc/raddb/sites-enabled/default
>>> +- entering group REJECT {...}
>>> [attr_filter.access_reject]     expand: %{User-Name} -> 1001
>>>  attr_filter: Matched entry DEFAULT at line 11
>>> ++[attr_filter.access_reject] returns updated
>>> Delaying reject of request 8 for 1 seconds
>>> Going to the next request
>>> Waking up in 0.9 seconds.
>>> Sending delayed reject for request 8
>>> Sending Access-Reject of id 49 to 127.0.0.1 port 52684
>>> Waking up in 4.9 seconds.
>>> Cleaning up request 8 ID 49 with timestamp +7674
>>> Ready to process requests.
>>> WARNING! No "known good" password found for the user
>>>
>>> Regards,
>>> Charles
>>>
>>>
>>> 2011/8/3 Tihomir Culjaga <tculjaga at gmail.com>
>>>
>>>> did u use the dictionary i have attached ?
>>>>
>>>>
>>>> On Tue, Aug 2, 2011 at 10:08 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>>>>
>>>>> i tried change to 'h323-conf-id' to 'h323-call-origin' in
>>>>> 02_unitest_rad-ANI-auth.xml, rad_auth.conf.xml, however, it still prompt
>>>>> '[ERR] mod_rad_auth.c:428 Unknown attribute: key:h323-conf-id, not
>>>>> found in dictionary', so where the mod_rad_auth read out the 'h323-conf-id'?
>>>>> very very strange, which dictionary it was using...
>>>>>
>>>>> Regards,
>>>>> Charles
>>>>>
>>>>>
>>>>> 2011/8/2 fieldpeak <fieldpeak at gmail.com>
>>>>>
>>>>>> Hi Tihomir,
>>>>>>
>>>>>> Finally the answer coming, i see the hope, thanks for your reply, :)
>>>>>>
>>>>>> As your advise, i only use one attribute(h323-conf-id) in my dialplan,
>>>>>> and only one attribute(h323-conf-id) in rad_auth.conf.xml, and using the
>>>>>> attached dictionary (from ciso) which contains this attribute, however, it
>>>>>> still prompt 'unknown attribute', so i suspected if it was reading
>>>>>> /usr/local/etc/radiusclient/dictionary, so i copy the same dictionary to
>>>>>> /usr/local/freeswitch/radius/, it did not any help at all... very strange...
>>>>>>
>>>>>> Log:
>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set
>>>>>> default_realm := .
>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set
>>>>>> radius_timeout := 3.
>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set
>>>>>> radius_retries := 2.
>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set
>>>>>> radius_deadtime := 0.
>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set bindaddr :=
>>>>>> *.
>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:371 ... radius:
>>>>>> User-Name: 38516060333
>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:380 ... radius:
>>>>>> User-Password: 003282
>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:396 ... radius:
>>>>>> Called-station-Id: 16094191500
>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:413 Handle
>>>>>> attribute: h323-conf-id
>>>>>> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:428 Unknown attribute:
>>>>>> key:h323-conf-id, not found in dictionary
>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:538 abort sending
>>>>>> radius packet.
>>>>>> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:546 An error occured
>>>>>> during RADIUS Authentication(RC=-1)
>>>>>> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:702 An error occured
>>>>>> during radius authorization.
>>>>>>
>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  AUTH_RESULT=)
>>>>>>
>>>>>>
>>>>>>
>>>>>>   <extension name="unitest_rad-ANI-auth">
>>>>>>     <condition field="destination_number" expression="^601$">
>>>>>>       <!-- <action application="log" data="INFO  Before Auth "/> -->
>>>>>>
>>>>>>       <action inline="true" application="set" data="CALLID=
>>>>>> h323-conf-id=${uuid}"/>
>>>>>>
>>>>>>       <action inline="true" application="set" data="USERNAME=1001"/>
>>>>>>       <action inline="true" application="set" data="PASSWD=1111"/>
>>>>>>
>>>>>>
>>>>>>       <action application="sleep" data="2000"/>
>>>>>>       <action application="auth_function" data="in ${DIALED_NUMBER},
>>>>>> in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>
>>>>>>
>>>>>>     </condition>
>>>>>>   </extension>
>>>>>>
>>>>>>
>>>>>>
>>>>>> <configuration name="rad_auth.conf" description="radius
>>>>>> authentification module">
>>>>>>   <settings>
>>>>>>
>>>>>>   </settings>
>>>>>>
>>>>>>   <client>
>>>>>>     <param name="authserver" value="127.0.0.1:1812:gateway"/>
>>>>>>     <param name="dictionary"
>>>>>> value="/usr/local/etc/radiusclient/dictionary"/>
>>>>>>     <param name="seqfile" value="/var/run/radius.seq"/>
>>>>>>     <param name="mapfile"
>>>>>> value="/usr/local/etc/radiusclient/port-id-map"/>
>>>>>>     <param name="default_realm" value=""/>
>>>>>>     <param name="radius_timeout" value="3"/>
>>>>>>     <param name="radius_retries" value="2"/>
>>>>>>     <param name="radius_deadtime" value="0"/>
>>>>>>     <param name="bindaddr" value="*"/>
>>>>>>   </client>
>>>>>>
>>>>>>   <vsas>
>>>>>>
>>>>>>
>>>>>>     <param name="h323-conf-id" id="24" value="CALLID" pec="9"
>>>>>> expr="1" direction="in"/>
>>>>>>
>>>>>>   </vsas>
>>>>>>  </configuration>
>>>>>>
>>>>>>
>>>>>>
>>>>>> 2011/8/2 Tihomir Culjaga <tculjaga at gmail.com>
>>>>>>
>>>>>>> hi,
>>>>>>>
>>>>>>> dictionary.all is just the name of a file containing all attributes i
>>>>>>> needed at that time.
>>>>>>>
>>>>>>> you can include other dictionaries by putting #INCLUDE <pathname> at
>>>>>>> the end of the dictionary file you reference in rad_auth.conf.xml.
>>>>>>> if the INCLUDE doesn't work, just append dictionary.cisco to your
>>>>>>> dictionary file... and make your own file.
>>>>>>>
>>>>>>>
>>>>>>> check inline comments down below...
>>>>>>>
>>>>>>>
>>>>>>> T.
>>>>>>>
>>>>>>>
>>>>>>> On Sun, Jul 31, 2011 at 10:46 AM, fieldpeak <fieldpeak at gmail.com>wrote:
>>>>>>>
>>>>>>>> Hello Gurus,
>>>>>>>>
>>>>>>>> i met a issue when using
>>>>>>>> mod_rad_auth(http://wiki.freeswitch.org/wiki/Mod_rad_auth) to works
>>>>>>>> with freeradius server+mysql for AAA, the details is below, Could
>>>>>>>> anyone give any hints, Thanks in advance.
>>>>>>>>
>>>>>>>> i setup a dial plan "unitest_rad-ANI-auth" as wiki above, however,
>>>>>>>> when i dialed 601 to trigger the dial plan, the console show errors,
>>>>>>>> it looks "h323-conf-id" is not in the directory, then i tried to add
>>>>>>>> this attribute to the dictionary, however, it does not help, in the
>>>>>>>> wiki, it mentioned the rad_auth.conf.xml contains <param
>>>>>>>> name="dictionary"
>>>>>>>> value="/usr/local/etc/radiusclient/dictionary.all"/>, however i did
>>>>>>>> not find the file "dictionary.all" at that directory, so i use
>>>>>>>> dictionary. BTW, the freeradius server + mysql works well.
>>>>>>>>
>>>>>>>
>>>>>>> i just appended the information needed into dictionary.all file...
>>>>>>> (vendor and attribute definition).
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>>
>>>>>>>> console errors:
>>>>>>>>
>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 auth_function(in , in
>>>>>>>> 38516060333, in 003282, out AUTH_RESULT)
>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:301 allocate
>>>>>>>> initial
>>>>>>>> structure.
>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:313 initialzed
>>>>>>>> configuration.
>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set authserver
>>>>>>>> := 127.0.0.1:1812:gateway.
>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set dictionary
>>>>>>>> := /usr/local/etc/radiusclient/dictionary.
>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set seqfile :=
>>>>>>>> /var/run/radius.seq.
>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set mapfile :=
>>>>>>>> /usr/local/etc/radiusclient/port-id-map.
>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>> default_realm := .
>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>> radius_timeout := 3.
>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>> radius_retries := 2.
>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>> radius_deadtime := 0.
>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set bindaddr
>>>>>>>> := *.
>>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:371 ... radius:
>>>>>>>> User-Name: 38516060333
>>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:380 ... radius:
>>>>>>>> User-Password: 003282
>>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:391 ... radius:
>>>>>>>> Called-station-Id is empty, ignoring...
>>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:413 Handle
>>>>>>>> attribute: h323-conf-id
>>>>>>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:428 Unknown
>>>>>>>> attribute:
>>>>>>>> key:h323-conf-id, not found in dictionary
>>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:538 abort sending
>>>>>>>> radius packet.
>>>>>>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:546 An error occured
>>>>>>>> during RADIUS Authentication(RC=-1)
>>>>>>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:702 An error occured
>>>>>>>> during radius authorization.
>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  AUTH_RESULT=)
>>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  AUTH_RESULT=
>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO
>>>>>>>>  billing_model=)
>>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  billing_model=
>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO
>>>>>>>>  credit_amount=)
>>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  credit_amount=
>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  currency=)
>>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  currency=
>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO
>>>>>>>>  preffered_lang=)
>>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202
>>>>>>>>  preffered_lang=
>>>>>>>>
>>>>>>>> added below in the
>>>>>>>> dictionary(/usr/local/etc/radiusclient/dictionary):
>>>>>>>>
>>>>>>>> ATTRIBUTE       h323-conf-id            1008    string
>>>>>>>>
>>>>>>>
>>>>>>> you need the vendor definition as well
>>>>>>>
>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> dial plan:
>>>>>>>> <extension name="unitest_rad-ANI-auth">
>>>>>>>>    <condition field="destination_number" expression="^601$">
>>>>>>>>      <action application="log" data="INFO  Before Auth "/>
>>>>>>>>
>>>>>>>>      <action inline="true" application="set"
>>>>>>>> data="CALLID=h323-conf-id=${uuid}"/>
>>>>>>>>      <action inline="true" application="set"
>>>>>>>> data="SERVICENUM=h323-prompt-id=${destination_number}"/>
>>>>>>>>      <action inline="true" application="set"
>>>>>>>> data="TRANSACTIONID=h323-ivr-out=transactionID:1234"/>
>>>>>>>>  <!--      <action inline="true" application="set"
>>>>>>>> data="CALLINGNUMBER=${caller_id_number}"/> -->
>>>>>>>>      <action inline="true" application="set"
>>>>>>>> data="CALLINGNUMBER=38516060333"/>
>>>>>>>>      <action inline="true" application="set"
>>>>>>>> data="USERNAME=38516060333"/>
>>>>>>>>  <!--      <action inline="true" application="set"
>>>>>>>> data="USERNAME=209354"/> -->
>>>>>>>>      <action inline="true" application="set" data="PASSWD=003282"/>
>>>>>>>>  <!--      <action inline="true" application="set"
>>>>>>>> data="DIALED_NUMBER=16094191500"/>  -->
>>>>>>>>
>>>>>>>>      <action application="sleep" data="2000"/>
>>>>>>>>      <action application="auth_function" data="in ${DIALED_NUMBER},
>>>>>>>> in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>
>>>>>>>>
>>>>>>>>
>>>>>>>>      <action application="log" data="INFO
>>>>>>>>  AUTH_RESULT=${AUTH_RESULT}"/>
>>>>>>>>      <action application="log" data="INFO
>>>>>>>>  billing_model=${billing_model}"/>
>>>>>>>>      <action application="log" data="INFO
>>>>>>>>  credit_amount=${credit_amount}"/>
>>>>>>>>      <action application="log" data="INFO  currency=${currency}"/>
>>>>>>>>      <action application="log" data="INFO
>>>>>>>>  preffered_lang=${preffered_lang}"/>
>>>>>>>>      <action application="log" data="INFO
>>>>>>>>  credit_time=${credit_time}"/>
>>>>>>>>      <action application="log" data="INFO
>>>>>>>> h323_ivr_duration=${h323_ivr_duration}"/>
>>>>>>>>      <action application="log" data="INFO
>>>>>>>>  return_code=${return_code}"/>
>>>>>>>>      <!-- <action application="execute_extension" data="AUTH XML
>>>>>>>> default"/> -->
>>>>>>>>    </condition>
>>>>>>>>  </extension>
>>>>>>>>
>>>>>>>>  radius_cdr.conf.xml:
>>>>>>>>  <configuration name="radius_cdr.conf" description="RADIUS CDR
>>>>>>>> Configuration">
>>>>>>>>
>>>>>>>>        <settings>
>>>>>>>>
>>>>>>>>                <!-- location of the radius dictionary files -->
>>>>>>>>
>>>>>>>>                <param name="dictionary"
>>>>>>>> value="/usr/local/freeswitch/conf/radius/dictionary"/>
>>>>>>>>
>>>>>>>>
>>>>>>> your dictionary file need to contain all the attributes you are
>>>>>>> trying to use or to include other dictionaries (In this case
>>>>>>> dictionary.cisco) from the dictionary file you are referencing here.
>>>>>>>
>>>>>>>
>>>>>>>>                <!-- number of retries for each server -->
>>>>>>>>
>>>>>>>>                <param name="radius_retries" value="3"/>
>>>>>>>>
>>>>>>>>                <!-- number of seconds to wait between retries -->
>>>>>>>>
>>>>>>>>                <param name="radius_timeout" value="5"/>
>>>>>>>>
>>>>>>>>                <!-- accounting servers, up to 8 allowed -->
>>>>>>>>
>>>>>>>>                <!-- value is "host:port:secret", port is optional
>>>>>>>> -->
>>>>>>>>
>>>>>>>>                <!-- use IP ADDRESSES, not hostnames -->
>>>>>>>>
>>>>>>>>                <param name="acct_server" value="127.0.0.1:1813
>>>>>>>> :testing123"/>
>>>>>>>>
>>>>>>>>
>>>>>>>>        </settings>
>>>>>>>>
>>>>>>>> </configuration>
>>>>>>>>
>>>>>>>>  the FS version:
>>>>>>>>  FreeSWITCH Version 1.0.head (git-492bc6b 2011-07-23 12-53-04 -0400)
>>>>>>>>
>>>>>>>>  Regards,
>>>>>>>>  Charles
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110805/548a1746/attachment-0001.html 

From fieldpeak at gmail.com  Fri Aug  5 17:59:56 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Fri, 5 Aug 2011 21:59:56 +0800
Subject: [Freeswitch-users] Mod_rad_auth issue for FS working with
 FreeRadius server
In-Reply-To: <CAOTmbMyDmeJKsX=B-nTM1a7aaTyHyraGNnTEqB86KJOZDFEXpg@mail.gmail.com>
References: <CAOTmbMzvd5OHL8z=_8vu6Orp2_prGxaw1gmXYCM2=2+pgzs_WA@mail.gmail.com>
	<CAJoKKdX01Qc8ibs25PUELi8GrzMDNOrHb_bQvYdq3ZqA4XLy4w@mail.gmail.com>
	<CAOTmbMxL3pLOuw655fyT5VzAON_N0PrDN+-i-++=oZoYU6+Fmg@mail.gmail.com>
	<CAOTmbMyvyDA-CRpaMC=WRGVkiyTG=Rs+n4hyeQWdyTZpqysSzw@mail.gmail.com>
	<CAJoKKdXSwF02S9tBkorGwOjj57AtkRpU0c177qXASEpodKVk=w@mail.gmail.com>
	<CAOTmbMzu2qDYh2T24JSpCYn1xH2J4iGiK7UkUB4Kq=-KdcDWGQ@mail.gmail.com>
	<CAJoKKdU66jK3f+mOEV90Y3zrmzpTSdWnwGsWimZxir42khiJ-w@mail.gmail.com>
	<CAOTmbMxW=EDr3-Ayo2R_MsCCTUZ4PnDkkRUuR2FisH9tK5xkWA@mail.gmail.com>
	<CAOTmbMyDmeJKsX=B-nTM1a7aaTyHyraGNnTEqB86KJOZDFEXpg@mail.gmail.com>
Message-ID: <CAOTmbMz5+_xxpY1S2BSa7g54_HMCvOxL+ue3crFDH60Epz+VMQ@mail.gmail.com>

Hi Tihomir,

I found there are additional attribuites within the response of FreeRadius,
e.g. credit_amount, credit_time etc. in the wiki (
http://wiki.freeswitch.org/wiki/Mod_rad_auth), i belive you added those
fields in the database,  can you please
share the DB schema of freeradius? It will definetely help a lot, Thanks in
advance.

Regards,
Charles

2011/8/5 fieldpeak <fieldpeak at gmail.com>

> Hi Tihomir,
>
> the issue was resovled by changing the secret of NAS client.
> very appreciated for your dedecated help :)
>
> Regards,
> Charels
>
>
> 2011/8/5 fieldpeak <fieldpeak at gmail.com>
>
>> Hi Tihomir,
>>
>> Thanks for your advise, i've added below to rad_auth.conf.xml (vsas
>> section), as well as tried auth-type to 0(local) and 1(system), however, the
>> issue still exist.
>>
>>
>> <param name="NAS-Port-Type" id="61" value="0" pec="0" expr="0"
>> direction="in"/>
>> <param name="Login-User" id="6" value="1" pec="0" expr="0"
>> direction="in"/>
>> <param name="Auth-Type" id="1000" value="0" pec="0" expr="0"
>> direction="in"/>
>>
>> FreeRadius output:
>>
>> Found Auth-Type = PAP
>> # Executing group from file /usr/local/etc/raddb/sites-enabled/default
>> +- entering group PAP {...}
>> [pap] login attempt with password "Q?????? ??????p???F?+??a"
>> [pap] Using clear text password "1111"
>> [pap] Passwords don't match
>> ++[pap] returns reject
>> Failed to authenticate the user.
>>   WARNING: Unprintable characters in the password.        Double-check the shared secret on the server and the NAS!
>>
>> Using Post-Auth-Type Reject
>> # Executing group from file /usr/local/etc/raddb/sites-enabled/default
>> +- entering group REJECT {...}
>> [attr_filter.access_reject]     expand: %{User-Name} -> 1001
>>  attr_filter: Matched entry DEFAULT at line 11
>> ++[attr_filter.access_reject] returns updated
>> Delaying reject of request 38 for 1 seconds
>>
>> Regards,
>> Charles
>>
>>
>> 2011/8/5 Tihomir Culjaga <tculjaga at gmail.com>
>>
>>> add to rad_auth.conf.xml
>>>
>>> <param name="NAS-Port-Type" id="61" value="0" pec="0" expr="0"
>>> direction="in"/>
>>> <param name="Login-User" id="6" value="1" pec="0" expr="0"
>>> direction="in"/>
>>>
>>>
>>>
>>> as for Auth Type im not sure if you need it ... this is up to your
>>> server.
>>> According to dictionary file you need to set it as follows:
>>>
>>> <param name="Auth-Type" id="1000" value="?" pec="0" expr="0"
>>> direction="in"/>
>>>
>>> the value (set as ?) is one of the folowing. Again, not sure what is
>>> required by your server.
>>>
>>> VALUE           Auth-Type               Local                   0
>>> VALUE           Auth-Type               System                  1
>>> VALUE           Auth-Type               SecurID                 2
>>> VALUE           Auth-Type               Crypt-Local             3
>>> VALUE           Auth-Type               Reject                  4
>>>
>>> #
>>> #       Cistron extensions
>>> #
>>> VALUE           Auth-Type               Pam                     253
>>> VALUE           Auth-Type               Accept                  254
>>>
>>>
>>>
>>> regards,
>>> Tihomir.
>>>
>>>
>>>
>>> On Wed, Aug 3, 2011 at 6:32 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>>>
>>>> Hi Tihomir,
>>>>
>>>> Sorry, i missed your mail in gmail before, just now saw it, and after
>>>> using your dictionary.all, the dictionary issue was resolved, very
>>>> appreciated for your kindly help! however, it did not fully functional yet,
>>>>
>>>> Attached are configuration files that i used, when i dial 601 to trigger
>>>> to auth, the freeradius server shows log below, the supecious log is the
>>>> value User-Password, it should be '1111' that i've set in the mysql db of
>>>> freeradisu server for the user 1001 .
>>>>
>>>> i searched in google, for "known good" password issue, i suggest change
>>>> user-password to cleartext-password, however, i did not find where it is.
>>>> and also the Auth-Type, where to configure it...
>>>>
>>>> Freeradius server log:
>>>>
>>>> rad_recv: Access-Request packet from host 127.0.0.1 port 52684, id=49,
>>>> length=111
>>>>         User-Name = "1001"
>>>>         User-Password = "?\210\365@\263\t\306\343\243iT?\311C\t\002"
>>>>         Called-Station-Id = "888"
>>>>         h323-conf-id = "749d2b5a-16ad-48e4-af58-24011949d1b5"
>>>>         Calling-Station-Id = "1001"
>>>>         NAS-Port = 0
>>>>         NAS-IP-Address = 127.0.0.1
>>>> # Executing section authorize from file
>>>> /usr/local/etc/raddb/sites-enabled/default
>>>> +- entering group authorize {...}
>>>> ++[preprocess] returns ok
>>>> [auth_log]      expand:
>>>> /usr/local/var/log/radius/radacct/%{Client-IP-Address}/auth-detail-%Y%m%d ->
>>>> /usr/local/var/log/radius/radacct/127.0.0.1/auth-detail-20110803
>>>> [auth_log]
>>>> /usr/local/var/log/radius/radacct/%{Client-IP-Address}/auth-detail-%Y%m%d
>>>> expands to /usr/local/var/log/radius/radacct/
>>>> 127.0.0.1/auth-detail-20110803
>>>> [auth_log]      expand: %t -> Wed Aug  3 12:06:33 2011
>>>> ++[auth_log] returns ok
>>>> ++[chap] returns noop
>>>> ++[mschap] returns noop
>>>> ++[digest] returns noop
>>>> [suffix] No '@' in User-Name = "1001", looking up realm NULL
>>>> [suffix] No such realm "NULL"
>>>> ++[suffix] returns noop
>>>> [eap] No EAP-Message, not doing EAP
>>>> ++[eap] returns noop
>>>> ++[unix] returns notfound
>>>> ++[files] returns noop
>>>> [sql]   expand: %{User-Name} -> 1001
>>>> [sql] sql_set_user escaped user --> '1001'
>>>> rlm_sql (sql): Reserving sql socket id: 4
>>>> [sql]   expand: SELECT id, username, attribute, value, op           FROM
>>>> radcheck           WHERE username = '%{SQL-User-Name}'           ORDER BY id
>>>> -> SELECT id, username, attribute, value, op           FROM
>>>> radcheck           WHERE username = '1001'           ORDER BY id
>>>> [sql]   expand: SELECT groupname           FROM radusergroup
>>>> WHERE username = '%{SQL-User-Name}'           ORDER BY priority -> SELECT
>>>> groupname           FROM radusergroup           WHERE username =
>>>> '1001'           ORDER BY priority
>>>> rlm_sql (sql): Released sql socket id: 4
>>>> [sql] User 1001 not found
>>>> ++[sql] returns notfound
>>>> ++[expiration] returns noop
>>>> ++[logintime] returns noop
>>>> [pap] WARNING! No "known good" password found for the user.
>>>> Authentication may fail because of this.
>>>> ++[pap] returns noop
>>>> ERROR: No authenticate method (Auth-Type) found for the request:
>>>> Rejecting the user
>>>> Failed to authenticate the user.
>>>>   WARNING: Unprintable characters in the password.        Double-check
>>>> the shared secret on the server and the NAS!
>>>> Using Post-Auth-Type Reject
>>>> # Executing group from file /usr/local/etc/raddb/sites-enabled/default
>>>> +- entering group REJECT {...}
>>>> [attr_filter.access_reject]     expand: %{User-Name} -> 1001
>>>>  attr_filter: Matched entry DEFAULT at line 11
>>>> ++[attr_filter.access_reject] returns updated
>>>> Delaying reject of request 8 for 1 seconds
>>>> Going to the next request
>>>> Waking up in 0.9 seconds.
>>>> Sending delayed reject for request 8
>>>> Sending Access-Reject of id 49 to 127.0.0.1 port 52684
>>>> Waking up in 4.9 seconds.
>>>> Cleaning up request 8 ID 49 with timestamp +7674
>>>> Ready to process requests.
>>>> WARNING! No "known good" password found for the user
>>>>
>>>> Regards,
>>>> Charles
>>>>
>>>>
>>>> 2011/8/3 Tihomir Culjaga <tculjaga at gmail.com>
>>>>
>>>>> did u use the dictionary i have attached ?
>>>>>
>>>>>
>>>>> On Tue, Aug 2, 2011 at 10:08 AM, fieldpeak <fieldpeak at gmail.com>wrote:
>>>>>
>>>>>> i tried change to 'h323-conf-id' to 'h323-call-origin' in
>>>>>> 02_unitest_rad-ANI-auth.xml, rad_auth.conf.xml, however, it still prompt
>>>>>> '[ERR] mod_rad_auth.c:428 Unknown attribute: key:h323-conf-id, not
>>>>>> found in dictionary', so where the mod_rad_auth read out the 'h323-conf-id'?
>>>>>> very very strange, which dictionary it was using...
>>>>>>
>>>>>> Regards,
>>>>>> Charles
>>>>>>
>>>>>>
>>>>>> 2011/8/2 fieldpeak <fieldpeak at gmail.com>
>>>>>>
>>>>>>> Hi Tihomir,
>>>>>>>
>>>>>>> Finally the answer coming, i see the hope, thanks for your reply, :)
>>>>>>>
>>>>>>> As your advise, i only use one attribute(h323-conf-id) in my
>>>>>>> dialplan, and only one attribute(h323-conf-id) in rad_auth.conf.xml, and
>>>>>>> using the attached dictionary (from ciso) which contains this attribute,
>>>>>>> however, it still prompt 'unknown attribute', so i suspected if it was
>>>>>>> reading /usr/local/etc/radiusclient/dictionary, so i copy the same
>>>>>>> dictionary to /usr/local/freeswitch/radius/, it did not any help at all...
>>>>>>> very strange...
>>>>>>>
>>>>>>> Log:
>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set
>>>>>>> default_realm := .
>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set
>>>>>>> radius_timeout := 3.
>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set
>>>>>>> radius_retries := 2.
>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set
>>>>>>> radius_deadtime := 0.
>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set bindaddr :=
>>>>>>> *.
>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:371 ... radius:
>>>>>>> User-Name: 38516060333
>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:380 ... radius:
>>>>>>> User-Password: 003282
>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:396 ... radius:
>>>>>>> Called-station-Id: 16094191500
>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:413 Handle
>>>>>>> attribute: h323-conf-id
>>>>>>> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:428 Unknown
>>>>>>> attribute: key:h323-conf-id, not found in dictionary
>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:538 abort sending
>>>>>>> radius packet.
>>>>>>> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:546 An error occured
>>>>>>> during RADIUS Authentication(RC=-1)
>>>>>>> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:702 An error occured
>>>>>>> during radius authorization.
>>>>>>>
>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  AUTH_RESULT=)
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>   <extension name="unitest_rad-ANI-auth">
>>>>>>>     <condition field="destination_number" expression="^601$">
>>>>>>>       <!-- <action application="log" data="INFO  Before Auth "/> -->
>>>>>>>
>>>>>>>       <action inline="true" application="set" data="CALLID=
>>>>>>> h323-conf-id=${uuid}"/>
>>>>>>>
>>>>>>>       <action inline="true" application="set" data="USERNAME=1001"/>
>>>>>>>       <action inline="true" application="set" data="PASSWD=1111"/>
>>>>>>>
>>>>>>>
>>>>>>>       <action application="sleep" data="2000"/>
>>>>>>>       <action application="auth_function" data="in ${DIALED_NUMBER},
>>>>>>> in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>
>>>>>>>
>>>>>>>     </condition>
>>>>>>>   </extension>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> <configuration name="rad_auth.conf" description="radius
>>>>>>> authentification module">
>>>>>>>   <settings>
>>>>>>>
>>>>>>>   </settings>
>>>>>>>
>>>>>>>   <client>
>>>>>>>     <param name="authserver" value="127.0.0.1:1812:gateway"/>
>>>>>>>     <param name="dictionary"
>>>>>>> value="/usr/local/etc/radiusclient/dictionary"/>
>>>>>>>     <param name="seqfile" value="/var/run/radius.seq"/>
>>>>>>>     <param name="mapfile"
>>>>>>> value="/usr/local/etc/radiusclient/port-id-map"/>
>>>>>>>     <param name="default_realm" value=""/>
>>>>>>>     <param name="radius_timeout" value="3"/>
>>>>>>>     <param name="radius_retries" value="2"/>
>>>>>>>     <param name="radius_deadtime" value="0"/>
>>>>>>>     <param name="bindaddr" value="*"/>
>>>>>>>   </client>
>>>>>>>
>>>>>>>   <vsas>
>>>>>>>
>>>>>>>
>>>>>>>     <param name="h323-conf-id" id="24" value="CALLID" pec="9"
>>>>>>> expr="1" direction="in"/>
>>>>>>>
>>>>>>>   </vsas>
>>>>>>>  </configuration>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> 2011/8/2 Tihomir Culjaga <tculjaga at gmail.com>
>>>>>>>
>>>>>>>> hi,
>>>>>>>>
>>>>>>>> dictionary.all is just the name of a file containing all attributes
>>>>>>>> i needed at that time.
>>>>>>>>
>>>>>>>> you can include other dictionaries by putting #INCLUDE <pathname> at
>>>>>>>> the end of the dictionary file you reference in rad_auth.conf.xml.
>>>>>>>> if the INCLUDE doesn't work, just append dictionary.cisco to your
>>>>>>>> dictionary file... and make your own file.
>>>>>>>>
>>>>>>>>
>>>>>>>> check inline comments down below...
>>>>>>>>
>>>>>>>>
>>>>>>>> T.
>>>>>>>>
>>>>>>>>
>>>>>>>> On Sun, Jul 31, 2011 at 10:46 AM, fieldpeak <fieldpeak at gmail.com>wrote:
>>>>>>>>
>>>>>>>>> Hello Gurus,
>>>>>>>>>
>>>>>>>>> i met a issue when using
>>>>>>>>> mod_rad_auth(http://wiki.freeswitch.org/wiki/Mod_rad_auth) to
>>>>>>>>> works
>>>>>>>>> with freeradius server+mysql for AAA, the details is below, Could
>>>>>>>>> anyone give any hints, Thanks in advance.
>>>>>>>>>
>>>>>>>>> i setup a dial plan "unitest_rad-ANI-auth" as wiki above, however,
>>>>>>>>> when i dialed 601 to trigger the dial plan, the console show
>>>>>>>>> errors,
>>>>>>>>> it looks "h323-conf-id" is not in the directory, then i tried to
>>>>>>>>> add
>>>>>>>>> this attribute to the dictionary, however, it does not help, in the
>>>>>>>>> wiki, it mentioned the rad_auth.conf.xml contains <param
>>>>>>>>> name="dictionary"
>>>>>>>>> value="/usr/local/etc/radiusclient/dictionary.all"/>, however i did
>>>>>>>>> not find the file "dictionary.all" at that directory, so i use
>>>>>>>>> dictionary. BTW, the freeradius server + mysql works well.
>>>>>>>>>
>>>>>>>>
>>>>>>>> i just appended the information needed into dictionary.all file...
>>>>>>>> (vendor and attribute definition).
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>>
>>>>>>>>> console errors:
>>>>>>>>>
>>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 auth_function(in , in
>>>>>>>>> 38516060333, in 003282, out AUTH_RESULT)
>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:301 allocate
>>>>>>>>> initial
>>>>>>>>> structure.
>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:313 initialzed
>>>>>>>>> configuration.
>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>> authserver
>>>>>>>>> := 127.0.0.1:1812:gateway.
>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>> dictionary
>>>>>>>>> := /usr/local/etc/radiusclient/dictionary.
>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set seqfile
>>>>>>>>> :=
>>>>>>>>> /var/run/radius.seq.
>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set mapfile
>>>>>>>>> :=
>>>>>>>>> /usr/local/etc/radiusclient/port-id-map.
>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>> default_realm := .
>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>> radius_timeout := 3.
>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>> radius_retries := 2.
>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>> radius_deadtime := 0.
>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set bindaddr
>>>>>>>>> := *.
>>>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:371 ... radius:
>>>>>>>>> User-Name: 38516060333
>>>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:380 ... radius:
>>>>>>>>> User-Password: 003282
>>>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:391 ... radius:
>>>>>>>>> Called-station-Id is empty, ignoring...
>>>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:413 Handle
>>>>>>>>> attribute: h323-conf-id
>>>>>>>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:428 Unknown
>>>>>>>>> attribute:
>>>>>>>>> key:h323-conf-id, not found in dictionary
>>>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:538 abort sending
>>>>>>>>> radius packet.
>>>>>>>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:546 An error
>>>>>>>>> occured
>>>>>>>>> during RADIUS Authentication(RC=-1)
>>>>>>>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:702 An error
>>>>>>>>> occured
>>>>>>>>> during radius authorization.
>>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO
>>>>>>>>>  AUTH_RESULT=)
>>>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  AUTH_RESULT=
>>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO
>>>>>>>>>  billing_model=)
>>>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202
>>>>>>>>>  billing_model=
>>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO
>>>>>>>>>  credit_amount=)
>>>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202
>>>>>>>>>  credit_amount=
>>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  currency=)
>>>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  currency=
>>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO
>>>>>>>>>  preffered_lang=)
>>>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202
>>>>>>>>>  preffered_lang=
>>>>>>>>>
>>>>>>>>> added below in the
>>>>>>>>> dictionary(/usr/local/etc/radiusclient/dictionary):
>>>>>>>>>
>>>>>>>>> ATTRIBUTE       h323-conf-id            1008    string
>>>>>>>>>
>>>>>>>>
>>>>>>>> you need the vendor definition as well
>>>>>>>>
>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> dial plan:
>>>>>>>>> <extension name="unitest_rad-ANI-auth">
>>>>>>>>>    <condition field="destination_number" expression="^601$">
>>>>>>>>>      <action application="log" data="INFO  Before Auth "/>
>>>>>>>>>
>>>>>>>>>      <action inline="true" application="set"
>>>>>>>>> data="CALLID=h323-conf-id=${uuid}"/>
>>>>>>>>>      <action inline="true" application="set"
>>>>>>>>> data="SERVICENUM=h323-prompt-id=${destination_number}"/>
>>>>>>>>>      <action inline="true" application="set"
>>>>>>>>> data="TRANSACTIONID=h323-ivr-out=transactionID:1234"/>
>>>>>>>>>  <!--      <action inline="true" application="set"
>>>>>>>>> data="CALLINGNUMBER=${caller_id_number}"/> -->
>>>>>>>>>      <action inline="true" application="set"
>>>>>>>>> data="CALLINGNUMBER=38516060333"/>
>>>>>>>>>      <action inline="true" application="set"
>>>>>>>>> data="USERNAME=38516060333"/>
>>>>>>>>>  <!--      <action inline="true" application="set"
>>>>>>>>> data="USERNAME=209354"/> -->
>>>>>>>>>      <action inline="true" application="set" data="PASSWD=003282"/>
>>>>>>>>>  <!--      <action inline="true" application="set"
>>>>>>>>> data="DIALED_NUMBER=16094191500"/>  -->
>>>>>>>>>
>>>>>>>>>      <action application="sleep" data="2000"/>
>>>>>>>>>      <action application="auth_function" data="in ${DIALED_NUMBER},
>>>>>>>>> in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>>  AUTH_RESULT=${AUTH_RESULT}"/>
>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>>  billing_model=${billing_model}"/>
>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>>  credit_amount=${credit_amount}"/>
>>>>>>>>>      <action application="log" data="INFO  currency=${currency}"/>
>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>>  preffered_lang=${preffered_lang}"/>
>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>>  credit_time=${credit_time}"/>
>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>> h323_ivr_duration=${h323_ivr_duration}"/>
>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>>  return_code=${return_code}"/>
>>>>>>>>>      <!-- <action application="execute_extension" data="AUTH XML
>>>>>>>>> default"/> -->
>>>>>>>>>    </condition>
>>>>>>>>>  </extension>
>>>>>>>>>
>>>>>>>>>  radius_cdr.conf.xml:
>>>>>>>>>  <configuration name="radius_cdr.conf" description="RADIUS CDR
>>>>>>>>> Configuration">
>>>>>>>>>
>>>>>>>>>        <settings>
>>>>>>>>>
>>>>>>>>>                <!-- location of the radius dictionary files -->
>>>>>>>>>
>>>>>>>>>                <param name="dictionary"
>>>>>>>>> value="/usr/local/freeswitch/conf/radius/dictionary"/>
>>>>>>>>>
>>>>>>>>>
>>>>>>>> your dictionary file need to contain all the attributes you are
>>>>>>>> trying to use or to include other dictionaries (In this case
>>>>>>>> dictionary.cisco) from the dictionary file you are referencing here.
>>>>>>>>
>>>>>>>>
>>>>>>>>>                <!-- number of retries for each server -->
>>>>>>>>>
>>>>>>>>>                <param name="radius_retries" value="3"/>
>>>>>>>>>
>>>>>>>>>                <!-- number of seconds to wait between retries -->
>>>>>>>>>
>>>>>>>>>                <param name="radius_timeout" value="5"/>
>>>>>>>>>
>>>>>>>>>                <!-- accounting servers, up to 8 allowed -->
>>>>>>>>>
>>>>>>>>>                <!-- value is "host:port:secret", port is optional
>>>>>>>>> -->
>>>>>>>>>
>>>>>>>>>                <!-- use IP ADDRESSES, not hostnames -->
>>>>>>>>>
>>>>>>>>>                <param name="acct_server" value="127.0.0.1:1813
>>>>>>>>> :testing123"/>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>        </settings>
>>>>>>>>>
>>>>>>>>> </configuration>
>>>>>>>>>
>>>>>>>>>  the FS version:
>>>>>>>>>  FreeSWITCH Version 1.0.head (git-492bc6b 2011-07-23 12-53-04
>>>>>>>>> -0400)
>>>>>>>>>
>>>>>>>>>  Regards,
>>>>>>>>>  Charles
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110805/4a0fa947/attachment-0001.html 

From peder at networkoblivion.com  Fri Aug  5 19:01:35 2011
From: peder at networkoblivion.com (Peder)
Date: Fri, 5 Aug 2011 10:01:35 -0500
Subject: [Freeswitch-users] Cisco Presence
Message-ID: <075e01cc5380$91cea4e0$b56beea0$@com>

Has anybody ever managed to get presence working on the Cisco 79x1 series
phones?  I know the 79x0 phones don't support it, but the x1 and newer
should.  It works on the SPA, but that isn't what I am looking for.



From cogs66 at gmail.com  Fri Aug  5 17:47:26 2011
From: cogs66 at gmail.com (cogs66)
Date: Fri, 5 Aug 2011 06:47:26 -0700 (PDT)
Subject: [Freeswitch-users] SIP auth challenge error
Message-ID: <1312552046061-6656611.post@n2.nabble.com>

Hello All

First post here and a newb......

I have just updated the latest GIT and now seeing the following in the CLI.
Does anyone know how to fix this or is it nothing to worry about?

[WARNING] sofia_reg.c:1339 SIP auth challenge (REGISTER) on sofia profile
'internal' for xxx from ip

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/SIP-auth-challenge-error-tp6656611p6656611.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From rhuddleston at gmail.com  Fri Aug  5 20:35:10 2011
From: rhuddleston at gmail.com (Robert Huddleston)
Date: Fri, 5 Aug 2011 12:35:10 -0400
Subject: [Freeswitch-users] SIP auth challenge error
In-Reply-To: <1312552046061-6656611.post@n2.nabble.com>
References: <1312552046061-6656611.post@n2.nabble.com>
Message-ID: <013e01cc538d$a5a30440$f0e90cc0$@com>

Fail2ban quick!

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of cogs66
Sent: Friday, August 05, 2011 9:47 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] SIP auth challenge error

Hello All

First post here and a newb......

I have just updated the latest GIT and now seeing the following in the CLI.
Does anyone know how to fix this or is it nothing to worry about?

[WARNING] sofia_reg.c:1339 SIP auth challenge (REGISTER) on sofia profile
'internal' for xxx from ip

--
View this message in context:
http://freeswitch-users.2379917.n2.nabble.com/SIP-auth-challenge-error-tp665
6611p6656611.html
Sent from the freeswitch-users mailing list archive at Nabble.com.

_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

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



From cogs66 at gmail.com  Fri Aug  5 20:47:00 2011
From: cogs66 at gmail.com (cogs66)
Date: Fri, 5 Aug 2011 09:47:00 -0700 (PDT)
Subject: [Freeswitch-users] SIP auth challenge error
In-Reply-To: <013e01cc538d$a5a30440$f0e90cc0$@com>
References: <1312552046061-6656611.post@n2.nabble.com>
	<013e01cc538d$a5a30440$f0e90cc0$@com>
Message-ID: <1312562820805-6657220.post@n2.nabble.com>

Hi

Thanks for your reply

The errors are coming from extensions I have registered. All works fine, I
just see these error every few mins.

Andy

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/SIP-auth-challenge-error-tp6656611p6657220.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From freeswitch at earthspike.net  Fri Aug  5 20:58:30 2011
From: freeswitch at earthspike.net (John)
Date: Fri, 05 Aug 2011 17:58:30 +0100
Subject: [Freeswitch-users] SIP auth challenge error
In-Reply-To: <013e01cc538d$a5a30440$f0e90cc0$@com>
References: <1312552046061-6656611.post@n2.nabble.com>
	<013e01cc538d$a5a30440$f0e90cc0$@com>
Message-ID: <4E3C2136.4090203@earthspike.net>

Err... not fail2ban if it's one of your own clients.  I see this happen
about every 50 minutes per client.  So long as you recognise the IP
address that is authenticating, then it is normal and you have nothing
to worry about.

John

On 05/08/11 17:35, Robert Huddleston wrote:
> Fail2ban quick!
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of cogs66
> Sent: Friday, August 05, 2011 9:47 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] SIP auth challenge error
>
> Hello All
>
> First post here and a newb......
>
> I have just updated the latest GIT and now seeing the following in the CLI.
> Does anyone know how to fix this or is it nothing to worry about?
>
> [WARNING] sofia_reg.c:1339 SIP auth challenge (REGISTER) on sofia profile
> 'internal' for xxx from ip
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/SIP-auth-challenge-error-tp665
> 6611p6656611.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From rhuddleston at gmail.com  Fri Aug  5 21:06:52 2011
From: rhuddleston at gmail.com (Robert Huddleston)
Date: Fri, 5 Aug 2011 13:06:52 -0400
Subject: [Freeswitch-users] SIP auth challenge error
In-Reply-To: <4E3C2136.4090203@earthspike.net>
References: <1312552046061-6656611.post@n2.nabble.com>
	<013e01cc538d$a5a30440$f0e90cc0$@com>
	<4E3C2136.4090203@earthspike.net>
Message-ID: <014001cc5392$13295bf0$397c13d0$@com>

I missed the internal part... But either way - wouldn't be a bad idea to set
up fail2ban...

Trust me I've banned myself on internal side before too

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of John
Sent: Friday, August 05, 2011 12:59 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] SIP auth challenge error

Err... not fail2ban if it's one of your own clients.  I see this happen
about every 50 minutes per client.  So long as you recognise the IP
address that is authenticating, then it is normal and you have nothing
to worry about.

John

On 05/08/11 17:35, Robert Huddleston wrote:
> Fail2ban quick!
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of cogs66
> Sent: Friday, August 05, 2011 9:47 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] SIP auth challenge error
>
> Hello All
>
> First post here and a newb......
>
> I have just updated the latest GIT and now seeing the following in the
CLI.
> Does anyone know how to fix this or is it nothing to worry about?
>
> [WARNING] sofia_reg.c:1339 SIP auth challenge (REGISTER) on sofia profile
> 'internal' for xxx from ip
>
> --
> View this message in context:
>
http://freeswitch-users.2379917.n2.nabble.com/SIP-auth-challenge-error-tp665
> 6611p6656611.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

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



From msc at freeswitch.org  Fri Aug  5 21:08:35 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 5 Aug 2011 10:08:35 -0700
Subject: [Freeswitch-users] Hangup hooks on B legs
In-Reply-To: <CABj_V-kr=aaS8--ucc9O1ooHW0SorBZ96QcmdZJnawvzS=qB=A@mail.gmail.com>
References: <CABj_V-mOmm05ycYJZRPaeePeVzB4LWdTfzjNXwM3O=Kjmrn5tQ@mail.gmail.com>
	<CALNZuwm7Zh1FZpeavRECM6VjPUERsJODi=gRxWsXboR4hKEiLg@mail.gmail.com>
	<CABj_V-mubvz2cpCY1=ynLrAkDv088ZVURhBJ+1BiYe4XWhUyTA@mail.gmail.com>
	<CALNZuwkCW=ZiC9b8g9UCAkqFvFfWDyA_BywGLa_nJtvXv-2Y_Q@mail.gmail.com>
	<CABj_V-=uB_X77ZUrcYDU_Rx0yJLLwhzMAU3xtfoiOy=PeEwOdA@mail.gmail.com>
	<CALNZuwkszOa6mhUA7UuL7NM9ZHDKSL7NcLxA2p9G0MP6-h_GQQ@mail.gmail.com>
	<CABj_V-mfm79xO6imjmdvMidOqkiwYNxLPTCjJNXcscWTdB0B2g@mail.gmail.com>
	<CAFiqYunySoaNVp6Fb2Sg6YDLATLx7wh9yBW=RXA5Jiqyb_VyZA@mail.gmail.com>
	<CABj_V-=-Mx3OmBeKfrJWjU_yF3aAgN9REjxmP7oR0hReQvzSSQ@mail.gmail.com>
	<CAKzWOxVwhUzjn32J4TRkvAtxhrOObbgoPb46NYk3D-4mra4dBg@mail.gmail.com>
	<CABj_V-kr=aaS8--ucc9O1ooHW0SorBZ96QcmdZJnawvzS=qB=A@mail.gmail.com>
Message-ID: <CAKzWOxUu4FYMfA1CBBaW4dgxyNRbZDRkg+kzCtH_J_DairmpKQ@mail.gmail.com>

On Fri, Aug 5, 2011 at 2:54 AM, Isaac Jurado <ijurado at econcept.es> wrote:

> On Fri, Aug 5, 2011 at 2:13 AM, Michael Collins <msc at freeswitch.org>
> wrote:
> >
> > To help connect the dots I added some links to/from that Lua wiki page
> > and channel_in_hangup_hook and also api_hangup_hook and
> > reporting_hangup_hook.
>
> That "channel_in_hangup_hook" sounds promising to me, although I can't
> find any mention of it in the wiki:
>
>
> http://wiki.freeswitch.org/wiki/Special:Search?search=channel_in_hangup_hook&go=Go


That's what I get for trying to do too much all at once. The correct chan
var name is:
session_in_hangup_hook

Sorry...
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110805/894afb86/attachment.html 

From steveayre at gmail.com  Fri Aug  5 21:09:44 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 5 Aug 2011 18:09:44 +0100
Subject: [Freeswitch-users] SIP auth challenge error
In-Reply-To: <1312562820805-6657220.post@n2.nabble.com>
References: <1312552046061-6656611.post@n2.nabble.com>
	<013e01cc538d$a5a30440$f0e90cc0$@com>
	<1312562820805-6657220.post@n2.nabble.com>
Message-ID: <CAFiqYunCtX8TKi5jv9Gvc12tpebgfqUQipsqSs5HDBAiG-nT9g@mail.gmail.com>

It's logging that's been put in to support fail2ban (
http://www.fail2ban.org/). It watches your logs for anyone attempting to
access your server who shouldn't be (trying to guess passwords, DOS your
server etc) and blocking their IP. It's really only an Info message, but
logs as warning so that you still get the messages fail2ban needs if you're
only logging errors of warning level or higher. It's nothing to be concerned
about.

If you want to, use fail2ban. If you don't you can either ignore those
messages or set log-auth-failures to false on the SIP profile to stop
logging them.

-Steve



On 5 August 2011 17:47, cogs66 <cogs66 at gmail.com> wrote:

> Hi
>
> Thanks for your reply
>
> The errors are coming from extensions I have registered. All works fine, I
> just see these error every few mins.
>
> Andy
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/SIP-auth-challenge-error-tp6656611p6657220.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> on tFreeSWITCH-users at lists.freeswitch.org<FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110805/06597a49/attachment.html 

From covici at ccs.covici.com  Fri Aug  5 21:49:50 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Fri, 05 Aug 2011 13:49:50 -0400
Subject: [Freeswitch-users] SIP auth challenge error
In-Reply-To: <1312562820805-6657220.post@n2.nabble.com>
References: <1312552046061-6656611.post@n2.nabble.com>
	<013e01cc538d$a5a30440$f0e90cc0$@com>
	<1312562820805-6657220.post@n2.nabble.com>
Message-ID: <27719.1312566590@ccs.covici.com>

This is normal, they are used if you have too many of these per minute,
you can use fail2ban to block the ip address.

cogs66 <cogs66 at gmail.com> wrote:

> Hi
> 
> Thanks for your reply
> 
> The errors are coming from extensions I have registered. All works fine, I
> just see these error every few mins.
> 
> Andy
> 
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/SIP-auth-challenge-error-tp6656611p6657220.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From ijurado at econcept.es  Fri Aug  5 22:04:28 2011
From: ijurado at econcept.es (Isaac Jurado)
Date: Fri, 5 Aug 2011 20:04:28 +0200
Subject: [Freeswitch-users] Hangup hooks on B legs
In-Reply-To: <CAKzWOxUu4FYMfA1CBBaW4dgxyNRbZDRkg+kzCtH_J_DairmpKQ@mail.gmail.com>
References: <CABj_V-mOmm05ycYJZRPaeePeVzB4LWdTfzjNXwM3O=Kjmrn5tQ@mail.gmail.com>
	<CALNZuwm7Zh1FZpeavRECM6VjPUERsJODi=gRxWsXboR4hKEiLg@mail.gmail.com>
	<CABj_V-mubvz2cpCY1=ynLrAkDv088ZVURhBJ+1BiYe4XWhUyTA@mail.gmail.com>
	<CALNZuwkCW=ZiC9b8g9UCAkqFvFfWDyA_BywGLa_nJtvXv-2Y_Q@mail.gmail.com>
	<CABj_V-=uB_X77ZUrcYDU_Rx0yJLLwhzMAU3xtfoiOy=PeEwOdA@mail.gmail.com>
	<CALNZuwkszOa6mhUA7UuL7NM9ZHDKSL7NcLxA2p9G0MP6-h_GQQ@mail.gmail.com>
	<CABj_V-mfm79xO6imjmdvMidOqkiwYNxLPTCjJNXcscWTdB0B2g@mail.gmail.com>
	<CAFiqYunySoaNVp6Fb2Sg6YDLATLx7wh9yBW=RXA5Jiqyb_VyZA@mail.gmail.com>
	<CABj_V-=-Mx3OmBeKfrJWjU_yF3aAgN9REjxmP7oR0hReQvzSSQ@mail.gmail.com>
	<CAKzWOxVwhUzjn32J4TRkvAtxhrOObbgoPb46NYk3D-4mra4dBg@mail.gmail.com>
	<CABj_V-kr=aaS8--ucc9O1ooHW0SorBZ96QcmdZJnawvzS=qB=A@mail.gmail.com>
	<CAKzWOxUu4FYMfA1CBBaW4dgxyNRbZDRkg+kzCtH_J_DairmpKQ@mail.gmail.com>
Message-ID: <CABj_V-=uDgV8Dcp98eOEhY=LE2+0Eg=G=AuJqioSwtJgg59Rxw@mail.gmail.com>

On Fri, Aug 5, 2011 at 7:08 PM, Michael Collins <msc at freeswitch.org> wrote:
>
> That's what I get for trying to do too much all at once. The correct
> chan var name is:
> session_in_hangup_hook
> Sorry...

Heh, I was washing de dishes and I just realized you were meaning that.
And I was going to write an apology too.

Cheers and thanks for all.

-- 
Isaac Jurado
Internet Busines Solutions eConcept


From msc at freeswitch.org  Fri Aug  5 22:31:03 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 5 Aug 2011 11:31:03 -0700
Subject: [Freeswitch-users] outboundsocket mode,
 bridge and bind_digit_action doesn't work
In-Reply-To: <20110804212458.4462c72d@gmail.com>
References: <20110804212458.4462c72d@gmail.com>
Message-ID: <CAKzWOxXFHq-s5xKh4tRF61jmTeRroNEio=VaE7X3jhucEXzPpQ@mail.gmail.com>

Michael,

I think the bridge app is clearing things out. I got it to work by bridging
first and then doing the BDA stuff:

connect

sendmsg
call-command: execute
execute-app-name: answer

sendmsg
call-command: execute
execute-app-name: bridge
execute-app-arg: user/1007

sendmsg
call-command: execute
execute-app-name: bind_digit_action
execute-app-arg: test,*0,exec:log,NOTICE TEST

sendmsg
call-command: execute
execute-app-name: digit_action_set_realm
execute-app-arg: test


Let me know if that works for you.
-MC

On Thu, Aug 4, 2011 at 12:24 PM, Michael Ricordeau <
michael.ricordeau at gmail.com> wrote:

> Hi,
>
> I think I'm doing something wrong but I don't find a way to fix my problem
> :
>
> when executing bind_digit_action in an outbound socket and doing a bridge,
> digit action binding is not executed (no log in Freeswitch)
>
> when doing same thing in XML dialplan it works (I found log in Freeswitch).
>
> I have checked logs and in both cases, digit realm is set :
> Digit parser DPTOOLS: binding 00/test/0 callback: 0xb6ba1b00 data:
> 0x82eafe8
>
>
>
> For outbound socket, here what I'm doing with netcat :
>
>
> nc -l -v 8084
>
>
> and the commands I passed to outbound socket :
>
>
> connect
>
> divert_events on
>
> sendmsg
> call-command: execute
> execute-app-name: answer
>
> sendmsg
> call-command: execute
> execute-app-name: bind_digit_action
> execute-app-arg: test,00,exec:log,NOTICE TEST
>
> sendmsg
> call-command: execute
> execute-app-name: bridge
> execute-app-arg: user/1000
>
>
>
>
> With XML Dialplan below, same binding/bridge works !!!!! :
>
>    <extension name="test">
>        <condition field="destination_number" expression="^0000$">
>            <action application="answer"/>
>            <action application="bind_digit_action"
> data="test,00,exec:log,NOTICE TEST"/>
>            <action application="bridge" data="user/1000"/>
>        </condition>
>    </extension>
>
>
>
> Expected result is when A leg presses "00", a log notice "TEST" is printed
> in the logger but only works with XML dialplan.
> In outbound socket mode, I can hear the digits pressed on B leg but binding
> is not executed (no log notice "TEST")
>
>
> Thanks
>
>
> Micha?l
>
>
>
>
>
>
>
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110805/e106553e/attachment.html 

From michael.ricordeau at gmail.com  Fri Aug  5 23:31:05 2011
From: michael.ricordeau at gmail.com (Michael Ricordeau)
Date: Fri, 5 Aug 2011 21:31:05 +0200
Subject: [Freeswitch-users] outboundsocket mode,
 bridge and bind_digit_action doesn't work
In-Reply-To: <CAKzWOxXFHq-s5xKh4tRF61jmTeRroNEio=VaE7X3jhucEXzPpQ@mail.gmail.com>
References: <20110804212458.4462c72d@gmail.com>
	<CAKzWOxXFHq-s5xKh4tRF61jmTeRroNEio=VaE7X3jhucEXzPpQ@mail.gmail.com>
Message-ID: <20110805213105.18c5a872@gmail.com>

Hi Michael,

it works with the BDA set after bridge !

Thanks for the quick and perfect answer =)

See you at Cluecon next week !


Michael



Le Fri, 5 Aug 2011 11:31:03 -0700,
Michael Collins <msc at freeswitch.org> a ?crit :

> Michael,
> 
> I think the bridge app is clearing things out. I got it to work by bridging
> first and then doing the BDA stuff:
> 
> connect
> 
> sendmsg
> call-command: execute
> execute-app-name: answer
> 
> sendmsg
> call-command: execute
> execute-app-name: bridge
> execute-app-arg: user/1007
> 
> sendmsg
> call-command: execute
> execute-app-name: bind_digit_action
> execute-app-arg: test,*0,exec:log,NOTICE TEST
> 
> sendmsg
> call-command: execute
> execute-app-name: digit_action_set_realm
> execute-app-arg: test
> 
> 
> Let me know if that works for you.
> -MC
> 
> On Thu, Aug 4, 2011 at 12:24 PM, Michael Ricordeau <
> michael.ricordeau at gmail.com> wrote:
> 
> > Hi,
> >
> > I think I'm doing something wrong but I don't find a way to fix my problem
> > :
> >
> > when executing bind_digit_action in an outbound socket and doing a bridge,
> > digit action binding is not executed (no log in Freeswitch)
> >
> > when doing same thing in XML dialplan it works (I found log in Freeswitch).
> >
> > I have checked logs and in both cases, digit realm is set :
> > Digit parser DPTOOLS: binding 00/test/0 callback: 0xb6ba1b00 data:
> > 0x82eafe8
> >
> >
> >
> > For outbound socket, here what I'm doing with netcat :
> >
> >
> > nc -l -v 8084
> >
> >
> > and the commands I passed to outbound socket :
> >
> >
> > connect
> >
> > divert_events on
> >
> > sendmsg
> > call-command: execute
> > execute-app-name: answer
> >
> > sendmsg
> > call-command: execute
> > execute-app-name: bind_digit_action
> > execute-app-arg: test,00,exec:log,NOTICE TEST
> >
> > sendmsg
> > call-command: execute
> > execute-app-name: bridge
> > execute-app-arg: user/1000
> >
> >
> >
> >
> > With XML Dialplan below, same binding/bridge works !!!!! :
> >
> >    <extension name="test">
> >        <condition field="destination_number" expression="^0000$">
> >            <action application="answer"/>
> >            <action application="bind_digit_action"
> > data="test,00,exec:log,NOTICE TEST"/>
> >            <action application="bridge" data="user/1000"/>
> >        </condition>
> >    </extension>
> >
> >
> >
> > Expected result is when A leg presses "00", a log notice "TEST" is printed
> > in the logger but only works with XML dialplan.
> > In outbound socket mode, I can hear the digits pressed on B leg but binding
> > is not executed (no log notice "TEST")
> >
> >
> > Thanks
> >
> >
> > Micha?l
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >


From michael.ricordeau at gmail.com  Fri Aug  5 23:47:26 2011
From: michael.ricordeau at gmail.com (Michael Ricordeau)
Date: Fri, 5 Aug 2011 21:47:26 +0200
Subject: [Freeswitch-users] outboundsocket mode,
 bridge and bind_digit_action doesn't work
In-Reply-To: <20110805213105.18c5a872@gmail.com>
References: <20110804212458.4462c72d@gmail.com>
	<CAKzWOxXFHq-s5xKh4tRF61jmTeRroNEio=VaE7X3jhucEXzPpQ@mail.gmail.com>
	<20110805213105.18c5a872@gmail.com>
Message-ID: <20110805214726.320a362c@gmail.com>

Well so why bridge in outbound eventsocket is clearing BDA and not in dialplan ?
(just for my knowledge)

Is it because dialplan is "preprocessed" before executing commands ?

  



Le Fri, 5 Aug 2011 21:31:05 +0200,
Michael Ricordeau <michael.ricordeau at gmail.com> a ?crit :

> Hi Michael,
> 
> it works with the BDA set after bridge !
> 
> Thanks for the quick and perfect answer =)
> 
> See you at Cluecon next week !
> 
> 
> Michael
> 
> 
> 
> Le Fri, 5 Aug 2011 11:31:03 -0700,
> Michael Collins <msc at freeswitch.org> a ?crit :
> 
> > Michael,
> > 
> > I think the bridge app is clearing things out. I got it to work by bridging
> > first and then doing the BDA stuff:
> > 
> > connect
> > 
> > sendmsg
> > call-command: execute
> > execute-app-name: answer
> > 
> > sendmsg
> > call-command: execute
> > execute-app-name: bridge
> > execute-app-arg: user/1007
> > 
> > sendmsg
> > call-command: execute
> > execute-app-name: bind_digit_action
> > execute-app-arg: test,*0,exec:log,NOTICE TEST
> > 
> > sendmsg
> > call-command: execute
> > execute-app-name: digit_action_set_realm
> > execute-app-arg: test
> > 
> > 
> > Let me know if that works for you.
> > -MC
> > 
> > On Thu, Aug 4, 2011 at 12:24 PM, Michael Ricordeau <
> > michael.ricordeau at gmail.com> wrote:
> > 
> > > Hi,
> > >
> > > I think I'm doing something wrong but I don't find a way to fix my problem
> > > :
> > >
> > > when executing bind_digit_action in an outbound socket and doing a bridge,
> > > digit action binding is not executed (no log in Freeswitch)
> > >
> > > when doing same thing in XML dialplan it works (I found log in Freeswitch).
> > >
> > > I have checked logs and in both cases, digit realm is set :
> > > Digit parser DPTOOLS: binding 00/test/0 callback: 0xb6ba1b00 data:
> > > 0x82eafe8
> > >
> > >
> > >
> > > For outbound socket, here what I'm doing with netcat :
> > >
> > >
> > > nc -l -v 8084
> > >
> > >
> > > and the commands I passed to outbound socket :
> > >
> > >
> > > connect
> > >
> > > divert_events on
> > >
> > > sendmsg
> > > call-command: execute
> > > execute-app-name: answer
> > >
> > > sendmsg
> > > call-command: execute
> > > execute-app-name: bind_digit_action
> > > execute-app-arg: test,00,exec:log,NOTICE TEST
> > >
> > > sendmsg
> > > call-command: execute
> > > execute-app-name: bridge
> > > execute-app-arg: user/1000
> > >
> > >
> > >
> > >
> > > With XML Dialplan below, same binding/bridge works !!!!! :
> > >
> > >    <extension name="test">
> > >        <condition field="destination_number" expression="^0000$">
> > >            <action application="answer"/>
> > >            <action application="bind_digit_action"
> > > data="test,00,exec:log,NOTICE TEST"/>
> > >            <action application="bridge" data="user/1000"/>
> > >        </condition>
> > >    </extension>
> > >
> > >
> > >
> > > Expected result is when A leg presses "00", a log notice "TEST" is printed
> > > in the logger but only works with XML dialplan.
> > > In outbound socket mode, I can hear the digits pressed on B leg but binding
> > > is not executed (no log notice "TEST")
> > >
> > >
> > > Thanks
> > >
> > >
> > > Micha?l
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > > _______________________________________________
> > > Join us at ClueCon 2011, Aug 9-11, Chicago
> > > http://www.cluecon.com 877-7-4ACLUE
> > >
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >


From msc at freeswitch.org  Sat Aug  6 00:03:24 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 5 Aug 2011 13:03:24 -0700
Subject: [Freeswitch-users] outboundsocket mode,
 bridge and bind_digit_action doesn't work
In-Reply-To: <20110805214726.320a362c@gmail.com>
References: <20110804212458.4462c72d@gmail.com>
	<CAKzWOxXFHq-s5xKh4tRF61jmTeRroNEio=VaE7X3jhucEXzPpQ@mail.gmail.com>
	<20110805213105.18c5a872@gmail.com>
	<20110805214726.320a362c@gmail.com>
Message-ID: <CAKzWOxW-TiVVdJH27VOZmR9iRE3hMQY81idM9Fur4M53kNnHUA@mail.gmail.com>

On Fri, Aug 5, 2011 at 12:47 PM, Michael Ricordeau <
michael.ricordeau at gmail.com> wrote:

> Well so why bridge in outbound eventsocket is clearing BDA and not in
> dialplan ?
> (just for my knowledge)
>
> Is it because dialplan is "preprocessed" before executing commands ?
>

I'm not sure. I discovered it when I was looking for an alternative way to
set the BDA on the channel. I was using uuid_bridge <uuid>
bind_digit_action::exec:log,foo,etc. and I noticed the same thing happen -
if I did it prior to the bridge then it didn't work. Tony can probably tell
us if this is a bug, feature, expected operation, etc.

-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110805/e9eb8d84/attachment.html 

From steveayre at gmail.com  Sat Aug  6 00:31:44 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 5 Aug 2011 21:31:44 +0100
Subject: [Freeswitch-users] outboundsocket mode,
	bridge and bind_digit_action doesn't work
In-Reply-To: <20110805214726.320a362c@gmail.com>
References: <20110804212458.4462c72d@gmail.com>
	<CAKzWOxXFHq-s5xKh4tRF61jmTeRroNEio=VaE7X3jhucEXzPpQ@mail.gmail.com>
	<20110805213105.18c5a872@gmail.com>
	<20110805214726.320a362c@gmail.com>
Message-ID: <AEEE5575-C51F-4291-9498-5DC9E21B1520@gmail.com>

Shouldn't be, they're applications so executed one at a time.

Steve on iPhone

On 5 Aug 2011, at 20:47, Michael Ricordeau <michael.ricordeau at gmail.com> wrote:

> Well so why bridge in outbound eventsocket is clearing BDA and not in dialplan ?
> (just for my knowledge)
> 
> Is it because dialplan is "preprocessed" before executing commands ?
> 
> 
> 
> 
> 
> Le Fri, 5 Aug 2011 21:31:05 +0200,
> Michael Ricordeau <michael.ricordeau at gmail.com> a ?crit :
> 
>> Hi Michael,
>> 
>> it works with the BDA set after bridge !
>> 
>> Thanks for the quick and perfect answer =)
>> 
>> See you at Cluecon next week !
>> 
>> 
>> Michael
>> 
>> 
>> 
>> Le Fri, 5 Aug 2011 11:31:03 -0700,
>> Michael Collins <msc at freeswitch.org> a ?crit :
>> 
>>> Michael,
>>> 
>>> I think the bridge app is clearing things out. I got it to work by bridging
>>> first and then doing the BDA stuff:
>>> 
>>> connect
>>> 
>>> sendmsg
>>> call-command: execute
>>> execute-app-name: answer
>>> 
>>> sendmsg
>>> call-command: execute
>>> execute-app-name: bridge
>>> execute-app-arg: user/1007
>>> 
>>> sendmsg
>>> call-command: execute
>>> execute-app-name: bind_digit_action
>>> execute-app-arg: test,*0,exec:log,NOTICE TEST
>>> 
>>> sendmsg
>>> call-command: execute
>>> execute-app-name: digit_action_set_realm
>>> execute-app-arg: test
>>> 
>>> 
>>> Let me know if that works for you.
>>> -MC
>>> 
>>> On Thu, Aug 4, 2011 at 12:24 PM, Michael Ricordeau <
>>> michael.ricordeau at gmail.com> wrote:
>>> 
>>>> Hi,
>>>> 
>>>> I think I'm doing something wrong but I don't find a way to fix my problem
>>>> :
>>>> 
>>>> when executing bind_digit_action in an outbound socket and doing a bridge,
>>>> digit action binding is not executed (no log in Freeswitch)
>>>> 
>>>> when doing same thing in XML dialplan it works (I found log in Freeswitch).
>>>> 
>>>> I have checked logs and in both cases, digit realm is set :
>>>> Digit parser DPTOOLS: binding 00/test/0 callback: 0xb6ba1b00 data:
>>>> 0x82eafe8
>>>> 
>>>> 
>>>> 
>>>> For outbound socket, here what I'm doing with netcat :
>>>> 
>>>> 
>>>> nc -l -v 8084
>>>> 
>>>> 
>>>> and the commands I passed to outbound socket :
>>>> 
>>>> 
>>>> connect
>>>> 
>>>> divert_events on
>>>> 
>>>> sendmsg
>>>> call-command: execute
>>>> execute-app-name: answer
>>>> 
>>>> sendmsg
>>>> call-command: execute
>>>> execute-app-name: bind_digit_action
>>>> execute-app-arg: test,00,exec:log,NOTICE TEST
>>>> 
>>>> sendmsg
>>>> call-command: execute
>>>> execute-app-name: bridge
>>>> execute-app-arg: user/1000
>>>> 
>>>> 
>>>> 
>>>> 
>>>> With XML Dialplan below, same binding/bridge works !!!!! :
>>>> 
>>>>   <extension name="test">
>>>>       <condition field="destination_number" expression="^0000$">
>>>>           <action application="answer"/>
>>>>           <action application="bind_digit_action"
>>>> data="test,00,exec:log,NOTICE TEST"/>
>>>>           <action application="bridge" data="user/1000"/>
>>>>       </condition>
>>>>   </extension>
>>>> 
>>>> 
>>>> 
>>>> Expected result is when A leg presses "00", a log notice "TEST" is printed
>>>> in the logger but only works with XML dialplan.
>>>> In outbound socket mode, I can hear the digits pressed on B leg but binding
>>>> is not executed (no log notice "TEST")
>>>> 
>>>> 
>>>> Thanks
>>>> 
>>>> 
>>>> Micha?l
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From steveayre at gmail.com  Sat Aug  6 00:42:19 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 5 Aug 2011 21:42:19 +0100
Subject: [Freeswitch-users] outboundsocket mode,
 bridge and bind_digit_action doesn't work
In-Reply-To: <AEEE5575-C51F-4291-9498-5DC9E21B1520@gmail.com>
References: <20110804212458.4462c72d@gmail.com>
	<CAKzWOxXFHq-s5xKh4tRF61jmTeRroNEio=VaE7X3jhucEXzPpQ@mail.gmail.com>
	<20110805213105.18c5a872@gmail.com> <20110805214726.320a362c@gmail.com>
	<AEEE5575-C51F-4291-9498-5DC9E21B1520@gmail.com>
Message-ID: <CAFiqYumjsSjov2A+08wGQJfofE4jWV5nXWO8gRUCUbzOgKuhtw@mail.gmail.com>

I wonder if it's a race condition of sorts where you're executing some apps
before the others have finished.

In the dialplan it wouldn't execute the next until the previous had
finished, but I'm not sure whether sendmsg queues apps in the same way or
tries executing them straight away.

Have you tried waiting for the CHANNEL_EXECUTE_COMPLETE event before sending
the next sendmsg?

-Steve



On 5 August 2011 21:31, Steven Ayre <steveayre at gmail.com> wrote:

> Shouldn't be, they're applications so executed one at a time.
>
> Steve on iPhone
>
> On 5 Aug 2011, at 20:47, Michael Ricordeau <michael.ricordeau at gmail.com>
> wrote:
>
> > Well so why bridge in outbound eventsocket is clearing BDA and not in
> dialplan ?
> > (just for my knowledge)
> >
> > Is it because dialplan is "preprocessed" before executing commands ?
> >
> >
> >
> >
> >
> > Le Fri, 5 Aug 2011 21:31:05 +0200,
> > Michael Ricordeau <michael.ricordeau at gmail.com> a ?crit :
> >
> >> Hi Michael,
> >>
> >> it works with the BDA set after bridge !
> >>
> >> Thanks for the quick and perfect answer =)
> >>
> >> See you at Cluecon next week !
> >>
> >>
> >> Michael
> >>
> >>
> >>
> >> Le Fri, 5 Aug 2011 11:31:03 -0700,
> >> Michael Collins <msc at freeswitch.org> a ?crit :
> >>
> >>> Michael,
> >>>
> >>> I think the bridge app is clearing things out. I got it to work by
> bridging
> >>> first and then doing the BDA stuff:
> >>>
> >>> connect
> >>>
> >>> sendmsg
> >>> call-command: execute
> >>> execute-app-name: answer
> >>>
> >>> sendmsg
> >>> call-command: execute
> >>> execute-app-name: bridge
> >>> execute-app-arg: user/1007
> >>>
> >>> sendmsg
> >>> call-command: execute
> >>> execute-app-name: bind_digit_action
> >>> execute-app-arg: test,*0,exec:log,NOTICE TEST
> >>>
> >>> sendmsg
> >>> call-command: execute
> >>> execute-app-name: digit_action_set_realm
> >>> execute-app-arg: test
> >>>
> >>>
> >>> Let me know if that works for you.
> >>> -MC
> >>>
> >>> On Thu, Aug 4, 2011 at 12:24 PM, Michael Ricordeau <
> >>> michael.ricordeau at gmail.com> wrote:
> >>>
> >>>> Hi,
> >>>>
> >>>> I think I'm doing something wrong but I don't find a way to fix my
> problem
> >>>> :
> >>>>
> >>>> when executing bind_digit_action in an outbound socket and doing a
> bridge,
> >>>> digit action binding is not executed (no log in Freeswitch)
> >>>>
> >>>> when doing same thing in XML dialplan it works (I found log in
> Freeswitch).
> >>>>
> >>>> I have checked logs and in both cases, digit realm is set :
> >>>> Digit parser DPTOOLS: binding 00/test/0 callback: 0xb6ba1b00 data:
> >>>> 0x82eafe8
> >>>>
> >>>>
> >>>>
> >>>> For outbound socket, here what I'm doing with netcat :
> >>>>
> >>>>
> >>>> nc -l -v 8084
> >>>>
> >>>>
> >>>> and the commands I passed to outbound socket :
> >>>>
> >>>>
> >>>> connect
> >>>>
> >>>> divert_events on
> >>>>
> >>>> sendmsg
> >>>> call-command: execute
> >>>> execute-app-name: answer
> >>>>
> >>>> sendmsg
> >>>> call-command: execute
> >>>> execute-app-name: bind_digit_action
> >>>> execute-app-arg: test,00,exec:log,NOTICE TEST
> >>>>
> >>>> sendmsg
> >>>> call-command: execute
> >>>> execute-app-name: bridge
> >>>> execute-app-arg: user/1000
> >>>>
> >>>>
> >>>>
> >>>>
> >>>> With XML Dialplan below, same binding/bridge works !!!!! :
> >>>>
> >>>>   <extension name="test">
> >>>>       <condition field="destination_number" expression="^0000$">
> >>>>           <action application="answer"/>
> >>>>           <action application="bind_digit_action"
> >>>> data="test,00,exec:log,NOTICE TEST"/>
> >>>>           <action application="bridge" data="user/1000"/>
> >>>>       </condition>
> >>>>   </extension>
> >>>>
> >>>>
> >>>>
> >>>> Expected result is when A leg presses "00", a log notice "TEST" is
> printed
> >>>> in the logger but only works with XML dialplan.
> >>>> In outbound socket mode, I can hear the digits pressed on B leg but
> binding
> >>>> is not executed (no log notice "TEST")
> >>>>
> >>>>
> >>>> Thanks
> >>>>
> >>>>
> >>>> Micha?l
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>> _______________________________________________
> >>>> Join us at ClueCon 2011, Aug 9-11, Chicago
> >>>> http://www.cluecon.com 877-7-4ACLUE
> >>>>
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>>
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110805/69c65b9a/attachment.html 

From msc at freeswitch.org  Sat Aug  6 02:30:14 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 5 Aug 2011 15:30:14 -0700
Subject: [Freeswitch-users] Problem with XML CDR
In-Reply-To: <CA+7EGb=nj_XKFkPMijtD_MSj9XurFG0THDZVq+mr+AExcqrEaA@mail.gmail.com>
References: <CA+7EGbmk6WhgC8tCfFtSqkrQDosd4j87x7x1Bei0J+Km=HWHpw@mail.gmail.com>
	<CALNZuwnK9g83RoGF4CnM0Ou8qAabYPB-nWQ7_uAuYL_Cr=1zLw@mail.gmail.com>
	<CA+7EGb=nj_XKFkPMijtD_MSj9XurFG0THDZVq+mr+AExcqrEaA@mail.gmail.com>
Message-ID: <CAKzWOxWJe85gThLNsEjPNo2cVgoxxv_wkjWgOeNWxZagd4SjSQ@mail.gmail.com>

Can you open the XML CDR in a hex editor and see if there are goofy
characters? I wonder if there's an encoding issue with the info sent over
from the carrier.

-MC

On Thu, Aug 4, 2011 at 3:03 AM, Nice Voip <nicevoip at googlemail.com> wrote:

> Hmm i've latest version too, but there is not much traffic, and also on
> this two month older version its rarely happen, in any case i'll try to move
> to latest version and will try to reproduce.... but when it will be
> reproduced my version will not be latest anymore :)
>
>
>
>
> On Thu, Aug 4, 2011 at 11:56 AM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> The usual response is.. can you update from your nearly two month old
>> version and reproduce the same issue?
>> -Avi
>>
>>
>> On Thu, Aug 4, 2011 at 12:48 PM, Nice Voip <nicevoip at googlemail.com>wrote:
>>
>>> Dear All,
>>>
>>> This problem is very hard to reproduce and i really don't know when it
>>> would happen, i don't have sip log or other traces but only the CDR file and
>>> its looks like this:
>>>
>>>
>>> <caller_id_name>TELESAT TRAJKOVI?
>>>       <ani>+3xxxxxxx</ani>          ? MILE</caller_id_name>
>>>       <aniii></aniii>
>>>
>>> and sometimes
>>>
>>> <oritinate_early_media> is writted <originate_uarly_media> (insead of
>>> early its "uarly")
>>>
>>> i noted down this issue on: FreeSWITCH Version 1.0.head (git-1d3417a
>>> 2011-06-07 17-35-49 -0400)
>>>
>>> Thanks.
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110805/c8aa673b/attachment.html 

From devel at omninet.eu  Sat Aug  6 11:38:13 2011
From: devel at omninet.eu (Anestis Mavro)
Date: Sat, 6 Aug 2011 10:38:13 +0300
Subject: [Freeswitch-users] incompatible destination because of crypto on
	incoming calls
Message-ID: <69CD4836D15D45B683FFFBAB557B5060@omni1.local>

Hi,

 

Suddenly (latest git) something happened with the incoming calls.

All calls have crypto enabled and they get dropped with "INCOMPATIBLE
DESTINATION"

I have TLS disabled in vars.xml but still I see in debug:

 

Sofia_glue:2900 Set Local Key [1 AES_CM_128_HMAC_SHA1_32 inline: ...]

 

Show channels and show calls show now a lot of calls, but they don't exist;
they don't get cleared.

 

Anybody any idea where this can come from?

 

Thank you

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110806/efe88ec3/attachment-0001.html 

From cogs66 at gmail.com  Sat Aug  6 10:39:31 2011
From: cogs66 at gmail.com (cogs66)
Date: Fri, 5 Aug 2011 23:39:31 -0700 (PDT)
Subject: [Freeswitch-users] SIP auth challenge error
In-Reply-To: <27719.1312566590@ccs.covici.com>
References: <1312552046061-6656611.post@n2.nabble.com>
	<013e01cc538d$a5a30440$f0e90cc0$@com>
	<1312562820805-6657220.post@n2.nabble.com>
	<27719.1312566590@ccs.covici.com>
Message-ID: <1312612771234-6659001.post@n2.nabble.com>

Thank you all for your replies and for clarifying for me, I will keep an eye
on the CLI and act according.

Andy



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/SIP-auth-challenge-error-tp6656611p6659001.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From u2nsam at gmail.com  Sat Aug  6 13:36:57 2011
From: u2nsam at gmail.com (Sam)
Date: Sat, 6 Aug 2011 15:06:57 +0530
Subject: [Freeswitch-users] conference count
Message-ID: <CAE2ghw7X+cmcagb_PFQ0GZrTdYV+=-3iJJEXku-XoknvDqKmRw@mail.gmail.com>

*I have a dial plan ,*

<extension name="dynamic_conferences_guest">
      <condition field="destination_number" expression="^1234">
    <action application="answer"/>
    <action application="info"/>
        <action application="set" data="conf_count=${conference
${conference_name} list count}"/>
        <action application="log" data="INFO ${conf_count}"/>
        <action application="say" data="en number pronounced
${conf_count}"/>
    <action application="conference" data="1000146 at meeting"/>
      </condition>
    </extension>


*and the output of it as below:-*

2011-08-06 14:32:53.205895 [DEBUG] sofia.c:5084 Channel sofia/internal/
61323123999 at 192.153.53.158 entering state [ready][200]
EXECUTE sofia/internal/61323123999 at 192.153.53.158 set(conf_count=Conference
1000146 (1 member rate: 16000)
14;sofia/internal/883510001282001 at sip.ntfone.in
;97f5f386-c00a-11e0-9517-738827116f37;883510001282001;883510001282001;hear|speak|floor;0;0;0;300
)
2011-08-06 14:32:53.205895 [DEBUG] mod_dptools.c:1063 sofia/internal/
61323123999 at 192.153.53.158 SET [conf_count]=[Conference 1000146 (1 member
rate: 16000)
14;sofia/internal/883510001282001 at sip.nfone.in
;97f5f386-c00a-11e0-9517-738827116f37;883510001282001;883510001282001;hear|speak|floor;0;0;0;300
]
EXECUTE sofia/internal/61323123999 at 192.153.53.158 log(INFO Conference
1000146 (1 member rate: 16000)
14;sofia/internal/883510001282001 at sip.ntfone.in
;97f5f386-c00a-11e0-9517-738827116f37;883510001282001;883510001282001;hear|speak|floor;0;0;0;300
)
2011-08-06 14:32:53.205895 [INFO] mod_dptools.c:1202 Conference 1000146 (1
member rate: 16000)
14;sofia/internal/883510001282001 at sip.nfone.in
;97f5f386-c00a-11e0-9517-738827116f37;883510001282001;883510001282001;hear|speak|floor;0;0;0;300

EXECUTE sofia/internal/61323123999 at 192.153.53.158 say(en number pronounced
Conference 1000146 (1 member rate: 16000)
14;sofia/internal/883510001282001 at sip.nfone.in
;97f5f386-c00a-11e0-9517-738827116f37;883510001282001;883510001282001;hear|speak|floor;0;0;0;300
)
2011-08-06 14:32:53.215905 [ERR] mod_say_en.c:130 Parse Error!

EXECUTE sofia/internal/61323123999 at 192.153.53.158 conference(1000146 at meeting
)




*What should be my dialplan to get the exact conf_count ?*


Thanks in advance
Sam
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110806/92a2b12e/attachment.html 

From tculjaga at gmail.com  Sat Aug  6 20:04:25 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Sat, 6 Aug 2011 18:04:25 +0200
Subject: [Freeswitch-users] Mod_rad_auth issue for FS working with
 FreeRadius server
In-Reply-To: <CAOTmbMxW=EDr3-Ayo2R_MsCCTUZ4PnDkkRUuR2FisH9tK5xkWA@mail.gmail.com>
References: <CAOTmbMzvd5OHL8z=_8vu6Orp2_prGxaw1gmXYCM2=2+pgzs_WA@mail.gmail.com>
	<CAJoKKdX01Qc8ibs25PUELi8GrzMDNOrHb_bQvYdq3ZqA4XLy4w@mail.gmail.com>
	<CAOTmbMxL3pLOuw655fyT5VzAON_N0PrDN+-i-++=oZoYU6+Fmg@mail.gmail.com>
	<CAOTmbMyvyDA-CRpaMC=WRGVkiyTG=Rs+n4hyeQWdyTZpqysSzw@mail.gmail.com>
	<CAJoKKdXSwF02S9tBkorGwOjj57AtkRpU0c177qXASEpodKVk=w@mail.gmail.com>
	<CAOTmbMzu2qDYh2T24JSpCYn1xH2J4iGiK7UkUB4Kq=-KdcDWGQ@mail.gmail.com>
	<CAJoKKdU66jK3f+mOEV90Y3zrmzpTSdWnwGsWimZxir42khiJ-w@mail.gmail.com>
	<CAOTmbMxW=EDr3-Ayo2R_MsCCTUZ4PnDkkRUuR2FisH9tK5xkWA@mail.gmail.com>
Message-ID: <CAJoKKdVSkQOV3Kou_ro5AQZTyaqH6r0WRE-hZdrA=ye8ARE_vQ@mail.gmail.com>

are u sure you are using the correct secret on both client and server ?

On Fri, Aug 5, 2011 at 10:12 AM, fieldpeak <fieldpeak at gmail.com> wrote:

> Hi Tihomir,
>
> Thanks for your advise, i've added below to rad_auth.conf.xml (vsas
> section), as well as tried auth-type to 0(local) and 1(system), however, the
> issue still exist.
>
>
> <param name="NAS-Port-Type" id="61" value="0" pec="0" expr="0"
> direction="in"/>
> <param name="Login-User" id="6" value="1" pec="0" expr="0" direction="in"/>
> <param name="Auth-Type" id="1000" value="0" pec="0" expr="0"
> direction="in"/>
>
> FreeRadius output:
>
> Found Auth-Type = PAP
> # Executing group from file /usr/local/etc/raddb/sites-enabled/default
> +- entering group PAP {...}
> [pap] login attempt with password "Q?????? ??????p???F?+??a"
> [pap] Using clear text password "1111"
> [pap] Passwords don't match
> ++[pap] returns reject
> Failed to authenticate the user.
>   WARNING: Unprintable characters in the password.        Double-check the shared secret on the server and the NAS!
>
> Using Post-Auth-Type Reject
> # Executing group from file /usr/local/etc/raddb/sites-enabled/default
> +- entering group REJECT {...}
> [attr_filter.access_reject]     expand: %{User-Name} -> 1001
>  attr_filter: Matched entry DEFAULT at line 11
> ++[attr_filter.access_reject] returns updated
> Delaying reject of request 38 for 1 seconds
>
> Regards,
> Charles
>
>
> 2011/8/5 Tihomir Culjaga <tculjaga at gmail.com>
>
>> add to rad_auth.conf.xml
>>
>> <param name="NAS-Port-Type" id="61" value="0" pec="0" expr="0"
>> direction="in"/>
>> <param name="Login-User" id="6" value="1" pec="0" expr="0"
>> direction="in"/>
>>
>>
>>
>> as for Auth Type im not sure if you need it ... this is up to your server.
>> According to dictionary file you need to set it as follows:
>>
>> <param name="Auth-Type" id="1000" value="?" pec="0" expr="0"
>> direction="in"/>
>>
>> the value (set as ?) is one of the folowing. Again, not sure what is
>> required by your server.
>>
>> VALUE           Auth-Type               Local                   0
>> VALUE           Auth-Type               System                  1
>> VALUE           Auth-Type               SecurID                 2
>> VALUE           Auth-Type               Crypt-Local             3
>> VALUE           Auth-Type               Reject                  4
>>
>> #
>> #       Cistron extensions
>> #
>> VALUE           Auth-Type               Pam                     253
>> VALUE           Auth-Type               Accept                  254
>>
>>
>>
>> regards,
>> Tihomir.
>>
>>
>>
>> On Wed, Aug 3, 2011 at 6:32 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>>
>>> Hi Tihomir,
>>>
>>> Sorry, i missed your mail in gmail before, just now saw it, and after
>>> using your dictionary.all, the dictionary issue was resolved, very
>>> appreciated for your kindly help! however, it did not fully functional yet,
>>>
>>> Attached are configuration files that i used, when i dial 601 to trigger
>>> to auth, the freeradius server shows log below, the supecious log is the
>>> value User-Password, it should be '1111' that i've set in the mysql db of
>>> freeradisu server for the user 1001 .
>>>
>>> i searched in google, for "known good" password issue, i suggest change
>>> user-password to cleartext-password, however, i did not find where it is.
>>> and also the Auth-Type, where to configure it...
>>>
>>> Freeradius server log:
>>>
>>> rad_recv: Access-Request packet from host 127.0.0.1 port 52684, id=49,
>>> length=111
>>>         User-Name = "1001"
>>>         User-Password = "?\210\365@\263\t\306\343\243iT?\311C\t\002"
>>>         Called-Station-Id = "888"
>>>         h323-conf-id = "749d2b5a-16ad-48e4-af58-24011949d1b5"
>>>         Calling-Station-Id = "1001"
>>>         NAS-Port = 0
>>>         NAS-IP-Address = 127.0.0.1
>>> # Executing section authorize from file
>>> /usr/local/etc/raddb/sites-enabled/default
>>> +- entering group authorize {...}
>>> ++[preprocess] returns ok
>>> [auth_log]      expand:
>>> /usr/local/var/log/radius/radacct/%{Client-IP-Address}/auth-detail-%Y%m%d ->
>>> /usr/local/var/log/radius/radacct/127.0.0.1/auth-detail-20110803
>>> [auth_log]
>>> /usr/local/var/log/radius/radacct/%{Client-IP-Address}/auth-detail-%Y%m%d
>>> expands to /usr/local/var/log/radius/radacct/
>>> 127.0.0.1/auth-detail-20110803
>>> [auth_log]      expand: %t -> Wed Aug  3 12:06:33 2011
>>> ++[auth_log] returns ok
>>> ++[chap] returns noop
>>> ++[mschap] returns noop
>>> ++[digest] returns noop
>>> [suffix] No '@' in User-Name = "1001", looking up realm NULL
>>> [suffix] No such realm "NULL"
>>> ++[suffix] returns noop
>>> [eap] No EAP-Message, not doing EAP
>>> ++[eap] returns noop
>>> ++[unix] returns notfound
>>> ++[files] returns noop
>>> [sql]   expand: %{User-Name} -> 1001
>>> [sql] sql_set_user escaped user --> '1001'
>>> rlm_sql (sql): Reserving sql socket id: 4
>>> [sql]   expand: SELECT id, username, attribute, value, op           FROM
>>> radcheck           WHERE username = '%{SQL-User-Name}'           ORDER BY id
>>> -> SELECT id, username, attribute, value, op           FROM
>>> radcheck           WHERE username = '1001'           ORDER BY id
>>> [sql]   expand: SELECT groupname           FROM radusergroup
>>> WHERE username = '%{SQL-User-Name}'           ORDER BY priority -> SELECT
>>> groupname           FROM radusergroup           WHERE username =
>>> '1001'           ORDER BY priority
>>> rlm_sql (sql): Released sql socket id: 4
>>> [sql] User 1001 not found
>>> ++[sql] returns notfound
>>> ++[expiration] returns noop
>>> ++[logintime] returns noop
>>> [pap] WARNING! No "known good" password found for the user.
>>> Authentication may fail because of this.
>>> ++[pap] returns noop
>>> ERROR: No authenticate method (Auth-Type) found for the request:
>>> Rejecting the user
>>> Failed to authenticate the user.
>>>   WARNING: Unprintable characters in the password.        Double-check
>>> the shared secret on the server and the NAS!
>>> Using Post-Auth-Type Reject
>>> # Executing group from file /usr/local/etc/raddb/sites-enabled/default
>>> +- entering group REJECT {...}
>>> [attr_filter.access_reject]     expand: %{User-Name} -> 1001
>>>  attr_filter: Matched entry DEFAULT at line 11
>>> ++[attr_filter.access_reject] returns updated
>>> Delaying reject of request 8 for 1 seconds
>>> Going to the next request
>>> Waking up in 0.9 seconds.
>>> Sending delayed reject for request 8
>>> Sending Access-Reject of id 49 to 127.0.0.1 port 52684
>>> Waking up in 4.9 seconds.
>>> Cleaning up request 8 ID 49 with timestamp +7674
>>> Ready to process requests.
>>> WARNING! No "known good" password found for the user
>>>
>>> Regards,
>>> Charles
>>>
>>>
>>> 2011/8/3 Tihomir Culjaga <tculjaga at gmail.com>
>>>
>>>> did u use the dictionary i have attached ?
>>>>
>>>>
>>>> On Tue, Aug 2, 2011 at 10:08 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>>>>
>>>>> i tried change to 'h323-conf-id' to 'h323-call-origin' in
>>>>> 02_unitest_rad-ANI-auth.xml, rad_auth.conf.xml, however, it still prompt
>>>>> '[ERR] mod_rad_auth.c:428 Unknown attribute: key:h323-conf-id, not
>>>>> found in dictionary', so where the mod_rad_auth read out the 'h323-conf-id'?
>>>>> very very strange, which dictionary it was using...
>>>>>
>>>>> Regards,
>>>>> Charles
>>>>>
>>>>>
>>>>> 2011/8/2 fieldpeak <fieldpeak at gmail.com>
>>>>>
>>>>>> Hi Tihomir,
>>>>>>
>>>>>> Finally the answer coming, i see the hope, thanks for your reply, :)
>>>>>>
>>>>>> As your advise, i only use one attribute(h323-conf-id) in my dialplan,
>>>>>> and only one attribute(h323-conf-id) in rad_auth.conf.xml, and using the
>>>>>> attached dictionary (from ciso) which contains this attribute, however, it
>>>>>> still prompt 'unknown attribute', so i suspected if it was reading
>>>>>> /usr/local/etc/radiusclient/dictionary, so i copy the same dictionary to
>>>>>> /usr/local/freeswitch/radius/, it did not any help at all... very strange...
>>>>>>
>>>>>> Log:
>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set
>>>>>> default_realm := .
>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set
>>>>>> radius_timeout := 3.
>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set
>>>>>> radius_retries := 2.
>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set
>>>>>> radius_deadtime := 0.
>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set bindaddr :=
>>>>>> *.
>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:371 ... radius:
>>>>>> User-Name: 38516060333
>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:380 ... radius:
>>>>>> User-Password: 003282
>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:396 ... radius:
>>>>>> Called-station-Id: 16094191500
>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:413 Handle
>>>>>> attribute: h323-conf-id
>>>>>> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:428 Unknown attribute:
>>>>>> key:h323-conf-id, not found in dictionary
>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:538 abort sending
>>>>>> radius packet.
>>>>>> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:546 An error occured
>>>>>> during RADIUS Authentication(RC=-1)
>>>>>> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:702 An error occured
>>>>>> during radius authorization.
>>>>>>
>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  AUTH_RESULT=)
>>>>>>
>>>>>>
>>>>>>
>>>>>>   <extension name="unitest_rad-ANI-auth">
>>>>>>     <condition field="destination_number" expression="^601$">
>>>>>>       <!-- <action application="log" data="INFO  Before Auth "/> -->
>>>>>>
>>>>>>       <action inline="true" application="set" data="CALLID=
>>>>>> h323-conf-id=${uuid}"/>
>>>>>>
>>>>>>       <action inline="true" application="set" data="USERNAME=1001"/>
>>>>>>       <action inline="true" application="set" data="PASSWD=1111"/>
>>>>>>
>>>>>>
>>>>>>       <action application="sleep" data="2000"/>
>>>>>>       <action application="auth_function" data="in ${DIALED_NUMBER},
>>>>>> in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>
>>>>>>
>>>>>>     </condition>
>>>>>>   </extension>
>>>>>>
>>>>>>
>>>>>>
>>>>>> <configuration name="rad_auth.conf" description="radius
>>>>>> authentification module">
>>>>>>   <settings>
>>>>>>
>>>>>>   </settings>
>>>>>>
>>>>>>   <client>
>>>>>>     <param name="authserver" value="127.0.0.1:1812:gateway"/>
>>>>>>     <param name="dictionary"
>>>>>> value="/usr/local/etc/radiusclient/dictionary"/>
>>>>>>     <param name="seqfile" value="/var/run/radius.seq"/>
>>>>>>     <param name="mapfile"
>>>>>> value="/usr/local/etc/radiusclient/port-id-map"/>
>>>>>>     <param name="default_realm" value=""/>
>>>>>>     <param name="radius_timeout" value="3"/>
>>>>>>     <param name="radius_retries" value="2"/>
>>>>>>     <param name="radius_deadtime" value="0"/>
>>>>>>     <param name="bindaddr" value="*"/>
>>>>>>   </client>
>>>>>>
>>>>>>   <vsas>
>>>>>>
>>>>>>
>>>>>>     <param name="h323-conf-id" id="24" value="CALLID" pec="9"
>>>>>> expr="1" direction="in"/>
>>>>>>
>>>>>>   </vsas>
>>>>>>  </configuration>
>>>>>>
>>>>>>
>>>>>>
>>>>>> 2011/8/2 Tihomir Culjaga <tculjaga at gmail.com>
>>>>>>
>>>>>>> hi,
>>>>>>>
>>>>>>> dictionary.all is just the name of a file containing all attributes i
>>>>>>> needed at that time.
>>>>>>>
>>>>>>> you can include other dictionaries by putting #INCLUDE <pathname> at
>>>>>>> the end of the dictionary file you reference in rad_auth.conf.xml.
>>>>>>> if the INCLUDE doesn't work, just append dictionary.cisco to your
>>>>>>> dictionary file... and make your own file.
>>>>>>>
>>>>>>>
>>>>>>> check inline comments down below...
>>>>>>>
>>>>>>>
>>>>>>> T.
>>>>>>>
>>>>>>>
>>>>>>> On Sun, Jul 31, 2011 at 10:46 AM, fieldpeak <fieldpeak at gmail.com>wrote:
>>>>>>>
>>>>>>>> Hello Gurus,
>>>>>>>>
>>>>>>>> i met a issue when using
>>>>>>>> mod_rad_auth(http://wiki.freeswitch.org/wiki/Mod_rad_auth) to works
>>>>>>>> with freeradius server+mysql for AAA, the details is below, Could
>>>>>>>> anyone give any hints, Thanks in advance.
>>>>>>>>
>>>>>>>> i setup a dial plan "unitest_rad-ANI-auth" as wiki above, however,
>>>>>>>> when i dialed 601 to trigger the dial plan, the console show errors,
>>>>>>>> it looks "h323-conf-id" is not in the directory, then i tried to add
>>>>>>>> this attribute to the dictionary, however, it does not help, in the
>>>>>>>> wiki, it mentioned the rad_auth.conf.xml contains <param
>>>>>>>> name="dictionary"
>>>>>>>> value="/usr/local/etc/radiusclient/dictionary.all"/>, however i did
>>>>>>>> not find the file "dictionary.all" at that directory, so i use
>>>>>>>> dictionary. BTW, the freeradius server + mysql works well.
>>>>>>>>
>>>>>>>
>>>>>>> i just appended the information needed into dictionary.all file...
>>>>>>> (vendor and attribute definition).
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>>
>>>>>>>> console errors:
>>>>>>>>
>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 auth_function(in , in
>>>>>>>> 38516060333, in 003282, out AUTH_RESULT)
>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:301 allocate
>>>>>>>> initial
>>>>>>>> structure.
>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:313 initialzed
>>>>>>>> configuration.
>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set authserver
>>>>>>>> := 127.0.0.1:1812:gateway.
>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set dictionary
>>>>>>>> := /usr/local/etc/radiusclient/dictionary.
>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set seqfile :=
>>>>>>>> /var/run/radius.seq.
>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set mapfile :=
>>>>>>>> /usr/local/etc/radiusclient/port-id-map.
>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>> default_realm := .
>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>> radius_timeout := 3.
>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>> radius_retries := 2.
>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>> radius_deadtime := 0.
>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set bindaddr
>>>>>>>> := *.
>>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:371 ... radius:
>>>>>>>> User-Name: 38516060333
>>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:380 ... radius:
>>>>>>>> User-Password: 003282
>>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:391 ... radius:
>>>>>>>> Called-station-Id is empty, ignoring...
>>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:413 Handle
>>>>>>>> attribute: h323-conf-id
>>>>>>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:428 Unknown
>>>>>>>> attribute:
>>>>>>>> key:h323-conf-id, not found in dictionary
>>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:538 abort sending
>>>>>>>> radius packet.
>>>>>>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:546 An error occured
>>>>>>>> during RADIUS Authentication(RC=-1)
>>>>>>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:702 An error occured
>>>>>>>> during radius authorization.
>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  AUTH_RESULT=)
>>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  AUTH_RESULT=
>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO
>>>>>>>>  billing_model=)
>>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  billing_model=
>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO
>>>>>>>>  credit_amount=)
>>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  credit_amount=
>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  currency=)
>>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  currency=
>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO
>>>>>>>>  preffered_lang=)
>>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202
>>>>>>>>  preffered_lang=
>>>>>>>>
>>>>>>>> added below in the
>>>>>>>> dictionary(/usr/local/etc/radiusclient/dictionary):
>>>>>>>>
>>>>>>>> ATTRIBUTE       h323-conf-id            1008    string
>>>>>>>>
>>>>>>>
>>>>>>> you need the vendor definition as well
>>>>>>>
>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> dial plan:
>>>>>>>> <extension name="unitest_rad-ANI-auth">
>>>>>>>>    <condition field="destination_number" expression="^601$">
>>>>>>>>      <action application="log" data="INFO  Before Auth "/>
>>>>>>>>
>>>>>>>>      <action inline="true" application="set"
>>>>>>>> data="CALLID=h323-conf-id=${uuid}"/>
>>>>>>>>      <action inline="true" application="set"
>>>>>>>> data="SERVICENUM=h323-prompt-id=${destination_number}"/>
>>>>>>>>      <action inline="true" application="set"
>>>>>>>> data="TRANSACTIONID=h323-ivr-out=transactionID:1234"/>
>>>>>>>>  <!--      <action inline="true" application="set"
>>>>>>>> data="CALLINGNUMBER=${caller_id_number}"/> -->
>>>>>>>>      <action inline="true" application="set"
>>>>>>>> data="CALLINGNUMBER=38516060333"/>
>>>>>>>>      <action inline="true" application="set"
>>>>>>>> data="USERNAME=38516060333"/>
>>>>>>>>  <!--      <action inline="true" application="set"
>>>>>>>> data="USERNAME=209354"/> -->
>>>>>>>>      <action inline="true" application="set" data="PASSWD=003282"/>
>>>>>>>>  <!--      <action inline="true" application="set"
>>>>>>>> data="DIALED_NUMBER=16094191500"/>  -->
>>>>>>>>
>>>>>>>>      <action application="sleep" data="2000"/>
>>>>>>>>      <action application="auth_function" data="in ${DIALED_NUMBER},
>>>>>>>> in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>
>>>>>>>>
>>>>>>>>
>>>>>>>>      <action application="log" data="INFO
>>>>>>>>  AUTH_RESULT=${AUTH_RESULT}"/>
>>>>>>>>      <action application="log" data="INFO
>>>>>>>>  billing_model=${billing_model}"/>
>>>>>>>>      <action application="log" data="INFO
>>>>>>>>  credit_amount=${credit_amount}"/>
>>>>>>>>      <action application="log" data="INFO  currency=${currency}"/>
>>>>>>>>      <action application="log" data="INFO
>>>>>>>>  preffered_lang=${preffered_lang}"/>
>>>>>>>>      <action application="log" data="INFO
>>>>>>>>  credit_time=${credit_time}"/>
>>>>>>>>      <action application="log" data="INFO
>>>>>>>> h323_ivr_duration=${h323_ivr_duration}"/>
>>>>>>>>      <action application="log" data="INFO
>>>>>>>>  return_code=${return_code}"/>
>>>>>>>>      <!-- <action application="execute_extension" data="AUTH XML
>>>>>>>> default"/> -->
>>>>>>>>    </condition>
>>>>>>>>  </extension>
>>>>>>>>
>>>>>>>>  radius_cdr.conf.xml:
>>>>>>>>  <configuration name="radius_cdr.conf" description="RADIUS CDR
>>>>>>>> Configuration">
>>>>>>>>
>>>>>>>>        <settings>
>>>>>>>>
>>>>>>>>                <!-- location of the radius dictionary files -->
>>>>>>>>
>>>>>>>>                <param name="dictionary"
>>>>>>>> value="/usr/local/freeswitch/conf/radius/dictionary"/>
>>>>>>>>
>>>>>>>>
>>>>>>> your dictionary file need to contain all the attributes you are
>>>>>>> trying to use or to include other dictionaries (In this case
>>>>>>> dictionary.cisco) from the dictionary file you are referencing here.
>>>>>>>
>>>>>>>
>>>>>>>>                <!-- number of retries for each server -->
>>>>>>>>
>>>>>>>>                <param name="radius_retries" value="3"/>
>>>>>>>>
>>>>>>>>                <!-- number of seconds to wait between retries -->
>>>>>>>>
>>>>>>>>                <param name="radius_timeout" value="5"/>
>>>>>>>>
>>>>>>>>                <!-- accounting servers, up to 8 allowed -->
>>>>>>>>
>>>>>>>>                <!-- value is "host:port:secret", port is optional
>>>>>>>> -->
>>>>>>>>
>>>>>>>>                <!-- use IP ADDRESSES, not hostnames -->
>>>>>>>>
>>>>>>>>                <param name="acct_server" value="127.0.0.1:1813
>>>>>>>> :testing123"/>
>>>>>>>>
>>>>>>>>
>>>>>>>>        </settings>
>>>>>>>>
>>>>>>>> </configuration>
>>>>>>>>
>>>>>>>>  the FS version:
>>>>>>>>  FreeSWITCH Version 1.0.head (git-492bc6b 2011-07-23 12-53-04 -0400)
>>>>>>>>
>>>>>>>>  Regards,
>>>>>>>>  Charles
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110806/b02ec27b/attachment-0001.html 

From lloydie.t at gmail.com  Sat Aug  6 20:57:12 2011
From: lloydie.t at gmail.com (lloyd thomas)
Date: Sat, 6 Aug 2011 17:57:12 +0100
Subject: [Freeswitch-users] outgoing SIP profiles setup
Message-ID: <CAE=3qOYs=VRBaDf8ho6x44GnxTZVktZ7AzoxGBfxv7GF8XR-pQ@mail.gmail.com>

Hi All,
I am trying to create a new sip profile as my provider does not use sip
registrations, but uses IP address and port number to identify wether a
connection should be allowed.
I am currently using the 'internal' profile to make calls '<action
application="bridge" data="sofia/internal/$1 at 80.40.150.150"/>', but this is
causing a couple of strange anomalies.
I have created a new xml file in sip_profiles and use 'sofia profile
phisys-2circles start' to add it.
But I get the following errors and the profile does not show in 'sofia
status'

errors-----------------------------------------------
2011-08-06 17:38:28.620292 [NOTICE] sofia_reg.c:2775 Added gateway '
example.com' to profile 'phisys-2circles'
2011-08-06 17:38:28.620292 [WARNING] sofia.c:1898 Ignoring duplicate gateway
'example.com'
2011-08-06 17:38:28.624397 [WARNING] sofia.c:1898 Ignoring duplicate gateway
'example.com'
2011-08-06 17:38:28.625645 [NOTICE] sofia.c:3999 Started Profile
phisys-2circles [sofia_reg_phisys-2circles]
2011-08-06 17:38:28.643631 [DEBUG] sofia.c:1471 Creating agent for
phisys-2circles
2011-08-06 17:38:28.800994 [ERR] sofia.c:1533 Error Creating SIP UA for
profile: phisys-2circles
2011-08-06 17:38:28.800994 [NOTICE] sofia_glue.c:5198 deleted gateway
example.com

I have added phisys_2circles.xml to the sip_profiles directory and the
contents are here http://pastebin.freeswitch.org/16998
Any advice welcome as I am at a lost what to do next.

Thanks

Lloyd T
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110806/b0266e4c/attachment.html 

From msc at freeswitch.org  Sat Aug  6 22:01:49 2011
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 6 Aug 2011 11:01:49 -0700
Subject: [Freeswitch-users] outgoing SIP profiles setup
In-Reply-To: <CAE=3qOYs=VRBaDf8ho6x44GnxTZVktZ7AzoxGBfxv7GF8XR-pQ@mail.gmail.com>
References: <CAE=3qOYs=VRBaDf8ho6x44GnxTZVktZ7AzoxGBfxv7GF8XR-pQ@mail.gmail.com>
Message-ID: <CAKzWOxUP-BY9_7nx5USmFHaq9X8b-9BiL3sYWQXwNjfmUJLBgg@mail.gmail.com>

Lloyd,

It looks to me like you made a copy of internal.xml or external.xml - which
is okay - I just want to make sure we know where you are coming from. My
initial guess based on what I see is that you may already have the internal
profile running on the same IP/port that you're trying to bind to.

Question: Is your FS box behind NAT? That's the most important thing to nail
down. Next thing is whether your FS box has only one NIC.

Last question: what are the "strange anomalies" that occur when using the
internal profile?

-MC

On Sat, Aug 6, 2011 at 9:57 AM, lloyd thomas <lloydie.t at gmail.com> wrote:

> Hi All,
> I am trying to create a new sip profile as my provider does not use sip
> registrations, but uses IP address and port number to identify wether a
> connection should be allowed.
> I am currently using the 'internal' profile to make calls '<action
> application="bridge" data="sofia/internal/$1 at 80.40.150.150"/>', but this
> is causing a couple of strange anomalies.
> I have created a new xml file in sip_profiles and use 'sofia profile
> phisys-2circles start' to add it.
> But I get the following errors and the profile does not show in 'sofia
> status'
>
> errors-----------------------------------------------
> 2011-08-06 17:38:28.620292 [NOTICE] sofia_reg.c:2775 Added gateway '
> example.com' to profile 'phisys-2circles'
> 2011-08-06 17:38:28.620292 [WARNING] sofia.c:1898 Ignoring duplicate
> gateway 'example.com'
> 2011-08-06 17:38:28.624397 [WARNING] sofia.c:1898 Ignoring duplicate
> gateway 'example.com'
> 2011-08-06 17:38:28.625645 [NOTICE] sofia.c:3999 Started Profile
> phisys-2circles [sofia_reg_phisys-2circles]
> 2011-08-06 17:38:28.643631 [DEBUG] sofia.c:1471 Creating agent for
> phisys-2circles
> 2011-08-06 17:38:28.800994 [ERR] sofia.c:1533 Error Creating SIP UA for
> profile: phisys-2circles
> 2011-08-06 17:38:28.800994 [NOTICE] sofia_glue.c:5198 deleted gateway
> example.com
>
> I have added phisys_2circles.xml to the sip_profiles directory and the
> contents are here http://pastebin.freeswitch.org/16998
> Any advice welcome as I am at a lost what to do next.
>
> Thanks
>
> Lloyd T
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110806/1fd265a4/attachment.html 

From msc at freeswitch.org  Sat Aug  6 22:15:14 2011
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 6 Aug 2011 11:15:14 -0700
Subject: [Freeswitch-users] conference count
In-Reply-To: <CAE2ghw7X+cmcagb_PFQ0GZrTdYV+=-3iJJEXku-XoknvDqKmRw@mail.gmail.com>
References: <CAE2ghw7X+cmcagb_PFQ0GZrTdYV+=-3iJJEXku-XoknvDqKmRw@mail.gmail.com>
Message-ID: <CAKzWOxXsSOnWuB_NicZAGKNwrzBeXYd8HZsbxY6s=fjXjmcSMQ@mail.gmail.com>

Sam,

My guess is that this line is causing you trouble:

<action application="set" data="conf_count=${conference ${conference_name}
list count}"/>

>From what I see in your logs and dialplan I have to assume that
${conference_name} is not populated with the name of your conference so your
API call is not working. What's actually happening is that this API is
getting called:

conference list count

instead of

conference 1000146 list count

Try them both at the fs_cli to see the difference in output. ;) (Hint: look
at the log line that contains "set(conf_count=" and you'll see it's getting
populated with all sorts of stuff instead of just the conference count)

If you know beforehand what the name of the conference is then just plug it
in:

<action application="set" data="conf_count=${conference 1000146 list
count}"/>

FYI, the ${conference_name} variable won't be populated until the current
channel is already a member of the conference, so it doesn't help you at
this point in the dialplan. Fortunately, though, since you are transferring
the caller into a conference room you already know what conference he's
going into so just use that information instead of the conference_name chan
var.

-MC

On Sat, Aug 6, 2011 at 2:36 AM, Sam <u2nsam at gmail.com> wrote:

> *I have a dial plan ,*
>
> <extension name="dynamic_conferences_guest">
>       <condition field="destination_number" expression="^1234">
>     <action application="answer"/>
>     <action application="info"/>
>         <action application="set" data="conf_count=${conference
> ${conference_name} list count}"/>
>         <action application="log" data="INFO ${conf_count}"/>
>         <action application="say" data="en number pronounced
> ${conf_count}"/>
>     <action application="conference" data="1000146 at meeting"/>
>       </condition>
>     </extension>
>
>
> *and the output of it as below:-*
>
> 2011-08-06 14:32:53.205895 [DEBUG] sofia.c:5084 Channel sofia/internal/
> 61323123999 at 192.153.53.158 entering state [ready][200]
> EXECUTE sofia/internal/61323123999 at 192.153.53.158set(conf_count=Conference 1000146 (1 member rate: 16000)
> 14;sofia/internal/883510001282001 at sip.ntfone.in
> ;97f5f386-c00a-11e0-9517-738827116f37;883510001282001;883510001282001;hear|speak|floor;0;0;0;300
> )
> 2011-08-06 14:32:53.205895 [DEBUG] mod_dptools.c:1063 sofia/internal/
> 61323123999 at 192.153.53.158 SET [conf_count]=[Conference 1000146 (1 member
> rate: 16000)
> 14;sofia/internal/883510001282001 at sip.nfone.in
> ;97f5f386-c00a-11e0-9517-738827116f37;883510001282001;883510001282001;hear|speak|floor;0;0;0;300
> ]
> EXECUTE sofia/internal/61323123999 at 192.153.53.158 log(INFO Conference
> 1000146 (1 member rate: 16000)
> 14;sofia/internal/883510001282001 at sip.ntfone.in
> ;97f5f386-c00a-11e0-9517-738827116f37;883510001282001;883510001282001;hear|speak|floor;0;0;0;300
> )
> 2011-08-06 14:32:53.205895 [INFO] mod_dptools.c:1202 Conference 1000146 (1
> member rate: 16000)
> 14;sofia/internal/883510001282001 at sip.nfone.in
> ;97f5f386-c00a-11e0-9517-738827116f37;883510001282001;883510001282001;hear|speak|floor;0;0;0;300
>
> EXECUTE sofia/internal/61323123999 at 192.153.53.158 say(en number pronounced
> Conference 1000146 (1 member rate: 16000)
> 14;sofia/internal/883510001282001 at sip.nfone.in
> ;97f5f386-c00a-11e0-9517-738827116f37;883510001282001;883510001282001;hear|speak|floor;0;0;0;300
> )
> 2011-08-06 14:32:53.215905 [ERR] mod_say_en.c:130 Parse Error!
>
> EXECUTE sofia/internal/61323123999 at 192.153.53.158conference(1000146 at meeting
> )
>
>
>
>
> *What should be my dialplan to get the exact conf_count ?*
>
>
> Thanks in advance
> Sam
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110806/6d59220c/attachment.html 

From lloydie.t at gmail.com  Sun Aug  7 01:14:09 2011
From: lloydie.t at gmail.com (lloyd thomas)
Date: Sat, 6 Aug 2011 22:14:09 +0100
Subject: [Freeswitch-users] outgoing SIP profiles setup
In-Reply-To: <CAKzWOxUP-BY9_7nx5USmFHaq9X8b-9BiL3sYWQXwNjfmUJLBgg@mail.gmail.com>
References: <CAE=3qOYs=VRBaDf8ho6x44GnxTZVktZ7AzoxGBfxv7GF8XR-pQ@mail.gmail.com>
	<CAKzWOxUP-BY9_7nx5USmFHaq9X8b-9BiL3sYWQXwNjfmUJLBgg@mail.gmail.com>
Message-ID: <CAE=3qOZeU81BbeBXoTi-X01i5GrzTR=Qy2VauofXYcyH_2o_-A@mail.gmail.com>

The FS box is on a public IP address, but phones are behind a NAT. I have
used a copy of the external profile. The FS box has only one NIC ( I'm using
an upgraded HP terminal services device).
The anamolies I get is when I dial out the effective_caller_id_number is
sent instead of the outbound_caller_id_number. In fact I just realised that
the is more of a problem in that the gateways are not registering.

On 6 August 2011 19:01, Michael Collins <msc at freeswitch.org> wrote:

> Lloyd,
>
> It looks to me like you made a copy of internal.xml or external.xml - which
> is okay - I just want to make sure we know where you are coming from. My
> initial guess based on what I see is that you may already have the internal
> profile running on the same IP/port that you're trying to bind to.
>
> Question: Is your FS box behind NAT? That's the most important thing to
> nail down. Next thing is whether your FS box has only one NIC.
>
> Last question: what are the "strange anomalies" that occur when using the
> internal profile?
>
> -MC
>
> On Sat, Aug 6, 2011 at 9:57 AM, lloyd thomas <lloydie.t at gmail.com> wrote:
>
>> Hi All,
>> I am trying to create a new sip profile as my provider does not use sip
>> registrations, but uses IP address and port number to identify wether a
>> connection should be allowed.
>> I am currently using the 'internal' profile to make calls '<action
>> application="bridge" data="sofia/internal/$1 at 80.40.150.150"/>', but this
>> is causing a couple of strange anomalies.
>> I have created a new xml file in sip_profiles and use 'sofia profile
>> phisys-2circles start' to add it.
>> But I get the following errors and the profile does not show in 'sofia
>> status'
>>
>> errors-----------------------------------------------
>> 2011-08-06 17:38:28.620292 [NOTICE] sofia_reg.c:2775 Added gateway '
>> example.com' to profile 'phisys-2circles'
>> 2011-08-06 17:38:28.620292 [WARNING] sofia.c:1898 Ignoring duplicate
>> gateway 'example.com'
>> 2011-08-06 17:38:28.624397 [WARNING] sofia.c:1898 Ignoring duplicate
>> gateway 'example.com'
>> 2011-08-06 17:38:28.625645 [NOTICE] sofia.c:3999 Started Profile
>> phisys-2circles [sofia_reg_phisys-2circles]
>> 2011-08-06 17:38:28.643631 [DEBUG] sofia.c:1471 Creating agent for
>> phisys-2circles
>> 2011-08-06 17:38:28.800994 [ERR] sofia.c:1533 Error Creating SIP UA for
>> profile: phisys-2circles
>> 2011-08-06 17:38:28.800994 [NOTICE] sofia_glue.c:5198 deleted gateway
>> example.com
>>
>> I have added phisys_2circles.xml to the sip_profiles directory and the
>> contents are here http://pastebin.freeswitch.org/16998
>> Any advice welcome as I am at a lost what to do next.
>>
>> Thanks
>>
>> Lloyd T
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110806/50517c32/attachment-0001.html 

From mikem at open.com.au  Sun Aug  7 03:48:41 2011
From: mikem at open.com.au (Mike McCauley)
Date: Sat, 6 Aug 2011 23:48:41 +0000 (UTC)
Subject: [Freeswitch-users] Radius/Authentication/Authorization
References: <COL109-W15576D4B6E4D7BFFFFEDD7F41F0@phx.gbl>
	<COL109-W11819E0B21AF7600E30CD4F41B0@phx.gbl>
	<1289153558035-5714752.post@n2.nabble.com>
Message-ID: <loom.20110807T014640-137@post.gmane.org>

aekas <ak at ...> writes:

> 
> 
> I'm currently successfully compiled & installed mod_rad_auth, but get some
> error:
> 
> 2010-11-07 23:00:52.610078 [CRIT] switch_loadable_module.c:928 Error Loading
> module /usr/local/freeswitch/mod/mod_rad_auth.so
> **/usr/local/freeswitch/mod/mod_rad_auth.so: undefined symbol: rc_conf_str**
> 
> FS ver.:
> FreeSWITCH Version 1.0.head (git-46a9fa3 2010-11-06 17-14-31 -0400)
> 
> OS:
>  # uname -a
> Linux lambda.core 2.6.33.6-147.fc13.i686.PAE #1 SMP Tue Jul 6 22:24:44 UTC
> 2010 i686 i686 i386 GNU/Linux
> 
> Some ideas?
> 
> Thanks.

I saw the same problem with freeswitch 1.0.7 from latest git on Ubuntu 10

The problem is in the Makefile for mod_rad_auth 

need to edit
/usr/local/src/freeswitch/src/mod/applications/mod_rad_auth/Makefile 
and change
FREERADIUSLA=/usr/local/lib/libfreeradius-client.so
to
FREERADIUSLA=/usr/local/lib/libfreeradius-client.la
make; make install




From dome at tel.co.th  Sun Aug  7 09:57:07 2011
From: dome at tel.co.th (Dome Charoenyost)
Date: Sun, 7 Aug 2011 12:57:07 +0700
Subject: [Freeswitch-users] FS for SIP registrar server
Message-ID: <CAMFECSBdmV-EB05u75hnhXH4-ETZPFvJa-S7ARPsEpS=J4dVkQ@mail.gmail.com>

Dear All,
              I use FS for my wholesale solution. it's work fine over
2000 concurrent in peak hr. so in experience when i use FS for SIP
registrar server with out ODBC (but use tmp for db path) i can handle
12,000 active in 1 server (25% CPU). All user send register packet
every 3 minutes. Now i need system for handle 500,000 subscriber I
have 2 choice
               1.  use Kamailio for SIP regitrar and use FS for
routing and billing
               2. still use multiple FS with ODBC

So anyone can help?

BG

Dome C.


From avi at avimarcus.net  Sun Aug  7 10:08:06 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Sun, 7 Aug 2011 09:08:06 +0300
Subject: [Freeswitch-users] FS for SIP registrar server
In-Reply-To: <CAMFECSBdmV-EB05u75hnhXH4-ETZPFvJa-S7ARPsEpS=J4dVkQ@mail.gmail.com>
References: <CAMFECSBdmV-EB05u75hnhXH4-ETZPFvJa-S7ARPsEpS=J4dVkQ@mail.gmail.com>
Message-ID: <CALNZuwn3V-eqxMBuosXusLv-wC8dDOJvaaxajqn6undJ4653YA@mail.gmail.com>

Do you need 3 minute registration?
-Avi


On Sun, Aug 7, 2011 at 8:57 AM, Dome Charoenyost <dome at tel.co.th> wrote:

> Dear All,
>              I use FS for my wholesale solution. it's work fine over
> 2000 concurrent in peak hr. so in experience when i use FS for SIP
> registrar server with out ODBC (but use tmp for db path) i can handle
> 12,000 active in 1 server (25% CPU). All user send register packet
> every 3 minutes. Now i need system for handle 500,000 subscriber I
> have 2 choice
>               1.  use Kamailio for SIP regitrar and use FS for
> routing and billing
>               2. still use multiple FS with ODBC
>
> So anyone can help?
>
> BG
>
> Dome C.
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110807/ea94e0ac/attachment.html 

From dome at tel.co.th  Sun Aug  7 10:48:43 2011
From: dome at tel.co.th (Dome Charoenyost)
Date: Sun, 7 Aug 2011 13:48:43 +0700
Subject: [Freeswitch-users] FS for SIP registrar server
In-Reply-To: <CALNZuwn3V-eqxMBuosXusLv-wC8dDOJvaaxajqn6undJ4653YA@mail.gmail.com>
References: <CAMFECSBdmV-EB05u75hnhXH4-ETZPFvJa-S7ARPsEpS=J4dVkQ@mail.gmail.com>
	<CALNZuwn3V-eqxMBuosXusLv-wC8dDOJvaaxajqn6undJ4653YA@mail.gmail.com>
Message-ID: <CAMFECSAtRORi7nt0MhS3hsDCmnpQJBXnt=y5vV7kKXAN3fq0oQ@mail.gmail.com>

2011/8/7 Avi Marcus <avi at avimarcus.net>:
> Do you need 3 minute registration?
Yes.

> -Avi
>
> On Sun, Aug 7, 2011 at 8:57 AM, Dome Charoenyost <dome at tel.co.th> wrote:
>>
>> Dear All,
>> ? ? ? ? ? ? ?I use FS for my wholesale solution. it's work fine over
>> 2000 concurrent in peak hr. so in experience when i use FS for SIP
>> registrar server with out ODBC (but use tmp for db path) i can handle
>> 12,000 active in 1 server (25% CPU). All user send register packet
>> every 3 minutes. Now i need system for handle 500,000 subscriber I
>> have 2 choice
>> ? ? ? ? ? ? ? 1. ?use Kamailio for SIP regitrar and use FS for
>> routing and billing
>> ? ? ? ? ? ? ? 2. still use multiple FS with ODBC
>>
>> So anyone can help?
>>
>> BG
>>
>> Dome C.
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From simon0922 at gmail.com  Sun Aug  7 11:55:37 2011
From: simon0922 at gmail.com (Simon Leck)
Date: Sun, 7 Aug 2011 15:55:37 +0800
Subject: [Freeswitch-users] media proxy
Message-ID: <004b01cc54d7$6630fc90$3292f5b0$@gmail.com>

Hi guys,

Has anybody been successful in trying to get mediaproxy to work with
Freeswitch? If so, it would be great if any of you could give us leads on
how this is been done. 

Mediaproxy URL 

http://mediaproxy.ag-projects.com/

Thanks a lot. 
Simon 





From lakersman2006 at yahoo.com  Sun Aug  7 12:54:50 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Sun, 7 Aug 2011 01:54:50 -0700 (PDT)
Subject: [Freeswitch-users] Hangup time is blank
Message-ID: <1312707290.15544.YahooMailNeo@web161006.mail.bf1.yahoo.com>

How come the "hangup_time" channel variable is 0 even after a call has been bridged successfully and hung up? I am trying to find out what time the bridged call hung up by looking at the channel variable "hangup_time".
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110807/592ff4df/attachment.html 

From jaybinks at gmail.com  Sun Aug  7 14:41:01 2011
From: jaybinks at gmail.com (Jay Binks)
Date: Sun, 7 Aug 2011 20:41:01 +1000
Subject: [Freeswitch-users] media proxy
In-Reply-To: <004b01cc54d7$6630fc90$3292f5b0$@gmail.com>
References: <004b01cc54d7$6630fc90$3292f5b0$@gmail.com>
Message-ID: <E942D570-B631-44E3-B753-60B602B2EC3F@gmail.com>

Question is why ???
Proxy_media mode would do the same in Freeswitch ..



On 07/08/2011, at 5:55 PM, "Simon Leck" <simon0922 at gmail.com> wrote:

> Hi guys,
> 
> Has anybody been successful in trying to get mediaproxy to work with
> Freeswitch? If so, it would be great if any of you could give us leads on
> how this is been done. 
> 
> Mediaproxy URL 
> 
> http://mediaproxy.ag-projects.com/
> 
> Thanks a lot. 
> Simon 
> 
> 
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From gavin.henry at gmail.com  Mon Aug  8 01:33:24 2011
From: gavin.henry at gmail.com (Gavin Henry)
Date: Sun, 7 Aug 2011 22:33:24 +0100
Subject: [Freeswitch-users] openssl deprecated on OS X Lion and MacPorts
	with latest Git head?
Message-ID: <CAA8_NKAgfcJWdLLBBih_wL6vmZbNAL--q+PT9Q2FP-nf7arFaQ@mail.gmail.com>

Hi all,

Anyone seeing this? My port tree is up to date and on latest MacPorts:

In file included from src/switch_core.c:39:
./src/include/switch_ssl.h: In function ?switch_ssl_init_ssl_locks?:
./src/include/switch_ssl.h:63: warning: ?CRYPTO_num_locks? is
deprecated (declared at /usr/include/openssl/crypto.h:415)
./src/include/switch_ssl.h:65: warning: ?CRYPTO_malloc? is deprecated
(declared at /usr/include/openssl/crypto.h:478)
./src/include/switch_ssl.h:65: warning: ?CRYPTO_num_locks? is
deprecated (declared at /usr/include/openssl/crypto.h:415)
./src/include/switch_ssl.h:75: warning: ?CRYPTO_set_id_callback? is
deprecated (declared at /usr/include/openssl/crypto.h:425)
./src/include/switch_ssl.h:76: warning: ?CRYPTO_set_locking_callback?
is deprecated (declared at /usr/include/openssl/crypto.h:418)
./src/include/switch_ssl.h: In function ?switch_ssl_destroy_ssl_locks?:
./src/include/switch_ssl.h:91: warning: ?CRYPTO_set_locking_callback?
is deprecated (declared at /usr/include/openssl/crypto.h:418)
./src/include/switch_ssl.h:92: warning: ?CRYPTO_num_locks? is
deprecated (declared at /usr/include/openssl/crypto.h:415)
./src/include/switch_ssl.h:98: warning: ?CRYPTO_free? is deprecated
(declared at /usr/include/openssl/crypto.h:480)
make[2]: *** [libfreeswitch_la-switch_core.lo] Error 1
make[1]: *** [all] Error 2
make: *** [current] Error 2

-- 
http://www.suretecsystems.com/services/openldap/
http://www.surevoip.co.uk


From lfurrea at gmail.com  Mon Aug  8 05:38:12 2011
From: lfurrea at gmail.com (Luis F Urrea)
Date: Sun, 7 Aug 2011 19:38:12 -0600
Subject: [Freeswitch-users] Best way to route Large block of DIDs
Message-ID: <CAB2KLyd6nn0bg-Vp9R8fygjYP_1LGDC26Zhw5FDvYB4G3vOtmg@mail.gmail.com>

Hi all,

I have a relatively large block  of DIDs coming from a T1 gateway to a FS
box. These DIDs are mapped to user extensions in most cases and in others
they go to an IVR.

My question is what would be the best way to write an extension in
public.xml dialplan if the block of DIDs that go directly to users is not
exactly divided so that I can write a regexp to cover the block.

Does writing a single rule for each user can bring performance implications?

What if I cannot arrange the block of DIDs that belong to users in a
consecutive group?

What are the best practices for such case?

TIA!!!

Luis
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110807/4d6634d9/attachment.html 

From krice at freeswitch.org  Mon Aug  8 05:55:54 2011
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 07 Aug 2011 20:55:54 -0500
Subject: [Freeswitch-users] Best way to route Large block of DIDs
In-Reply-To: <CAB2KLyd6nn0bg-Vp9R8fygjYP_1LGDC26Zhw5FDvYB4G3vOtmg@mail.gmail.com>
Message-ID: <CA64AC5A.19B67%krice@freeswitch.org>

Mod_easyroute might be used for this... it uses a DB backend via ODBC...

This doesn?t really handle ranges as is, but you can do something to create
ranges in there...


On 8/7/11 8:38 PM, "Luis F Urrea" <lfurrea at gmail.com> wrote:

> Hi all,
> 
> I have a relatively large block ?of DIDs coming from a T1 gateway to a FS box.
> These DIDs are mapped to user extensions in most cases and in others they go
> to an IVR.
> 
> My question is what would be the best way to write an extension in public.xml
> dialplan if the block of DIDs that go directly to users is not exactly divided
> so that I can write a regexp to cover the block.
> 
> Does writing a single rule for each user can bring performance implications?
> 
> What if I cannot arrange the block of DIDs that belong to users in a
> consecutive group?
> 
> What are the best practices for such case?
> 
> TIA!!!
> 
> Luis
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110807/c50a64b1/attachment-0001.html 

From curriegrad2004 at gmail.com  Mon Aug  8 08:12:37 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sun, 7 Aug 2011 21:12:37 -0700
Subject: [Freeswitch-users] openssl deprecated on OS X Lion and MacPorts
 with latest Git head?
In-Reply-To: <CAA8_NKAgfcJWdLLBBih_wL6vmZbNAL--q+PT9Q2FP-nf7arFaQ@mail.gmail.com>
References: <CAA8_NKAgfcJWdLLBBih_wL6vmZbNAL--q+PT9Q2FP-nf7arFaQ@mail.gmail.com>
Message-ID: <CAMWiMy-Rxt3WQWnoUNm1Y3e7GARtieea6tbfQ4YY+uL=F5G6EQ@mail.gmail.com>

it's more like a compiler issue from what I can tell here

On Sun, Aug 7, 2011 at 2:33 PM, Gavin Henry <gavin.henry at gmail.com> wrote:
> Hi all,
>
> Anyone seeing this? My port tree is up to date and on latest MacPorts:
>
> In file included from src/switch_core.c:39:
> ./src/include/switch_ssl.h: In function ?switch_ssl_init_ssl_locks?:
> ./src/include/switch_ssl.h:63: warning: ?CRYPTO_num_locks? is
> deprecated (declared at /usr/include/openssl/crypto.h:415)
> ./src/include/switch_ssl.h:65: warning: ?CRYPTO_malloc? is deprecated
> (declared at /usr/include/openssl/crypto.h:478)
> ./src/include/switch_ssl.h:65: warning: ?CRYPTO_num_locks? is
> deprecated (declared at /usr/include/openssl/crypto.h:415)
> ./src/include/switch_ssl.h:75: warning: ?CRYPTO_set_id_callback? is
> deprecated (declared at /usr/include/openssl/crypto.h:425)
> ./src/include/switch_ssl.h:76: warning: ?CRYPTO_set_locking_callback?
> is deprecated (declared at /usr/include/openssl/crypto.h:418)
> ./src/include/switch_ssl.h: In function ?switch_ssl_destroy_ssl_locks?:
> ./src/include/switch_ssl.h:91: warning: ?CRYPTO_set_locking_callback?
> is deprecated (declared at /usr/include/openssl/crypto.h:418)
> ./src/include/switch_ssl.h:92: warning: ?CRYPTO_num_locks? is
> deprecated (declared at /usr/include/openssl/crypto.h:415)
> ./src/include/switch_ssl.h:98: warning: ?CRYPTO_free? is deprecated
> (declared at /usr/include/openssl/crypto.h:480)
> make[2]: *** [libfreeswitch_la-switch_core.lo] Error 1
> make[1]: *** [all] Error 2
> make: *** [current] Error 2
>
> --
> http://www.suretecsystems.com/services/openldap/
> http://www.surevoip.co.uk
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From u2nsam at gmail.com  Mon Aug  8 10:11:43 2011
From: u2nsam at gmail.com (Sam)
Date: Mon, 8 Aug 2011 11:41:43 +0530
Subject: [Freeswitch-users] conference count
In-Reply-To: <CAKzWOxXsSOnWuB_NicZAGKNwrzBeXYd8HZsbxY6s=fjXjmcSMQ@mail.gmail.com>
References: <CAE2ghw7X+cmcagb_PFQ0GZrTdYV+=-3iJJEXku-XoknvDqKmRw@mail.gmail.com>
	<CAKzWOxXsSOnWuB_NicZAGKNwrzBeXYd8HZsbxY6s=fjXjmcSMQ@mail.gmail.com>
Message-ID: <CAE2ghw4aKvHMVb7zVY3_Ww6vxAUX=QgWxDK8A4gZcOrfW9fqjA@mail.gmail.com>

Thanks MC , good info, it works.

Regards
Sam

On Sat, Aug 6, 2011 at 11:45 PM, Michael Collins <msc at freeswitch.org> wrote:

> Sam,
>
> My guess is that this line is causing you trouble:
>
> <action application="set" data="conf_count=${conference ${conference_name}
> list count}"/>
>
> From what I see in your logs and dialplan I have to assume that
> ${conference_name} is not populated with the name of your conference so your
> API call is not working. What's actually happening is that this API is
> getting called:
>
> conference list count
>
> instead of
>
> conference 1000146 list count
>
> Try them both at the fs_cli to see the difference in output. ;) (Hint: look
> at the log line that contains "set(conf_count=" and you'll see it's getting
> populated with all sorts of stuff instead of just the conference count)
>
> If you know beforehand what the name of the conference is then just plug it
> in:
>
> <action application="set" data="conf_count=${conference 1000146 list
> count}"/>
>
> FYI, the ${conference_name} variable won't be populated until the current
> channel is already a member of the conference, so it doesn't help you at
> this point in the dialplan. Fortunately, though, since you are transferring
> the caller into a conference room you already know what conference he's
> going into so just use that information instead of the conference_name chan
> var.
>
> -MC
>
>   On Sat, Aug 6, 2011 at 2:36 AM, Sam <u2nsam at gmail.com> wrote:
>
>>  *I have a dial plan ,*
>>
>> <extension name="dynamic_conferences_guest">
>>       <condition field="destination_number" expression="^1234">
>>     <action application="answer"/>
>>     <action application="info"/>
>>         <action application="set" data="conf_count=${conference
>> ${conference_name} list count}"/>
>>         <action application="log" data="INFO ${conf_count}"/>
>>         <action application="say" data="en number pronounced
>> ${conf_count}"/>
>>     <action application="conference" data="1000146 at meeting"/>
>>       </condition>
>>     </extension>
>>
>>
>> *and the output of it as below:-*
>>
>> 2011-08-06 14:32:53.205895 [DEBUG] sofia.c:5084 Channel sofia/internal/
>> 61323123999 at 192.153.53.158 entering state [ready][200]
>> EXECUTE sofia/internal/61323123999 at 192.153.53.158set(conf_count=Conference 1000146 (1 member rate: 16000)
>> 14;sofia/internal/883510001282001 at sip.ntfone.in
>> ;97f5f386-c00a-11e0-9517-738827116f37;883510001282001;883510001282001;hear|speak|floor;0;0;0;300
>> )
>> 2011-08-06 14:32:53.205895 [DEBUG] mod_dptools.c:1063 sofia/internal/
>> 61323123999 at 192.153.53.158 SET [conf_count]=[Conference 1000146 (1 member
>> rate: 16000)
>> 14;sofia/internal/883510001282001 at sip.nfone.in
>> ;97f5f386-c00a-11e0-9517-738827116f37;883510001282001;883510001282001;hear|speak|floor;0;0;0;300
>> ]
>> EXECUTE sofia/internal/61323123999 at 192.153.53.158 log(INFO Conference
>> 1000146 (1 member rate: 16000)
>> 14;sofia/internal/883510001282001 at sip.ntfone.in
>> ;97f5f386-c00a-11e0-9517-738827116f37;883510001282001;883510001282001;hear|speak|floor;0;0;0;300
>> )
>> 2011-08-06 14:32:53.205895 [INFO] mod_dptools.c:1202 Conference 1000146 (1
>> member rate: 16000)
>> 14;sofia/internal/883510001282001 at sip.nfone.in
>> ;97f5f386-c00a-11e0-9517-738827116f37;883510001282001;883510001282001;hear|speak|floor;0;0;0;300
>>
>> EXECUTE sofia/internal/61323123999 at 192.153.53.158 say(en number
>> pronounced Conference 1000146 (1 member rate: 16000)
>> 14;sofia/internal/883510001282001 at sip.nfone.in
>> ;97f5f386-c00a-11e0-9517-738827116f37;883510001282001;883510001282001;hear|speak|floor;0;0;0;300
>> )
>> 2011-08-06 14:32:53.215905 [ERR] mod_say_en.c:130 Parse Error!
>>
>> EXECUTE sofia/internal/61323123999 at 192.153.53.158conference(1000146 at meeting
>> )
>>
>>
>>
>>
>> *What should be my dialplan to get the exact conf_count ?*
>>
>>
>> Thanks in advance
>> Sam
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110808/7f47174b/attachment.html 

From u2nsam at gmail.com  Mon Aug  8 10:13:49 2011
From: u2nsam at gmail.com (Sam)
Date: Mon, 8 Aug 2011 11:43:49 +0530
Subject: [Freeswitch-users] event
Message-ID: <CAE2ghw4GeQp-seD-c9nxa=W-wnkFbEQW6Ji_2sNqGBuaj8n+ug@mail.gmail.com>

Hello,

What makes the below event to occour and how to stop it reccouring.

2011-08-08 11:38:11.415945 [INFO] mod_sofia.c:4919 EVENT_TRAP: IP change
detected
2011-08-08 11:38:11.415945 [INFO] mod_sofia.c:4920 IP change detected
[192.168.53.188]->[192.168.53.189] []->[]
2011-08-08 11:38:11.615887 [NOTICE] sofia_glue.c:5192 Reload XML [Success]
2011-08-08 11:38:11.615887 [INFO] mod_enum.c:775 ENUM Reloaded
2011-08-08 11:38:11.615887 [INFO] switch_time.c:1028 Timezone reloaded 530
definitions

Regards
Sam
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110808/02734ec0/attachment.html 

From fieldpeak at gmail.com  Mon Aug  8 10:18:39 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Mon, 8 Aug 2011 14:18:39 +0800
Subject: [Freeswitch-users] Mod_rad_auth issue for FS working with
 FreeRadius server
In-Reply-To: <CAJoKKdVSkQOV3Kou_ro5AQZTyaqH6r0WRE-hZdrA=ye8ARE_vQ@mail.gmail.com>
References: <CAOTmbMzvd5OHL8z=_8vu6Orp2_prGxaw1gmXYCM2=2+pgzs_WA@mail.gmail.com>
	<CAJoKKdX01Qc8ibs25PUELi8GrzMDNOrHb_bQvYdq3ZqA4XLy4w@mail.gmail.com>
	<CAOTmbMxL3pLOuw655fyT5VzAON_N0PrDN+-i-++=oZoYU6+Fmg@mail.gmail.com>
	<CAOTmbMyvyDA-CRpaMC=WRGVkiyTG=Rs+n4hyeQWdyTZpqysSzw@mail.gmail.com>
	<CAJoKKdXSwF02S9tBkorGwOjj57AtkRpU0c177qXASEpodKVk=w@mail.gmail.com>
	<CAOTmbMzu2qDYh2T24JSpCYn1xH2J4iGiK7UkUB4Kq=-KdcDWGQ@mail.gmail.com>
	<CAJoKKdU66jK3f+mOEV90Y3zrmzpTSdWnwGsWimZxir42khiJ-w@mail.gmail.com>
	<CAOTmbMxW=EDr3-Ayo2R_MsCCTUZ4PnDkkRUuR2FisH9tK5xkWA@mail.gmail.com>
	<CAJoKKdVSkQOV3Kou_ro5AQZTyaqH6r0WRE-hZdrA=ye8ARE_vQ@mail.gmail.com>
Message-ID: <CAOTmbMyg5_q-pFFtpzOooUcSZ8Gkb38y4sk1pYeJLYqguA=Jbg@mail.gmail.com>

Hi Tihomir,

The issue has been resolved by correcting the client secrect, appreciated
very much for your kindly help!

Regards,
Charles

2011/8/7 Tihomir Culjaga <tculjaga at gmail.com>

> are u sure you are using the correct secret on both client and server ?
>
>
> On Fri, Aug 5, 2011 at 10:12 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>
>> Hi Tihomir,
>>
>> Thanks for your advise, i've added below to rad_auth.conf.xml (vsas
>> section), as well as tried auth-type to 0(local) and 1(system), however, the
>> issue still exist.
>>
>>
>> <param name="NAS-Port-Type" id="61" value="0" pec="0" expr="0"
>> direction="in"/>
>> <param name="Login-User" id="6" value="1" pec="0" expr="0"
>> direction="in"/>
>> <param name="Auth-Type" id="1000" value="0" pec="0" expr="0"
>> direction="in"/>
>>
>> FreeRadius output:
>>
>> Found Auth-Type = PAP
>> # Executing group from file /usr/local/etc/raddb/sites-enabled/default
>> +- entering group PAP {...}
>> [pap] login attempt with password "Q?????? ??????p???F?+??a"
>> [pap] Using clear text password "1111"
>> [pap] Passwords don't match
>> ++[pap] returns reject
>> Failed to authenticate the user.
>>   WARNING: Unprintable characters in the password.        Double-check the shared secret on the server and the NAS!
>>
>> Using Post-Auth-Type Reject
>> # Executing group from file /usr/local/etc/raddb/sites-enabled/default
>> +- entering group REJECT {...}
>> [attr_filter.access_reject]     expand: %{User-Name} -> 1001
>>  attr_filter: Matched entry DEFAULT at line 11
>> ++[attr_filter.access_reject] returns updated
>> Delaying reject of request 38 for 1 seconds
>>
>> Regards,
>> Charles
>>
>>
>> 2011/8/5 Tihomir Culjaga <tculjaga at gmail.com>
>>
>>> add to rad_auth.conf.xml
>>>
>>> <param name="NAS-Port-Type" id="61" value="0" pec="0" expr="0"
>>> direction="in"/>
>>> <param name="Login-User" id="6" value="1" pec="0" expr="0"
>>> direction="in"/>
>>>
>>>
>>>
>>> as for Auth Type im not sure if you need it ... this is up to your
>>> server.
>>> According to dictionary file you need to set it as follows:
>>>
>>> <param name="Auth-Type" id="1000" value="?" pec="0" expr="0"
>>> direction="in"/>
>>>
>>> the value (set as ?) is one of the folowing. Again, not sure what is
>>> required by your server.
>>>
>>> VALUE           Auth-Type               Local                   0
>>> VALUE           Auth-Type               System                  1
>>> VALUE           Auth-Type               SecurID                 2
>>> VALUE           Auth-Type               Crypt-Local             3
>>> VALUE           Auth-Type               Reject                  4
>>>
>>> #
>>> #       Cistron extensions
>>> #
>>> VALUE           Auth-Type               Pam                     253
>>> VALUE           Auth-Type               Accept                  254
>>>
>>>
>>>
>>> regards,
>>> Tihomir.
>>>
>>>
>>>
>>> On Wed, Aug 3, 2011 at 6:32 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>>>
>>>> Hi Tihomir,
>>>>
>>>> Sorry, i missed your mail in gmail before, just now saw it, and after
>>>> using your dictionary.all, the dictionary issue was resolved, very
>>>> appreciated for your kindly help! however, it did not fully functional yet,
>>>>
>>>> Attached are configuration files that i used, when i dial 601 to trigger
>>>> to auth, the freeradius server shows log below, the supecious log is the
>>>> value User-Password, it should be '1111' that i've set in the mysql db of
>>>> freeradisu server for the user 1001 .
>>>>
>>>> i searched in google, for "known good" password issue, i suggest change
>>>> user-password to cleartext-password, however, i did not find where it is.
>>>> and also the Auth-Type, where to configure it...
>>>>
>>>> Freeradius server log:
>>>>
>>>> rad_recv: Access-Request packet from host 127.0.0.1 port 52684, id=49,
>>>> length=111
>>>>         User-Name = "1001"
>>>>         User-Password = "?\210\365@\263\t\306\343\243iT?\311C\t\002"
>>>>         Called-Station-Id = "888"
>>>>         h323-conf-id = "749d2b5a-16ad-48e4-af58-24011949d1b5"
>>>>         Calling-Station-Id = "1001"
>>>>         NAS-Port = 0
>>>>         NAS-IP-Address = 127.0.0.1
>>>> # Executing section authorize from file
>>>> /usr/local/etc/raddb/sites-enabled/default
>>>> +- entering group authorize {...}
>>>> ++[preprocess] returns ok
>>>> [auth_log]      expand:
>>>> /usr/local/var/log/radius/radacct/%{Client-IP-Address}/auth-detail-%Y%m%d ->
>>>> /usr/local/var/log/radius/radacct/127.0.0.1/auth-detail-20110803
>>>> [auth_log]
>>>> /usr/local/var/log/radius/radacct/%{Client-IP-Address}/auth-detail-%Y%m%d
>>>> expands to /usr/local/var/log/radius/radacct/
>>>> 127.0.0.1/auth-detail-20110803
>>>> [auth_log]      expand: %t -> Wed Aug  3 12:06:33 2011
>>>> ++[auth_log] returns ok
>>>> ++[chap] returns noop
>>>> ++[mschap] returns noop
>>>> ++[digest] returns noop
>>>> [suffix] No '@' in User-Name = "1001", looking up realm NULL
>>>> [suffix] No such realm "NULL"
>>>> ++[suffix] returns noop
>>>> [eap] No EAP-Message, not doing EAP
>>>> ++[eap] returns noop
>>>> ++[unix] returns notfound
>>>> ++[files] returns noop
>>>> [sql]   expand: %{User-Name} -> 1001
>>>> [sql] sql_set_user escaped user --> '1001'
>>>> rlm_sql (sql): Reserving sql socket id: 4
>>>> [sql]   expand: SELECT id, username, attribute, value, op           FROM
>>>> radcheck           WHERE username = '%{SQL-User-Name}'           ORDER BY id
>>>> -> SELECT id, username, attribute, value, op           FROM
>>>> radcheck           WHERE username = '1001'           ORDER BY id
>>>> [sql]   expand: SELECT groupname           FROM radusergroup
>>>> WHERE username = '%{SQL-User-Name}'           ORDER BY priority -> SELECT
>>>> groupname           FROM radusergroup           WHERE username =
>>>> '1001'           ORDER BY priority
>>>> rlm_sql (sql): Released sql socket id: 4
>>>> [sql] User 1001 not found
>>>> ++[sql] returns notfound
>>>> ++[expiration] returns noop
>>>> ++[logintime] returns noop
>>>> [pap] WARNING! No "known good" password found for the user.
>>>> Authentication may fail because of this.
>>>> ++[pap] returns noop
>>>> ERROR: No authenticate method (Auth-Type) found for the request:
>>>> Rejecting the user
>>>> Failed to authenticate the user.
>>>>   WARNING: Unprintable characters in the password.        Double-check
>>>> the shared secret on the server and the NAS!
>>>> Using Post-Auth-Type Reject
>>>> # Executing group from file /usr/local/etc/raddb/sites-enabled/default
>>>> +- entering group REJECT {...}
>>>> [attr_filter.access_reject]     expand: %{User-Name} -> 1001
>>>>  attr_filter: Matched entry DEFAULT at line 11
>>>> ++[attr_filter.access_reject] returns updated
>>>> Delaying reject of request 8 for 1 seconds
>>>> Going to the next request
>>>> Waking up in 0.9 seconds.
>>>> Sending delayed reject for request 8
>>>> Sending Access-Reject of id 49 to 127.0.0.1 port 52684
>>>> Waking up in 4.9 seconds.
>>>> Cleaning up request 8 ID 49 with timestamp +7674
>>>> Ready to process requests.
>>>> WARNING! No "known good" password found for the user
>>>>
>>>> Regards,
>>>> Charles
>>>>
>>>>
>>>> 2011/8/3 Tihomir Culjaga <tculjaga at gmail.com>
>>>>
>>>>> did u use the dictionary i have attached ?
>>>>>
>>>>>
>>>>> On Tue, Aug 2, 2011 at 10:08 AM, fieldpeak <fieldpeak at gmail.com>wrote:
>>>>>
>>>>>> i tried change to 'h323-conf-id' to 'h323-call-origin' in
>>>>>> 02_unitest_rad-ANI-auth.xml, rad_auth.conf.xml, however, it still prompt
>>>>>> '[ERR] mod_rad_auth.c:428 Unknown attribute: key:h323-conf-id, not
>>>>>> found in dictionary', so where the mod_rad_auth read out the 'h323-conf-id'?
>>>>>> very very strange, which dictionary it was using...
>>>>>>
>>>>>> Regards,
>>>>>> Charles
>>>>>>
>>>>>>
>>>>>> 2011/8/2 fieldpeak <fieldpeak at gmail.com>
>>>>>>
>>>>>>> Hi Tihomir,
>>>>>>>
>>>>>>> Finally the answer coming, i see the hope, thanks for your reply, :)
>>>>>>>
>>>>>>> As your advise, i only use one attribute(h323-conf-id) in my
>>>>>>> dialplan, and only one attribute(h323-conf-id) in rad_auth.conf.xml, and
>>>>>>> using the attached dictionary (from ciso) which contains this attribute,
>>>>>>> however, it still prompt 'unknown attribute', so i suspected if it was
>>>>>>> reading /usr/local/etc/radiusclient/dictionary, so i copy the same
>>>>>>> dictionary to /usr/local/freeswitch/radius/, it did not any help at all...
>>>>>>> very strange...
>>>>>>>
>>>>>>> Log:
>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set
>>>>>>> default_realm := .
>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set
>>>>>>> radius_timeout := 3.
>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set
>>>>>>> radius_retries := 2.
>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set
>>>>>>> radius_deadtime := 0.
>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set bindaddr :=
>>>>>>> *.
>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:371 ... radius:
>>>>>>> User-Name: 38516060333
>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:380 ... radius:
>>>>>>> User-Password: 003282
>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:396 ... radius:
>>>>>>> Called-station-Id: 16094191500
>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:413 Handle
>>>>>>> attribute: h323-conf-id
>>>>>>> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:428 Unknown
>>>>>>> attribute: key:h323-conf-id, not found in dictionary
>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:538 abort sending
>>>>>>> radius packet.
>>>>>>> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:546 An error occured
>>>>>>> during RADIUS Authentication(RC=-1)
>>>>>>> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:702 An error occured
>>>>>>> during radius authorization.
>>>>>>>
>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  AUTH_RESULT=)
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>   <extension name="unitest_rad-ANI-auth">
>>>>>>>     <condition field="destination_number" expression="^601$">
>>>>>>>       <!-- <action application="log" data="INFO  Before Auth "/> -->
>>>>>>>
>>>>>>>       <action inline="true" application="set" data="CALLID=
>>>>>>> h323-conf-id=${uuid}"/>
>>>>>>>
>>>>>>>       <action inline="true" application="set" data="USERNAME=1001"/>
>>>>>>>       <action inline="true" application="set" data="PASSWD=1111"/>
>>>>>>>
>>>>>>>
>>>>>>>       <action application="sleep" data="2000"/>
>>>>>>>       <action application="auth_function" data="in ${DIALED_NUMBER},
>>>>>>> in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>
>>>>>>>
>>>>>>>     </condition>
>>>>>>>   </extension>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> <configuration name="rad_auth.conf" description="radius
>>>>>>> authentification module">
>>>>>>>   <settings>
>>>>>>>
>>>>>>>   </settings>
>>>>>>>
>>>>>>>   <client>
>>>>>>>     <param name="authserver" value="127.0.0.1:1812:gateway"/>
>>>>>>>     <param name="dictionary"
>>>>>>> value="/usr/local/etc/radiusclient/dictionary"/>
>>>>>>>     <param name="seqfile" value="/var/run/radius.seq"/>
>>>>>>>     <param name="mapfile"
>>>>>>> value="/usr/local/etc/radiusclient/port-id-map"/>
>>>>>>>     <param name="default_realm" value=""/>
>>>>>>>     <param name="radius_timeout" value="3"/>
>>>>>>>     <param name="radius_retries" value="2"/>
>>>>>>>     <param name="radius_deadtime" value="0"/>
>>>>>>>     <param name="bindaddr" value="*"/>
>>>>>>>   </client>
>>>>>>>
>>>>>>>   <vsas>
>>>>>>>
>>>>>>>
>>>>>>>     <param name="h323-conf-id" id="24" value="CALLID" pec="9"
>>>>>>> expr="1" direction="in"/>
>>>>>>>
>>>>>>>   </vsas>
>>>>>>>  </configuration>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> 2011/8/2 Tihomir Culjaga <tculjaga at gmail.com>
>>>>>>>
>>>>>>>> hi,
>>>>>>>>
>>>>>>>> dictionary.all is just the name of a file containing all attributes
>>>>>>>> i needed at that time.
>>>>>>>>
>>>>>>>> you can include other dictionaries by putting #INCLUDE <pathname> at
>>>>>>>> the end of the dictionary file you reference in rad_auth.conf.xml.
>>>>>>>> if the INCLUDE doesn't work, just append dictionary.cisco to your
>>>>>>>> dictionary file... and make your own file.
>>>>>>>>
>>>>>>>>
>>>>>>>> check inline comments down below...
>>>>>>>>
>>>>>>>>
>>>>>>>> T.
>>>>>>>>
>>>>>>>>
>>>>>>>> On Sun, Jul 31, 2011 at 10:46 AM, fieldpeak <fieldpeak at gmail.com>wrote:
>>>>>>>>
>>>>>>>>> Hello Gurus,
>>>>>>>>>
>>>>>>>>> i met a issue when using
>>>>>>>>> mod_rad_auth(http://wiki.freeswitch.org/wiki/Mod_rad_auth) to
>>>>>>>>> works
>>>>>>>>> with freeradius server+mysql for AAA, the details is below, Could
>>>>>>>>> anyone give any hints, Thanks in advance.
>>>>>>>>>
>>>>>>>>> i setup a dial plan "unitest_rad-ANI-auth" as wiki above, however,
>>>>>>>>> when i dialed 601 to trigger the dial plan, the console show
>>>>>>>>> errors,
>>>>>>>>> it looks "h323-conf-id" is not in the directory, then i tried to
>>>>>>>>> add
>>>>>>>>> this attribute to the dictionary, however, it does not help, in the
>>>>>>>>> wiki, it mentioned the rad_auth.conf.xml contains <param
>>>>>>>>> name="dictionary"
>>>>>>>>> value="/usr/local/etc/radiusclient/dictionary.all"/>, however i did
>>>>>>>>> not find the file "dictionary.all" at that directory, so i use
>>>>>>>>> dictionary. BTW, the freeradius server + mysql works well.
>>>>>>>>>
>>>>>>>>
>>>>>>>> i just appended the information needed into dictionary.all file...
>>>>>>>> (vendor and attribute definition).
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>>
>>>>>>>>> console errors:
>>>>>>>>>
>>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 auth_function(in , in
>>>>>>>>> 38516060333, in 003282, out AUTH_RESULT)
>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:301 allocate
>>>>>>>>> initial
>>>>>>>>> structure.
>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:313 initialzed
>>>>>>>>> configuration.
>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>> authserver
>>>>>>>>> := 127.0.0.1:1812:gateway.
>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>> dictionary
>>>>>>>>> := /usr/local/etc/radiusclient/dictionary.
>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set seqfile
>>>>>>>>> :=
>>>>>>>>> /var/run/radius.seq.
>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set mapfile
>>>>>>>>> :=
>>>>>>>>> /usr/local/etc/radiusclient/port-id-map.
>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>> default_realm := .
>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>> radius_timeout := 3.
>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>> radius_retries := 2.
>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>> radius_deadtime := 0.
>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set bindaddr
>>>>>>>>> := *.
>>>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:371 ... radius:
>>>>>>>>> User-Name: 38516060333
>>>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:380 ... radius:
>>>>>>>>> User-Password: 003282
>>>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:391 ... radius:
>>>>>>>>> Called-station-Id is empty, ignoring...
>>>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:413 Handle
>>>>>>>>> attribute: h323-conf-id
>>>>>>>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:428 Unknown
>>>>>>>>> attribute:
>>>>>>>>> key:h323-conf-id, not found in dictionary
>>>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:538 abort sending
>>>>>>>>> radius packet.
>>>>>>>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:546 An error
>>>>>>>>> occured
>>>>>>>>> during RADIUS Authentication(RC=-1)
>>>>>>>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:702 An error
>>>>>>>>> occured
>>>>>>>>> during radius authorization.
>>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO
>>>>>>>>>  AUTH_RESULT=)
>>>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  AUTH_RESULT=
>>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO
>>>>>>>>>  billing_model=)
>>>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202
>>>>>>>>>  billing_model=
>>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO
>>>>>>>>>  credit_amount=)
>>>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202
>>>>>>>>>  credit_amount=
>>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  currency=)
>>>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  currency=
>>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO
>>>>>>>>>  preffered_lang=)
>>>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202
>>>>>>>>>  preffered_lang=
>>>>>>>>>
>>>>>>>>> added below in the
>>>>>>>>> dictionary(/usr/local/etc/radiusclient/dictionary):
>>>>>>>>>
>>>>>>>>> ATTRIBUTE       h323-conf-id            1008    string
>>>>>>>>>
>>>>>>>>
>>>>>>>> you need the vendor definition as well
>>>>>>>>
>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> dial plan:
>>>>>>>>> <extension name="unitest_rad-ANI-auth">
>>>>>>>>>    <condition field="destination_number" expression="^601$">
>>>>>>>>>      <action application="log" data="INFO  Before Auth "/>
>>>>>>>>>
>>>>>>>>>      <action inline="true" application="set"
>>>>>>>>> data="CALLID=h323-conf-id=${uuid}"/>
>>>>>>>>>      <action inline="true" application="set"
>>>>>>>>> data="SERVICENUM=h323-prompt-id=${destination_number}"/>
>>>>>>>>>      <action inline="true" application="set"
>>>>>>>>> data="TRANSACTIONID=h323-ivr-out=transactionID:1234"/>
>>>>>>>>>  <!--      <action inline="true" application="set"
>>>>>>>>> data="CALLINGNUMBER=${caller_id_number}"/> -->
>>>>>>>>>      <action inline="true" application="set"
>>>>>>>>> data="CALLINGNUMBER=38516060333"/>
>>>>>>>>>      <action inline="true" application="set"
>>>>>>>>> data="USERNAME=38516060333"/>
>>>>>>>>>  <!--      <action inline="true" application="set"
>>>>>>>>> data="USERNAME=209354"/> -->
>>>>>>>>>      <action inline="true" application="set" data="PASSWD=003282"/>
>>>>>>>>>  <!--      <action inline="true" application="set"
>>>>>>>>> data="DIALED_NUMBER=16094191500"/>  -->
>>>>>>>>>
>>>>>>>>>      <action application="sleep" data="2000"/>
>>>>>>>>>      <action application="auth_function" data="in ${DIALED_NUMBER},
>>>>>>>>> in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>>  AUTH_RESULT=${AUTH_RESULT}"/>
>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>>  billing_model=${billing_model}"/>
>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>>  credit_amount=${credit_amount}"/>
>>>>>>>>>      <action application="log" data="INFO  currency=${currency}"/>
>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>>  preffered_lang=${preffered_lang}"/>
>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>>  credit_time=${credit_time}"/>
>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>> h323_ivr_duration=${h323_ivr_duration}"/>
>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>>  return_code=${return_code}"/>
>>>>>>>>>      <!-- <action application="execute_extension" data="AUTH XML
>>>>>>>>> default"/> -->
>>>>>>>>>    </condition>
>>>>>>>>>  </extension>
>>>>>>>>>
>>>>>>>>>  radius_cdr.conf.xml:
>>>>>>>>>  <configuration name="radius_cdr.conf" description="RADIUS CDR
>>>>>>>>> Configuration">
>>>>>>>>>
>>>>>>>>>        <settings>
>>>>>>>>>
>>>>>>>>>                <!-- location of the radius dictionary files -->
>>>>>>>>>
>>>>>>>>>                <param name="dictionary"
>>>>>>>>> value="/usr/local/freeswitch/conf/radius/dictionary"/>
>>>>>>>>>
>>>>>>>>>
>>>>>>>> your dictionary file need to contain all the attributes you are
>>>>>>>> trying to use or to include other dictionaries (In this case
>>>>>>>> dictionary.cisco) from the dictionary file you are referencing here.
>>>>>>>>
>>>>>>>>
>>>>>>>>>                <!-- number of retries for each server -->
>>>>>>>>>
>>>>>>>>>                <param name="radius_retries" value="3"/>
>>>>>>>>>
>>>>>>>>>                <!-- number of seconds to wait between retries -->
>>>>>>>>>
>>>>>>>>>                <param name="radius_timeout" value="5"/>
>>>>>>>>>
>>>>>>>>>                <!-- accounting servers, up to 8 allowed -->
>>>>>>>>>
>>>>>>>>>                <!-- value is "host:port:secret", port is optional
>>>>>>>>> -->
>>>>>>>>>
>>>>>>>>>                <!-- use IP ADDRESSES, not hostnames -->
>>>>>>>>>
>>>>>>>>>                <param name="acct_server" value="127.0.0.1:1813
>>>>>>>>> :testing123"/>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>        </settings>
>>>>>>>>>
>>>>>>>>> </configuration>
>>>>>>>>>
>>>>>>>>>  the FS version:
>>>>>>>>>  FreeSWITCH Version 1.0.head (git-492bc6b 2011-07-23 12-53-04
>>>>>>>>> -0400)
>>>>>>>>>
>>>>>>>>>  Regards,
>>>>>>>>>  Charles
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110808/b829c9b6/attachment-0001.html 

From steveayre at gmail.com  Mon Aug  8 10:39:09 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 8 Aug 2011 07:39:09 +0100
Subject: [Freeswitch-users] Best way to route Large block of DIDs
In-Reply-To: <CAB2KLyd6nn0bg-Vp9R8fygjYP_1LGDC26Zhw5FDvYB4G3vOtmg@mail.gmail.com>
References: <CAB2KLyd6nn0bg-Vp9R8fygjYP_1LGDC26Zhw5FDvYB4G3vOtmg@mail.gmail.com>
Message-ID: <FD9E6D50-D623-4097-BBBC-09FA8ECB1216@gmail.com>

It will because it'll check the regex for each extension until it finds a match. A fairly small hit usually but if you have thousands of extensions not so good.

You're probably best off looking the numbers up in a database, which can be done easily from a lua script and will let you handle all the DIDs in a single extension.

Steve on iPhone

On 8 Aug 2011, at 02:38, Luis F Urrea <lfurrea at gmail.com> wrote:

> Hi all,
> 
> I have a relatively large block  of DIDs coming from a T1 gateway to a FS box. These DIDs are mapped to user extensions in most cases and in others they go to an IVR.
> 
> My question is what would be the best way to write an extension in public.xml dialplan if the block of DIDs that go directly to users is not exactly divided so that I can write a regexp to cover the block.
> 
> Does writing a single rule for each user can bring performance implications?
> 
> What if I cannot arrange the block of DIDs that belong to users in a consecutive group?
> 
> What are the best practices for such case?
> 
> TIA!!!
> 
> Luis
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From david.villasmil.work at gmail.com  Mon Aug  8 12:30:01 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Mon, 8 Aug 2011 10:30:01 +0200
Subject: [Freeswitch-users] event
In-Reply-To: <CAE2ghw4GeQp-seD-c9nxa=W-wnkFbEQW6Ji_2sNqGBuaj8n+ug@mail.gmail.com>
References: <CAE2ghw4GeQp-seD-c9nxa=W-wnkFbEQW6Ji_2sNqGBuaj8n+ug@mail.gmail.com>
Message-ID: <CAFGRPVoSWV8ARFYcgKaWDh_iawpwn1UMOq6gz7dUefxS1_UVCg@mail.gmail.com>

Hello,

Maybe by binding it to one IP?

David

On Mon, Aug 8, 2011 at 8:13 AM, Sam <u2nsam at gmail.com> wrote:

> Hello,
>
> What makes the below event to occour and how to stop it reccouring.
>
> 2011-08-08 11:38:11.415945 [INFO] mod_sofia.c:4919 EVENT_TRAP: IP change
> detected
> 2011-08-08 11:38:11.415945 [INFO] mod_sofia.c:4920 IP change detected
> [192.168.53.188]->[192.168.53.189] []->[]
> 2011-08-08 11:38:11.615887 [NOTICE] sofia_glue.c:5192 Reload XML [Success]
> 2011-08-08 11:38:11.615887 [INFO] mod_enum.c:775 ENUM Reloaded
> 2011-08-08 11:38:11.615887 [INFO] switch_time.c:1028 Timezone reloaded 530
> definitions
>
> Regards
> Sam
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110808/54757343/attachment.html 

From avi at avimarcus.net  Mon Aug  8 12:33:23 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 8 Aug 2011 11:33:23 +0300
Subject: [Freeswitch-users] Best way to route Large block of DIDs
In-Reply-To: <FD9E6D50-D623-4097-BBBC-09FA8ECB1216@gmail.com>
References: <CAB2KLyd6nn0bg-Vp9R8fygjYP_1LGDC26Zhw5FDvYB4G3vOtmg@mail.gmail.com>
	<FD9E6D50-D623-4097-BBBC-09FA8ECB1216@gmail.com>
Message-ID: <CALNZuwna21Ji3HnJLVm+e507C14Yh1RZscNF02NwvMNyWHPiLg@mail.gmail.com>

Yes, a lua script to run a single query - or from the git contrib
mod_odbc_query <http://wiki.freeswitch.org/wiki/Mod_odbc_query> can easily
look up from a database and route accordingly.

-Avi


On Mon, Aug 8, 2011 at 9:39 AM, Steven Ayre <steveayre at gmail.com> wrote:

> It will because it'll check the regex for each extension until it finds a
> match. A fairly small hit usually but if you have thousands of extensions
> not so good.
>
> You're probably best off looking the numbers up in a database, which can be
> done easily from a lua script and will let you handle all the DIDs in a
> single extension.
>
> Steve on iPhone
>
> On 8 Aug 2011, at 02:38, Luis F Urrea <lfurrea at gmail.com> wrote:
>
> > Hi all,
> >
> > I have a relatively large block  of DIDs coming from a T1 gateway to a FS
> box. These DIDs are mapped to user extensions in most cases and in others
> they go to an IVR.
> >
> > My question is what would be the best way to write an extension in
> public.xml dialplan if the block of DIDs that go directly to users is not
> exactly divided so that I can write a regexp to cover the block.
> >
> > Does writing a single rule for each user can bring performance
> implications?
> >
> > What if I cannot arrange the block of DIDs that belong to users in a
> consecutive group?
> >
> > What are the best practices for such case?
> >
> > TIA!!!
> >
> > Luis
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110808/38fe68ad/attachment.html 

From dujinfang at gmail.com  Mon Aug  8 13:58:51 2011
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 8 Aug 2011 17:58:51 +0800
Subject: [Freeswitch-users] openssl deprecated on OS X Lion and MacPorts
 with latest Git head?
In-Reply-To: <CAMWiMy-Rxt3WQWnoUNm1Y3e7GARtieea6tbfQ4YY+uL=F5G6EQ@mail.gmail.com>
References: <CAA8_NKAgfcJWdLLBBih_wL6vmZbNAL--q+PT9Q2FP-nf7arFaQ@mail.gmail.com>
	<CAMWiMy-Rxt3WQWnoUNm1Y3e7GARtieea6tbfQ4YY+uL=F5G6EQ@mail.gmail.com>
Message-ID: <A9400E88F0BB42BD8FE7861B1E2B0398@gmail.com>

Apple deprecated a lot of functions on Lion and there is jira you can follow: http://jira.freeswitch.org/browse/FS-3447

Also you can find a temporary work around at: http://jira.freeswitch.org/browse/FS-3450 
-- 
Seven Du
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj: http://www.freeswitch.org.cn
Sent with Sparrow (http://www.sparrowmailapp.com)

On Monday, August 8, 2011 at 12:12 PM, curriegrad2004 wrote:

> it's more like a compiler issue from what I can tell here
> 
> On Sun, Aug 7, 2011 at 2:33 PM, Gavin Henry <gavin.henry at gmail.com (mailto:gavin.henry at gmail.com)> wrote:
> > Hi all,
> > 
> > Anyone seeing this? My port tree is up to date and on latest MacPorts:
> > 
> > In file included from src/switch_core.c:39:
> > ./src/include/switch_ssl.h: In function ?switch_ssl_init_ssl_locks?:
> > ./src/include/switch_ssl.h:63: warning: ?CRYPTO_num_locks? is
> > deprecated (declared at /usr/include/openssl/crypto.h:415)
> > ./src/include/switch_ssl.h:65: warning: ?CRYPTO_malloc? is deprecated
> > (declared at /usr/include/openssl/crypto.h:478)
> > ./src/include/switch_ssl.h:65: warning: ?CRYPTO_num_locks? is
> > deprecated (declared at /usr/include/openssl/crypto.h:415)
> > ./src/include/switch_ssl.h:75: warning: ?CRYPTO_set_id_callback? is
> > deprecated (declared at /usr/include/openssl/crypto.h:425)
> > ./src/include/switch_ssl.h:76: warning: ?CRYPTO_set_locking_callback?
> > is deprecated (declared at /usr/include/openssl/crypto.h:418)
> > ./src/include/switch_ssl.h: In function ?switch_ssl_destroy_ssl_locks?:
> > ./src/include/switch_ssl.h:91: warning: ?CRYPTO_set_locking_callback?
> > is deprecated (declared at /usr/include/openssl/crypto.h:418)
> > ./src/include/switch_ssl.h:92: warning: ?CRYPTO_num_locks? is
> > deprecated (declared at /usr/include/openssl/crypto.h:415)
> > ./src/include/switch_ssl.h:98: warning: ?CRYPTO_free? is deprecated
> > (declared at /usr/include/openssl/crypto.h:480)
> > make[2]: *** [libfreeswitch_la-switch_core.lo] Error 1
> > make[1]: *** [all] Error 2
> > make: *** [current] Error 2
> > 
> > --
> > http://www.suretecsystems.com/services/openldap/
> > http://www.surevoip.co.uk
> > 
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org)
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org)
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110808/4c162824/attachment.html 

From steveayre at gmail.com  Mon Aug  8 14:17:46 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 8 Aug 2011 11:17:46 +0100
Subject: [Freeswitch-users] event
In-Reply-To: <CAE2ghw4GeQp-seD-c9nxa=W-wnkFbEQW6Ji_2sNqGBuaj8n+ug@mail.gmail.com>
References: <CAE2ghw4GeQp-seD-c9nxa=W-wnkFbEQW6Ji_2sNqGBuaj8n+ug@mail.gmail.com>
Message-ID: <CAFiqYumEKWBJd5_WR1A9cx72Rw5Digr6EVx7Q96bsBxRT6mJoA@mail.gmail.com>

Well it looks like your machine changed its IP from .188 to .189. Did that
actually happen, perhaps if you're using DHCP? If it really did happen, then
FS needs to know so it can rebind on the new IP.

If it didn't really happen (e.g. if your server is listening on both IPs)
look at the <param name="auto-restart" value="false"/> settting on the SIP
profile. See:
http://wiki.freeswitch.org/wiki/Sofia#Forcing_SIP_profile_to_use_a_static_IP_address

That parameter will make mod_sofia ignore the notification it receives from
the OS of a network address change.

-Steve



On 8 August 2011 07:13, Sam <u2nsam at gmail.com> wrote:

> Hello,
>
> What makes the below event to occour and how to stop it reccouring.
>
> 2011-08-08 11:38:11.415945 [INFO] mod_sofia.c:4919 EVENT_TRAP: IP change
> detected
> 2011-08-08 11:38:11.415945 [INFO] mod_sofia.c:4920 IP change detected
> [192.168.53.188]->[192.168.53.189] []->[]
> 2011-08-08 11:38:11.615887 [NOTICE] sofia_glue.c:5192 Reload XML [Success]
> 2011-08-08 11:38:11.615887 [INFO] mod_enum.c:775 ENUM Reloaded
> 2011-08-08 11:38:11.615887 [INFO] switch_time.c:1028 Timezone reloaded 530
> definitions
>
> Regards
> Sam
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110808/1b0d24a5/attachment-0001.html 

From u2nsam at gmail.com  Mon Aug  8 17:14:42 2011
From: u2nsam at gmail.com (Sam)
Date: Mon, 8 Aug 2011 18:44:42 +0530
Subject: [Freeswitch-users] event
In-Reply-To: <CAFiqYumEKWBJd5_WR1A9cx72Rw5Digr6EVx7Q96bsBxRT6mJoA@mail.gmail.com>
References: <CAE2ghw4GeQp-seD-c9nxa=W-wnkFbEQW6Ji_2sNqGBuaj8n+ug@mail.gmail.com>
	<CAFiqYumEKWBJd5_WR1A9cx72Rw5Digr6EVx7Q96bsBxRT6mJoA@mail.gmail.com>
Message-ID: <CAE2ghw7B7HRLE4yQcPGnKgsSwDjWvyeJOvCcOpP_2CmLAEExdw@mail.gmail.com>

Hi,

The server is having 3 static ips on the ethernet interfaces and the server
is listening on single ip .
I will try <param name="auto-restart" value="false"/> and check if it works.

Regards
Sam



On Mon, Aug 8, 2011 at 3:47 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Well it looks like your machine changed its IP from .188 to .189. Did that
> actually happen, perhaps if you're using DHCP? If it really did happen, then
> FS needs to know so it can rebind on the new IP.
>
> If it didn't really happen (e.g. if your server is listening on both IPs)
> look at the <param name="auto-restart" value="false"/> settting on the SIP
> profile. See:
>
> http://wiki.freeswitch.org/wiki/Sofia#Forcing_SIP_profile_to_use_a_static_IP_address
>
> That parameter will make mod_sofia ignore the notification it receives from
> the OS of a network address change.
>
> -Steve
>
>
>
>   On 8 August 2011 07:13, Sam <u2nsam at gmail.com> wrote:
>
>>   Hello,
>>
>> What makes the below event to occour and how to stop it reccouring.
>>
>> 2011-08-08 11:38:11.415945 [INFO] mod_sofia.c:4919 EVENT_TRAP: IP change
>> detected
>> 2011-08-08 11:38:11.415945 [INFO] mod_sofia.c:4920 IP change detected
>> [192.168.53.188]->[192.168.53.189] []->[]
>> 2011-08-08 11:38:11.615887 [NOTICE] sofia_glue.c:5192 Reload XML [Success]
>> 2011-08-08 11:38:11.615887 [INFO] mod_enum.c:775 ENUM Reloaded
>> 2011-08-08 11:38:11.615887 [INFO] switch_time.c:1028 Timezone reloaded 530
>> definitions
>>
>> Regards
>> Sam
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110808/e7091282/attachment.html 

From u2nsam at gmail.com  Mon Aug  8 17:15:17 2011
From: u2nsam at gmail.com (Sam)
Date: Mon, 8 Aug 2011 18:45:17 +0530
Subject: [Freeswitch-users] event
In-Reply-To: <CAFiqYumEKWBJd5_WR1A9cx72Rw5Digr6EVx7Q96bsBxRT6mJoA@mail.gmail.com>
References: <CAE2ghw4GeQp-seD-c9nxa=W-wnkFbEQW6Ji_2sNqGBuaj8n+ug@mail.gmail.com>
	<CAFiqYumEKWBJd5_WR1A9cx72Rw5Digr6EVx7Q96bsBxRT6mJoA@mail.gmail.com>
Message-ID: <CAE2ghw58FQVnvqQckzJzym6OuKFKwHefhVeAiqLB0BdoALaPeg@mail.gmail.com>

Hi,

The server is having 3 static ips on the ethernet interfaces and the FS is
listening on single ip .
I will try <param name="auto-restart" value="false"/> and check if it works.

Regards
Sam

On Mon, Aug 8, 2011 at 3:47 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Well it looks like your machine changed its IP from .188 to .189. Did that
> actually happen, perhaps if you're using DHCP? If it really did happen, then
> FS needs to know so it can rebind on the new IP.
>
> If it didn't really happen (e.g. if your server is listening on both IPs)
> look at the <param name="auto-restart" value="false"/> settting on the SIP
> profile. See:
>
> http://wiki.freeswitch.org/wiki/Sofia#Forcing_SIP_profile_to_use_a_static_IP_address
>
> That parameter will make mod_sofia ignore the notification it receives from
> the OS of a network address change.
>
> -Steve
>
>
>
>   On 8 August 2011 07:13, Sam <u2nsam at gmail.com> wrote:
>
>>   Hello,
>>
>> What makes the below event to occour and how to stop it reccouring.
>>
>> 2011-08-08 11:38:11.415945 [INFO] mod_sofia.c:4919 EVENT_TRAP: IP change
>> detected
>> 2011-08-08 11:38:11.415945 [INFO] mod_sofia.c:4920 IP change detected
>> [192.168.53.188]->[192.168.53.189] []->[]
>> 2011-08-08 11:38:11.615887 [NOTICE] sofia_glue.c:5192 Reload XML [Success]
>> 2011-08-08 11:38:11.615887 [INFO] mod_enum.c:775 ENUM Reloaded
>> 2011-08-08 11:38:11.615887 [INFO] switch_time.c:1028 Timezone reloaded 530
>> definitions
>>
>> Regards
>> Sam
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110808/8610ac88/attachment.html 

From steveayre at gmail.com  Mon Aug  8 18:10:20 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 8 Aug 2011 15:10:20 +0100
Subject: [Freeswitch-users] event
In-Reply-To: <CAE2ghw7B7HRLE4yQcPGnKgsSwDjWvyeJOvCcOpP_2CmLAEExdw@mail.gmail.com>
References: <CAE2ghw4GeQp-seD-c9nxa=W-wnkFbEQW6Ji_2sNqGBuaj8n+ug@mail.gmail.com>
	<CAFiqYumEKWBJd5_WR1A9cx72Rw5Digr6EVx7Q96bsBxRT6mJoA@mail.gmail.com>
	<CAE2ghw7B7HRLE4yQcPGnKgsSwDjWvyeJOvCcOpP_2CmLAEExdw@mail.gmail.com>
Message-ID: <CAFiqYuky+pD9z=xJ_eRZfdH72UZcc-hnJgjHrwUv+nUuUiRGQA@mail.gmail.com>

It should, as I use that myself. My servers have 3 static IPs (2 public, 1
private) with SIP profiles on the 2 public interfaces using
auto-restart=false.

-Steve



On 8 August 2011 14:14, Sam <u2nsam at gmail.com> wrote:

> Hi,
>
> The server is having 3 static ips on the ethernet interfaces and the server
> is listening on single ip .
> I will try <param name="auto-restart" value="false"/> and check if it
> works.
>
> Regards
> Sam
>
>
>
> On Mon, Aug 8, 2011 at 3:47 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> Well it looks like your machine changed its IP from .188 to .189. Did that
>> actually happen, perhaps if you're using DHCP? If it really did happen, then
>> FS needs to know so it can rebind on the new IP.
>>
>> If it didn't really happen (e.g. if your server is listening on both IPs)
>> look at the <param name="auto-restart" value="false"/> settting on the SIP
>> profile. See:
>>
>> http://wiki.freeswitch.org/wiki/Sofia#Forcing_SIP_profile_to_use_a_static_IP_address
>>
>> That parameter will make mod_sofia ignore the notification it receives
>> from the OS of a network address change.
>>
>> -Steve
>>
>>
>>
>>   On 8 August 2011 07:13, Sam <u2nsam at gmail.com> wrote:
>>
>>>   Hello,
>>>
>>> What makes the below event to occour and how to stop it reccouring.
>>>
>>> 2011-08-08 11:38:11.415945 [INFO] mod_sofia.c:4919 EVENT_TRAP: IP change
>>> detected
>>> 2011-08-08 11:38:11.415945 [INFO] mod_sofia.c:4920 IP change detected
>>> [192.168.53.188]->[192.168.53.189] []->[]
>>> 2011-08-08 11:38:11.615887 [NOTICE] sofia_glue.c:5192 Reload XML
>>> [Success]
>>> 2011-08-08 11:38:11.615887 [INFO] mod_enum.c:775 ENUM Reloaded
>>> 2011-08-08 11:38:11.615887 [INFO] switch_time.c:1028 Timezone reloaded
>>> 530 definitions
>>>
>>> Regards
>>> Sam
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110808/ed0f49af/attachment.html 

From covici at ccs.covici.com  Mon Aug  8 21:08:57 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Mon, 08 Aug 2011 13:08:57 -0400
Subject: [Freeswitch-users] how to have fs interrupt a file or tts if digit
	is pressed
Message-ID: <23058.1312823337@ccs.covici.com>

Hi.  I am using Perl, if that makes any difference and I would like to
arrange things in such a way that during a prompt, if the person presses
a key, then the prompt will stop speaking and I can see what the key is
and do something.  Here is an excerpt of the script I am using.

$session->setInputCallback('got_press',""); #listen for key presses in
the background


while($session->ready())
{
$session->streamFile("test_break.wav");

    if ($press_so_far != "")
    {
        $session->say($press_so_far,"EN", "NAME_SPELLED", "ITERATED");
        $press_so_far="";
            }
}
 
Instead of streamFile, I tried speaking text through tts_command line,
and a phrase macro which did the same.  I was sure that at least
streamFile would break, but it did not.

Thanks in advance for any ideas.

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From Hector.Geraldino at ip-soft.net  Mon Aug  8 21:30:15 2011
From: Hector.Geraldino at ip-soft.net (Hector Geraldino)
Date: Mon, 8 Aug 2011 13:30:15 -0400
Subject: [Freeswitch-users] how to have fs interrupt a file or tts if
 digit	is pressed
In-Reply-To: <23058.1312823337@ccs.covici.com>
References: <23058.1312823337@ccs.covici.com>
Message-ID: <6A6B4C284AD15042B429EB9D904544AD021FD8A4A6@NY1-EXMB-01.ip-soft.net>

You can send a "break" command in the DTMF callback event. This will stop the audio playback or TTS.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of covici at ccs.covici.com
Sent: Monday, August 08, 2011 1:09 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] how to have fs interrupt a file or tts if digit is pressed

Hi.  I am using Perl, if that makes any difference and I would like to
arrange things in such a way that during a prompt, if the person presses
a key, then the prompt will stop speaking and I can see what the key is
and do something.  Here is an excerpt of the script I am using.

$session->setInputCallback('got_press',""); #listen for key presses in
the background


while($session->ready())
{
$session->streamFile("test_break.wav");

    if ($press_so_far != "")
    {
        $session->say($press_so_far,"EN", "NAME_SPELLED", "ITERATED");
        $press_so_far="";
            }
}
 
Instead of streamFile, I tried speaking text through tts_command line,
and a phrase macro which did the same.  I was sure that at least
streamFile would break, but it did not.

Thanks in advance for any ideas.

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com

_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

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


From tculjaga at gmail.com  Mon Aug  8 21:56:45 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 8 Aug 2011 19:56:45 +0200
Subject: [Freeswitch-users] Mod_rad_auth issue for FS working with
 FreeRadius server
In-Reply-To: <CAOTmbMyg5_q-pFFtpzOooUcSZ8Gkb38y4sk1pYeJLYqguA=Jbg@mail.gmail.com>
References: <CAOTmbMzvd5OHL8z=_8vu6Orp2_prGxaw1gmXYCM2=2+pgzs_WA@mail.gmail.com>
	<CAJoKKdX01Qc8ibs25PUELi8GrzMDNOrHb_bQvYdq3ZqA4XLy4w@mail.gmail.com>
	<CAOTmbMxL3pLOuw655fyT5VzAON_N0PrDN+-i-++=oZoYU6+Fmg@mail.gmail.com>
	<CAOTmbMyvyDA-CRpaMC=WRGVkiyTG=Rs+n4hyeQWdyTZpqysSzw@mail.gmail.com>
	<CAJoKKdXSwF02S9tBkorGwOjj57AtkRpU0c177qXASEpodKVk=w@mail.gmail.com>
	<CAOTmbMzu2qDYh2T24JSpCYn1xH2J4iGiK7UkUB4Kq=-KdcDWGQ@mail.gmail.com>
	<CAJoKKdU66jK3f+mOEV90Y3zrmzpTSdWnwGsWimZxir42khiJ-w@mail.gmail.com>
	<CAOTmbMxW=EDr3-Ayo2R_MsCCTUZ4PnDkkRUuR2FisH9tK5xkWA@mail.gmail.com>
	<CAJoKKdVSkQOV3Kou_ro5AQZTyaqH6r0WRE-hZdrA=ye8ARE_vQ@mail.gmail.com>
	<CAOTmbMyg5_q-pFFtpzOooUcSZ8Gkb38y4sk1pYeJLYqguA=Jbg@mail.gmail.com>
Message-ID: <CAJoKKdWqjt46Gvbj=4Tz-apQb+EV2UHJTsRBTE_bJZ95ogo2DA@mail.gmail.com>

im glad it works :=)

T.

On Mon, Aug 8, 2011 at 8:18 AM, fieldpeak <fieldpeak at gmail.com> wrote:

> Hi Tihomir,
>
> The issue has been resolved by correcting the client secrect, appreciated
> very much for your kindly help!
>
> Regards,
> Charles
>
> 2011/8/7 Tihomir Culjaga <tculjaga at gmail.com>
>
>> are u sure you are using the correct secret on both client and server ?
>>
>>
>> On Fri, Aug 5, 2011 at 10:12 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>>
>>> Hi Tihomir,
>>>
>>> Thanks for your advise, i've added below to rad_auth.conf.xml (vsas
>>> section), as well as tried auth-type to 0(local) and 1(system), however, the
>>> issue still exist.
>>>
>>>
>>> <param name="NAS-Port-Type" id="61" value="0" pec="0" expr="0"
>>> direction="in"/>
>>> <param name="Login-User" id="6" value="1" pec="0" expr="0"
>>> direction="in"/>
>>> <param name="Auth-Type" id="1000" value="0" pec="0" expr="0"
>>> direction="in"/>
>>>
>>> FreeRadius output:
>>>
>>> Found Auth-Type = PAP
>>> # Executing group from file /usr/local/etc/raddb/sites-enabled/default
>>> +- entering group PAP {...}
>>> [pap] login attempt with password "Q?????? ??????p???F?+??a"
>>> [pap] Using clear text password "1111"
>>> [pap] Passwords don't match
>>> ++[pap] returns reject
>>> Failed to authenticate the user.
>>>   WARNING: Unprintable characters in the password.        Double-check the shared secret on the server and the NAS!
>>>
>>> Using Post-Auth-Type Reject
>>> # Executing group from file /usr/local/etc/raddb/sites-enabled/default
>>> +- entering group REJECT {...}
>>> [attr_filter.access_reject]     expand: %{User-Name} -> 1001
>>>  attr_filter: Matched entry DEFAULT at line 11
>>> ++[attr_filter.access_reject] returns updated
>>> Delaying reject of request 38 for 1 seconds
>>>
>>> Regards,
>>> Charles
>>>
>>>
>>> 2011/8/5 Tihomir Culjaga <tculjaga at gmail.com>
>>>
>>>> add to rad_auth.conf.xml
>>>>
>>>> <param name="NAS-Port-Type" id="61" value="0" pec="0" expr="0"
>>>> direction="in"/>
>>>> <param name="Login-User" id="6" value="1" pec="0" expr="0"
>>>> direction="in"/>
>>>>
>>>>
>>>>
>>>> as for Auth Type im not sure if you need it ... this is up to your
>>>> server.
>>>> According to dictionary file you need to set it as follows:
>>>>
>>>> <param name="Auth-Type" id="1000" value="?" pec="0" expr="0"
>>>> direction="in"/>
>>>>
>>>> the value (set as ?) is one of the folowing. Again, not sure what is
>>>> required by your server.
>>>>
>>>> VALUE           Auth-Type               Local                   0
>>>> VALUE           Auth-Type               System                  1
>>>> VALUE           Auth-Type               SecurID                 2
>>>> VALUE           Auth-Type               Crypt-Local             3
>>>> VALUE           Auth-Type               Reject                  4
>>>>
>>>> #
>>>> #       Cistron extensions
>>>> #
>>>> VALUE           Auth-Type               Pam                     253
>>>> VALUE           Auth-Type               Accept                  254
>>>>
>>>>
>>>>
>>>> regards,
>>>> Tihomir.
>>>>
>>>>
>>>>
>>>> On Wed, Aug 3, 2011 at 6:32 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>>>>
>>>>> Hi Tihomir,
>>>>>
>>>>> Sorry, i missed your mail in gmail before, just now saw it, and after
>>>>> using your dictionary.all, the dictionary issue was resolved, very
>>>>> appreciated for your kindly help! however, it did not fully functional yet,
>>>>>
>>>>> Attached are configuration files that i used, when i dial 601 to
>>>>> trigger to auth, the freeradius server shows log below, the supecious log is
>>>>> the value User-Password, it should be '1111' that i've set in the mysql db
>>>>> of freeradisu server for the user 1001 .
>>>>>
>>>>> i searched in google, for "known good" password issue, i suggest
>>>>> change user-password to cleartext-password, however, i did not find where it
>>>>> is.
>>>>> and also the Auth-Type, where to configure it...
>>>>>
>>>>> Freeradius server log:
>>>>>
>>>>> rad_recv: Access-Request packet from host 127.0.0.1 port 52684, id=49,
>>>>> length=111
>>>>>         User-Name = "1001"
>>>>>         User-Password = "?\210\365@\263\t\306\343\243iT?\311C\t\002"
>>>>>         Called-Station-Id = "888"
>>>>>         h323-conf-id = "749d2b5a-16ad-48e4-af58-24011949d1b5"
>>>>>         Calling-Station-Id = "1001"
>>>>>         NAS-Port = 0
>>>>>         NAS-IP-Address = 127.0.0.1
>>>>> # Executing section authorize from file
>>>>> /usr/local/etc/raddb/sites-enabled/default
>>>>> +- entering group authorize {...}
>>>>> ++[preprocess] returns ok
>>>>> [auth_log]      expand:
>>>>> /usr/local/var/log/radius/radacct/%{Client-IP-Address}/auth-detail-%Y%m%d ->
>>>>> /usr/local/var/log/radius/radacct/127.0.0.1/auth-detail-20110803
>>>>> [auth_log]
>>>>> /usr/local/var/log/radius/radacct/%{Client-IP-Address}/auth-detail-%Y%m%d
>>>>> expands to /usr/local/var/log/radius/radacct/
>>>>> 127.0.0.1/auth-detail-20110803
>>>>> [auth_log]      expand: %t -> Wed Aug  3 12:06:33 2011
>>>>> ++[auth_log] returns ok
>>>>> ++[chap] returns noop
>>>>> ++[mschap] returns noop
>>>>> ++[digest] returns noop
>>>>> [suffix] No '@' in User-Name = "1001", looking up realm NULL
>>>>> [suffix] No such realm "NULL"
>>>>> ++[suffix] returns noop
>>>>> [eap] No EAP-Message, not doing EAP
>>>>> ++[eap] returns noop
>>>>> ++[unix] returns notfound
>>>>> ++[files] returns noop
>>>>> [sql]   expand: %{User-Name} -> 1001
>>>>> [sql] sql_set_user escaped user --> '1001'
>>>>> rlm_sql (sql): Reserving sql socket id: 4
>>>>> [sql]   expand: SELECT id, username, attribute, value, op
>>>>> FROM radcheck           WHERE username = '%{SQL-User-Name}'           ORDER
>>>>> BY id -> SELECT id, username, attribute, value, op           FROM
>>>>> radcheck           WHERE username = '1001'           ORDER BY id
>>>>> [sql]   expand: SELECT groupname           FROM radusergroup
>>>>> WHERE username = '%{SQL-User-Name}'           ORDER BY priority -> SELECT
>>>>> groupname           FROM radusergroup           WHERE username =
>>>>> '1001'           ORDER BY priority
>>>>> rlm_sql (sql): Released sql socket id: 4
>>>>> [sql] User 1001 not found
>>>>> ++[sql] returns notfound
>>>>> ++[expiration] returns noop
>>>>> ++[logintime] returns noop
>>>>> [pap] WARNING! No "known good" password found for the user.
>>>>> Authentication may fail because of this.
>>>>> ++[pap] returns noop
>>>>> ERROR: No authenticate method (Auth-Type) found for the request:
>>>>> Rejecting the user
>>>>> Failed to authenticate the user.
>>>>>   WARNING: Unprintable characters in the password.        Double-check
>>>>> the shared secret on the server and the NAS!
>>>>> Using Post-Auth-Type Reject
>>>>> # Executing group from file /usr/local/etc/raddb/sites-enabled/default
>>>>> +- entering group REJECT {...}
>>>>> [attr_filter.access_reject]     expand: %{User-Name} -> 1001
>>>>>  attr_filter: Matched entry DEFAULT at line 11
>>>>> ++[attr_filter.access_reject] returns updated
>>>>> Delaying reject of request 8 for 1 seconds
>>>>> Going to the next request
>>>>> Waking up in 0.9 seconds.
>>>>> Sending delayed reject for request 8
>>>>> Sending Access-Reject of id 49 to 127.0.0.1 port 52684
>>>>> Waking up in 4.9 seconds.
>>>>> Cleaning up request 8 ID 49 with timestamp +7674
>>>>> Ready to process requests.
>>>>> WARNING! No "known good" password found for the user
>>>>>
>>>>> Regards,
>>>>> Charles
>>>>>
>>>>>
>>>>> 2011/8/3 Tihomir Culjaga <tculjaga at gmail.com>
>>>>>
>>>>>> did u use the dictionary i have attached ?
>>>>>>
>>>>>>
>>>>>> On Tue, Aug 2, 2011 at 10:08 AM, fieldpeak <fieldpeak at gmail.com>wrote:
>>>>>>
>>>>>>> i tried change to 'h323-conf-id' to 'h323-call-origin' in
>>>>>>> 02_unitest_rad-ANI-auth.xml, rad_auth.conf.xml, however, it still prompt
>>>>>>> '[ERR] mod_rad_auth.c:428 Unknown attribute: key:h323-conf-id, not
>>>>>>> found in dictionary', so where the mod_rad_auth read out the 'h323-conf-id'?
>>>>>>> very very strange, which dictionary it was using...
>>>>>>>
>>>>>>> Regards,
>>>>>>> Charles
>>>>>>>
>>>>>>>
>>>>>>> 2011/8/2 fieldpeak <fieldpeak at gmail.com>
>>>>>>>
>>>>>>>> Hi Tihomir,
>>>>>>>>
>>>>>>>> Finally the answer coming, i see the hope, thanks for your reply, :)
>>>>>>>>
>>>>>>>> As your advise, i only use one attribute(h323-conf-id) in my
>>>>>>>> dialplan, and only one attribute(h323-conf-id) in rad_auth.conf.xml, and
>>>>>>>> using the attached dictionary (from ciso) which contains this attribute,
>>>>>>>> however, it still prompt 'unknown attribute', so i suspected if it was
>>>>>>>> reading /usr/local/etc/radiusclient/dictionary, so i copy the same
>>>>>>>> dictionary to /usr/local/freeswitch/radius/, it did not any help at all...
>>>>>>>> very strange...
>>>>>>>>
>>>>>>>> Log:
>>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>> default_realm := .
>>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>> radius_timeout := 3.
>>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>> radius_retries := 2.
>>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>> radius_deadtime := 0.
>>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set bindaddr
>>>>>>>> := *.
>>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:371 ... radius:
>>>>>>>> User-Name: 38516060333
>>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:380 ... radius:
>>>>>>>> User-Password: 003282
>>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:396 ... radius:
>>>>>>>> Called-station-Id: 16094191500
>>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:413 Handle
>>>>>>>> attribute: h323-conf-id
>>>>>>>> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:428 Unknown
>>>>>>>> attribute: key:h323-conf-id, not found in dictionary
>>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:538 abort sending
>>>>>>>> radius packet.
>>>>>>>> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:546 An error occured
>>>>>>>> during RADIUS Authentication(RC=-1)
>>>>>>>> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:702 An error occured
>>>>>>>> during radius authorization.
>>>>>>>>
>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  AUTH_RESULT=)
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>   <extension name="unitest_rad-ANI-auth">
>>>>>>>>     <condition field="destination_number" expression="^601$">
>>>>>>>>       <!-- <action application="log" data="INFO  Before Auth "/> -->
>>>>>>>>
>>>>>>>>       <action inline="true" application="set" data="CALLID=
>>>>>>>> h323-conf-id=${uuid}"/>
>>>>>>>>
>>>>>>>>       <action inline="true" application="set" data="USERNAME=1001"/>
>>>>>>>>       <action inline="true" application="set" data="PASSWD=1111"/>
>>>>>>>>
>>>>>>>>
>>>>>>>>       <action application="sleep" data="2000"/>
>>>>>>>>       <action application="auth_function" data="in ${DIALED_NUMBER},
>>>>>>>> in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>
>>>>>>>>
>>>>>>>>     </condition>
>>>>>>>>   </extension>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> <configuration name="rad_auth.conf" description="radius
>>>>>>>> authentification module">
>>>>>>>>   <settings>
>>>>>>>>
>>>>>>>>   </settings>
>>>>>>>>
>>>>>>>>   <client>
>>>>>>>>     <param name="authserver" value="127.0.0.1:1812:gateway"/>
>>>>>>>>     <param name="dictionary"
>>>>>>>> value="/usr/local/etc/radiusclient/dictionary"/>
>>>>>>>>     <param name="seqfile" value="/var/run/radius.seq"/>
>>>>>>>>     <param name="mapfile"
>>>>>>>> value="/usr/local/etc/radiusclient/port-id-map"/>
>>>>>>>>     <param name="default_realm" value=""/>
>>>>>>>>     <param name="radius_timeout" value="3"/>
>>>>>>>>     <param name="radius_retries" value="2"/>
>>>>>>>>     <param name="radius_deadtime" value="0"/>
>>>>>>>>     <param name="bindaddr" value="*"/>
>>>>>>>>   </client>
>>>>>>>>
>>>>>>>>   <vsas>
>>>>>>>>
>>>>>>>>
>>>>>>>>     <param name="h323-conf-id" id="24" value="CALLID" pec="9"
>>>>>>>> expr="1" direction="in"/>
>>>>>>>>
>>>>>>>>   </vsas>
>>>>>>>>  </configuration>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> 2011/8/2 Tihomir Culjaga <tculjaga at gmail.com>
>>>>>>>>
>>>>>>>>> hi,
>>>>>>>>>
>>>>>>>>> dictionary.all is just the name of a file containing all attributes
>>>>>>>>> i needed at that time.
>>>>>>>>>
>>>>>>>>> you can include other dictionaries by putting #INCLUDE <pathname>
>>>>>>>>> at the end of the dictionary file you reference in rad_auth.conf.xml.
>>>>>>>>> if the INCLUDE doesn't work, just append dictionary.cisco to your
>>>>>>>>> dictionary file... and make your own file.
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> check inline comments down below...
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> T.
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On Sun, Jul 31, 2011 at 10:46 AM, fieldpeak <fieldpeak at gmail.com>wrote:
>>>>>>>>>
>>>>>>>>>> Hello Gurus,
>>>>>>>>>>
>>>>>>>>>> i met a issue when using
>>>>>>>>>> mod_rad_auth(http://wiki.freeswitch.org/wiki/Mod_rad_auth) to
>>>>>>>>>> works
>>>>>>>>>> with freeradius server+mysql for AAA, the details is below, Could
>>>>>>>>>> anyone give any hints, Thanks in advance.
>>>>>>>>>>
>>>>>>>>>> i setup a dial plan "unitest_rad-ANI-auth" as wiki above, however,
>>>>>>>>>> when i dialed 601 to trigger the dial plan, the console show
>>>>>>>>>> errors,
>>>>>>>>>> it looks "h323-conf-id" is not in the directory, then i tried to
>>>>>>>>>> add
>>>>>>>>>> this attribute to the dictionary, however, it does not help, in
>>>>>>>>>> the
>>>>>>>>>> wiki, it mentioned the rad_auth.conf.xml contains <param
>>>>>>>>>> name="dictionary"
>>>>>>>>>> value="/usr/local/etc/radiusclient/dictionary.all"/>, however i
>>>>>>>>>> did
>>>>>>>>>> not find the file "dictionary.all" at that directory, so i use
>>>>>>>>>> dictionary. BTW, the freeradius server + mysql works well.
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>> i just appended the information needed into dictionary.all file...
>>>>>>>>> (vendor and attribute definition).
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> console errors:
>>>>>>>>>>
>>>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 auth_function(in , in
>>>>>>>>>> 38516060333, in 003282, out AUTH_RESULT)
>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:301 allocate
>>>>>>>>>> initial
>>>>>>>>>> structure.
>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:313 initialzed
>>>>>>>>>> configuration.
>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>>> authserver
>>>>>>>>>> := 127.0.0.1:1812:gateway.
>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>>> dictionary
>>>>>>>>>> := /usr/local/etc/radiusclient/dictionary.
>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set seqfile
>>>>>>>>>> :=
>>>>>>>>>> /var/run/radius.seq.
>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set mapfile
>>>>>>>>>> :=
>>>>>>>>>> /usr/local/etc/radiusclient/port-id-map.
>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>>> default_realm := .
>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>>> radius_timeout := 3.
>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>>> radius_retries := 2.
>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>>> radius_deadtime := 0.
>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set bindaddr
>>>>>>>>>> := *.
>>>>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:371 ... radius:
>>>>>>>>>> User-Name: 38516060333
>>>>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:380 ... radius:
>>>>>>>>>> User-Password: 003282
>>>>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:391 ... radius:
>>>>>>>>>> Called-station-Id is empty, ignoring...
>>>>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:413 Handle
>>>>>>>>>> attribute: h323-conf-id
>>>>>>>>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:428 Unknown
>>>>>>>>>> attribute:
>>>>>>>>>> key:h323-conf-id, not found in dictionary
>>>>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:538 abort
>>>>>>>>>> sending
>>>>>>>>>> radius packet.
>>>>>>>>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:546 An error
>>>>>>>>>> occured
>>>>>>>>>> during RADIUS Authentication(RC=-1)
>>>>>>>>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:702 An error
>>>>>>>>>> occured
>>>>>>>>>> during radius authorization.
>>>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO
>>>>>>>>>>  AUTH_RESULT=)
>>>>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  AUTH_RESULT=
>>>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO
>>>>>>>>>>  billing_model=)
>>>>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202
>>>>>>>>>>  billing_model=
>>>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO
>>>>>>>>>>  credit_amount=)
>>>>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202
>>>>>>>>>>  credit_amount=
>>>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  currency=)
>>>>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  currency=
>>>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO
>>>>>>>>>>  preffered_lang=)
>>>>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202
>>>>>>>>>>  preffered_lang=
>>>>>>>>>>
>>>>>>>>>> added below in the
>>>>>>>>>> dictionary(/usr/local/etc/radiusclient/dictionary):
>>>>>>>>>>
>>>>>>>>>> ATTRIBUTE       h323-conf-id            1008    string
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>> you need the vendor definition as well
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> dial plan:
>>>>>>>>>> <extension name="unitest_rad-ANI-auth">
>>>>>>>>>>    <condition field="destination_number" expression="^601$">
>>>>>>>>>>      <action application="log" data="INFO  Before Auth "/>
>>>>>>>>>>
>>>>>>>>>>      <action inline="true" application="set"
>>>>>>>>>> data="CALLID=h323-conf-id=${uuid}"/>
>>>>>>>>>>      <action inline="true" application="set"
>>>>>>>>>> data="SERVICENUM=h323-prompt-id=${destination_number}"/>
>>>>>>>>>>      <action inline="true" application="set"
>>>>>>>>>> data="TRANSACTIONID=h323-ivr-out=transactionID:1234"/>
>>>>>>>>>>  <!--      <action inline="true" application="set"
>>>>>>>>>> data="CALLINGNUMBER=${caller_id_number}"/> -->
>>>>>>>>>>      <action inline="true" application="set"
>>>>>>>>>> data="CALLINGNUMBER=38516060333"/>
>>>>>>>>>>      <action inline="true" application="set"
>>>>>>>>>> data="USERNAME=38516060333"/>
>>>>>>>>>>  <!--      <action inline="true" application="set"
>>>>>>>>>> data="USERNAME=209354"/> -->
>>>>>>>>>>      <action inline="true" application="set"
>>>>>>>>>> data="PASSWD=003282"/>
>>>>>>>>>>  <!--      <action inline="true" application="set"
>>>>>>>>>> data="DIALED_NUMBER=16094191500"/>  -->
>>>>>>>>>>
>>>>>>>>>>      <action application="sleep" data="2000"/>
>>>>>>>>>>      <action application="auth_function" data="in
>>>>>>>>>> ${DIALED_NUMBER},
>>>>>>>>>> in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>>>  AUTH_RESULT=${AUTH_RESULT}"/>
>>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>>>  billing_model=${billing_model}"/>
>>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>>>  credit_amount=${credit_amount}"/>
>>>>>>>>>>      <action application="log" data="INFO  currency=${currency}"/>
>>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>>>  preffered_lang=${preffered_lang}"/>
>>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>>>  credit_time=${credit_time}"/>
>>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>>> h323_ivr_duration=${h323_ivr_duration}"/>
>>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>>>  return_code=${return_code}"/>
>>>>>>>>>>      <!-- <action application="execute_extension" data="AUTH XML
>>>>>>>>>> default"/> -->
>>>>>>>>>>    </condition>
>>>>>>>>>>  </extension>
>>>>>>>>>>
>>>>>>>>>>  radius_cdr.conf.xml:
>>>>>>>>>>  <configuration name="radius_cdr.conf" description="RADIUS CDR
>>>>>>>>>> Configuration">
>>>>>>>>>>
>>>>>>>>>>        <settings>
>>>>>>>>>>
>>>>>>>>>>                <!-- location of the radius dictionary files -->
>>>>>>>>>>
>>>>>>>>>>                <param name="dictionary"
>>>>>>>>>> value="/usr/local/freeswitch/conf/radius/dictionary"/>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>> your dictionary file need to contain all the attributes you are
>>>>>>>>> trying to use or to include other dictionaries (In this case
>>>>>>>>> dictionary.cisco) from the dictionary file you are referencing here.
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>>                <!-- number of retries for each server -->
>>>>>>>>>>
>>>>>>>>>>                <param name="radius_retries" value="3"/>
>>>>>>>>>>
>>>>>>>>>>                <!-- number of seconds to wait between retries -->
>>>>>>>>>>
>>>>>>>>>>                <param name="radius_timeout" value="5"/>
>>>>>>>>>>
>>>>>>>>>>                <!-- accounting servers, up to 8 allowed -->
>>>>>>>>>>
>>>>>>>>>>                <!-- value is "host:port:secret", port is optional
>>>>>>>>>> -->
>>>>>>>>>>
>>>>>>>>>>                <!-- use IP ADDRESSES, not hostnames -->
>>>>>>>>>>
>>>>>>>>>>                <param name="acct_server" value="127.0.0.1:1813
>>>>>>>>>> :testing123"/>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>        </settings>
>>>>>>>>>>
>>>>>>>>>> </configuration>
>>>>>>>>>>
>>>>>>>>>>  the FS version:
>>>>>>>>>>  FreeSWITCH Version 1.0.head (git-492bc6b 2011-07-23 12-53-04
>>>>>>>>>> -0400)
>>>>>>>>>>
>>>>>>>>>>  Regards,
>>>>>>>>>>  Charles
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110808/b5ce2ed0/attachment-0001.html 

From jmoran at secureachsystems.com  Mon Aug  8 22:12:47 2011
From: jmoran at secureachsystems.com (Jason Moran)
Date: Mon, 8 Aug 2011 14:12:47 -0400
Subject: [Freeswitch-users] how to have fs interrupt a file or tts
	ifdigit	is pressed
References: <23058.1312823337@ccs.covici.com>
	<6A6B4C284AD15042B429EB9D904544AD021FD8A4A6@NY1-EXMB-01.ip-soft.net>
Message-ID: <361E98F99D3CC3439EED59BC1924ED69546843@SERVER2003.SecuReachSystems.local>

This is in javascript instead of Perl, but you can get the gist of it
where you would call the getDTMFPlaybackStopOnAny function:

function GetDTMFStopOnAny(session, type, data, arg)
{
   arg.digits += data.digit;
   return false;
}
function getDTMFPlaybackStopOnAny(playbackFile)
{
   var dtmf = new Object();
   dtmf.digits = "";
   session.flushDigits();
   session.streamFile(playbackFile, GetDTMFStopOnAny, dtmf);
   if(dtmf.digits == "") { session.collectInput(GetDTMFStopOnAny, dtmf,
15000); }
   var returnDTMF = ReplaceAll(dtmf.digits, "#", "");
   return returnDTMF;
}

-----Original Message-----
From: Hector Geraldino [mailto:Hector.Geraldino at ip-soft.net] 
Sent: Monday, August 08, 2011 1:30 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] how to have fs interrupt a file or tts
ifdigit is pressed

You can send a "break" command in the DTMF callback event. This will
stop the audio playback or TTS.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
covici at ccs.covici.com
Sent: Monday, August 08, 2011 1:09 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] how to have fs interrupt a file or tts if
digit is pressed

Hi.  I am using Perl, if that makes any difference and I would like to
arrange things in such a way that during a prompt, if the person presses
a key, then the prompt will stop speaking and I can see what the key is
and do something.  Here is an excerpt of the script I am using.

$session->setInputCallback('got_press',""); #listen for key presses in
the background


while($session->ready())
{
$session->streamFile("test_break.wav");

    if ($press_so_far != "")
    {
        $session->say($press_so_far,"EN", "NAME_SPELLED", "ITERATED");
        $press_so_far="";
            }
}
 
Instead of streamFile, I tried speaking text through tts_command line,
and a phrase macro which did the same.  I was sure that at least
streamFile would break, but it did not.

Thanks in advance for any ideas.

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com

_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

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




From lists at telefaks.de  Tue Aug  9 00:34:25 2011
From: lists at telefaks.de (Peter Steinbach)
Date: Mon, 08 Aug 2011 22:34:25 +0200
Subject: [Freeswitch-users] Fidelio
In-Reply-To: <062201cc512b$f308b720$d91a2560$@com>
References: <CAHKqMb5dnE1mgdFaV65R=tk-zxvfz30=hi+3AyxezENQXVsrYw@mail.gmail.com>	<CAB2KLyfDvLjmqUU2coOJzG6ss6xzKvsGD5OOsS+bVwKaXx+P0g@mail.gmail.com>	<CAFiqYukzYyKZ_Tj8w8PFrM4FFpXzZiguRYNfqP8H=ah0iZLkhA@mail.gmail.com>	<CAB2KLycjzFtLc8jTBM9377CDi488K9nDmVD=Zz-13MZ7_w-cxA@mail.gmail.com>	<CABH8QX2y1mu1L=9-KZk2HH-Xh9MFOS98p4ctvvQrF7AW_RJV_A@mail.gmail.com>	<CAHKqMb5uAypVLHgqkt+GFNC_oj5wDPnFbzKHxjwr0QhcEhYJPw@mail.gmail.com>	<CABH8QX1HHvqL_YK0rbs-m1Ky1GtbZ6OQKyKMKkAne8XV5WBSig@mail.gmail.com>	<CAHKqMb51RbLBo+pmKwhJ82=ZWaGcyM6bd6y881Vh=4cAjztt5A@mail.gmail.com>	<CABH8QX0cgqbPcCtS=2Z0ron_vqX=5-YMrnkGJgqsOYXx81+x9Q@mail.gmail.com>	<4E37FED3.7080306@anew.com.ve>	<CABH8QX3+gukdj_wLvefbKd_YVEq2Rv2Y8FcDpQ+Xc_cGD+7UWA@mail.gmail.com>
	<062201cc512b$f308b720$d91a2560$@com>
Message-ID: <4E404851.9030301@telefaks.de>

Hello

we have been developing a Fidelio interface which works with Freeswitch. 
It's actually based on an Asterisk interface, where we emulate the 
Asterisk side. If anybody is interested, just contact me.

Best regards
Peter


Am 02.08.2011 17:50, schrieb Robert Huddleston:
>
> Maybe we should start a new mailing list -- as this Fidelio 
> conversation is very small audienced
>
> *From:*freeswitch-users-bounces at lists.freeswitch.org 
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of 
> *Jo?o Mesquita
> *Sent:* Tuesday, August 02, 2011 11:32 AM
> *To:* Saugort Dario Garcia Tovar
> *Cc:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Fidelio
>
> Dear Dario,
>
> Ante todo, un saludo de uno m?s que habla espa?ol!
>
> I have worked with an Argentinian company called BitSense for them to 
> develop their hospitality solution called Coral. They have CSTA 
> integrated with Siemens amongst other things and their base is in fact 
> FreeSWITCH. I am unsure of how many hotels they managed to install and 
> their integration with Fidelio. I have worked in the past for a 
> brazilian company that has a similar product (I think they are the 
> current Siemens solution) and I do know ppl inside of Micros, hence my 
> last email.
>
> Yo conozco a ANEW por otros motivos ajenos a FS y de hecho cuando 
> recibi tu mail me puse en contacto con Jhonny Lagos, pero me enter? 
> que ya no esta mas en ANEW. Est?n buscando una soluci?n hotelera? Te 
> puedo presentar a esta gente, son clientes mios. Ser? un gusto 
> ayudarlos en lo que pueda.
>
> Saludos,
> Jo?o Mesquita
>
>
> 2011/8/2 Saugort Dario Garcia Tovar <dgarcia at anew.com.ve 
> <mailto:dgarcia at anew.com.ve>>
>
> Hello all
> If you have money to afford the certification process go ahead, at the 
> end you will get back the money if you see that you can put a hotel 
> solution based on FS in your local market.
>
> I am new in FS world. I barely
>
> I used to work with Alcatel-Lucent OXE (IP-PBX). First the OXE, by 
> itself, offer hotel funcionality: voicemail for guest, guest room 
> admin (ready/free, cleaning, occupied,etc), alarm clock (at the hour 
> programmed by the guest from their phone or for "hotel phone console" 
> or admin interfase; the phone ring), fax service, room billing (quite 
> different fron a normal cdr), etc, etc, etc. Second, OXE offer a 
> propetary protocol to make possible PMS like fidelio to interface 
> with, that means, Fidelio become the administrative interface, so from 
> fidelio control room status, services, etc.
>
> So the first, do you have to adapt FS to offer a minimum of hotel 
> funcionality. Second about make a PMS-FS integration could be two 
> options or more: 1) adapt FS to PMS (using the integration doc); 2) 
> many pbx vendors offer info about their hotel link protocol for free 
> (not Alcatel), and are supported by fidelio like, OXE does, so you 
> could develop a interface that emulate an already supported hotel link 
> towards Fidelio and FS.
>
> I have a question:  has someone work with FS to offer an hotel solution?
>
>
>
>
>
> On 8/1/2011 9:19 PM, Jo?o Mesquita wrote:
>
>     Like I said, if you can raise enough money to be able to get the
>     certification going, I can talk to the ppl I know on Micros and
>     see if we can get FS certified (no guarantees), but it will cost
>     at least 5k USD, so I really don't know if there is enough interest.
>
>     Regards,
>     Jo?o Mesquita
>
>
>     On Mon, Aug 1, 2011 at 10:36 PM, Nandy Dagondon <gcd at i.ph
>     <mailto:gcd at i.ph>> wrote:
>
>     yes, certification is big obstacle. too much for our intended
>     project.  perhaps FS can apply for certification as a platform not
>     on a per-company/model basis. i just hope Fidelio would be open to
>     the idea.
>
>     another option would be to ask Asterisk-FIAS connectors like
>     PBillX the possibility to include FreeSwitch.
>
>     2011/8/2 Jo?o Mesquita <jmesquita at freeswitch.org
>     <mailto:jmesquita at freeswitch.org>>
>
>     I think there are none doing this at this moment. Here's what I
>     know tho:
>
>     1. I haven't seen the contracts or legal papers involved in the
>     certification
>     2. I've never done a connector even tho I worked with companies
>     that did in the past
>     3. I know that the homologation/certification costs money and not
>     little money. As far as I know, there are differences between
>     countries when it comes to the fee to be paid. Here is Argentina
>     it is something close to USD$5.000,00.
>     4. For you to be officially certified by them, you need to have
>     your system installed on at least 3 hotels with Fidelio, otherwise
>     you just lost money.
>
>     That should pretty much give you a hint of what you are facing
>     when it comes to the certification process. Does your project
>     accomodate these imposed barriers? If they do, I might be able to
>     help.
>
>     Regards,
>     Jo?o Mesquita
>
>
>
>
>     On Mon, Aug 1, 2011 at 9:38 PM, Nandy Dagondon <gcd at i.ph
>     <mailto:gcd at i.ph>> wrote:
>
>     hi Jo?o,
>
>     since you had FIAS experience, would a commercial  mod_fias
>     license/certification be possible just like g.729?
>
>     re companies selling FIAS connectors, i can't find one that
>     connects FS to Fidelio. otherwise, they would sound off in this
>     thread.
>
>     -nandy
>
>     2011/7/6 Jo?o Mesquita <jmesquita at freeswitch.org
>     <mailto:jmesquita at freeswitch.org>>
>
>         Guys, be careful because I think this document as well as the
>         protocol are confidential. I had to sign an NDA with Fidelio
>         to get my hands on it and pay a fee for it as well. You might
>         as well confirm it since you all seem to be in the US where
>         this type of information might be easier to get.
>
>         There are LOTS of companies selling their connectors to Fidelio...
>
>         One other point is that you need to have the certification
>         with them to be considered compatible, otherwise, no
>         consultant will install the connector on the fidelio side.
>
>         Regards,
>
>         Jo?o Mesquita
>
>
>
>
>         On Tue, Jul 5, 2011 at 1:52 PM, Luis F Urrea
>         <lfurrea at gmail.com <mailto:lfurrea at gmail.com>> wrote:
>
>         Awesome! great suggestions to get started,
>
>         There is also a FIAS simulator floating around.
>
>         That one may be a little harder to find? :)
>
>         On Tue, Jul 5, 2011 at 10:44 AM, Steven Ayre
>         <steveayre at gmail.com <mailto:steveayre at gmail.com>> wrote:
>
>         I'm assuming it's this document:
>         ftp://ftp.veracomp.com.pl/net/nomadix/Nomadix%20-%20PMS%20info/FIAS150.pdf
>         Quite easy to google once I had the version number.
>
>         You may find the nicest approach is to write a FOSS libfias,
>         then write an endpoint module to tie FS and libfias together.
>         Plenty of existing endpoint modules (mod_sofia mod_skinny
>         mod_opal mod_h323 etc) can show you examples to get you
>         started. Don't forget to read the FS API documentation too:
>         http://docs.freeswitch.org/
>
>         I'm assuming there are no license/patent restrictions to using
>         FIAS?
>
>         Good luck!
>
>         -Steve
>
>
>
>
>         On 5 July 2011 17:30, Luis F Urrea <lfurrea at gmail.com
>         <mailto:lfurrea at gmail.com>> wrote:
>
>         Hello Nandy,
>
>         A couple of months ago I started some research on the subject
>         and concluded I had to write my own interface to FS, however I
>         haven't had the time to get the project off ground yet.
>
>         I do have a copy of FIAS specification version 1.5 from 2001
>         which is publicly available I am sure it's not the latest but
>         it should cover the basics.
>
>         Please contact me off list if you have a hard time getting it
>         online.
>
>         Regards
>
>         On Mon, Jul 4, 2011 at 4:07 PM, Nandy Dagondon <gcd at i.ph
>         <mailto:gcd at i.ph>> wrote:
>
>             hi everybody,
>
>             anyone working on interfacing FS with Fidelio Hotel PMS?
>              i can't find the FIAS protocol/specs online.  is this
>             freely available?
>
>             tks,
>
>             nandy
>
>             _______________________________________________
>             Join us at ClueCon 2011, Aug 9-11, Chicago
>             http://www.cluecon.com 877-7-4ACLUE
>
>             FreeSWITCH-users mailing list
>             FreeSWITCH-users at lists.freeswitch.org
>             <mailto:FreeSWITCH-users at lists.freeswitch.org>
>             http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>             UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>             http://www.freeswitch.org
>
>
>         _______________________________________________
>         Join us at ClueCon 2011, Aug 9-11, Chicago
>         http://www.cluecon.com 877-7-4ACLUE
>
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>
>
>         _______________________________________________
>         Join us at ClueCon 2011, Aug 9-11, Chicago
>         http://www.cluecon.com 877-7-4ACLUE
>
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>
>
>         _______________________________________________
>         Join us at ClueCon 2011, Aug 9-11, Chicago
>         http://www.cluecon.com 877-7-4ACLUE
>
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>
>
>         _______________________________________________
>         Join us at ClueCon 2011, Aug 9-11, Chicago
>         http://www.cluecon.com 877-7-4ACLUE
>
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>
>
>     _______________________________________________
>     Join us at ClueCon 2011, Aug 9-11, Chicago
>     http://www.cluecon.com 877-7-4ACLUE
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>     _______________________________________________
>     Join us at ClueCon 2011, Aug 9-11, Chicago
>     http://www.cluecon.com 877-7-4ACLUE
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>     _______________________________________________
>     Join us at ClueCon 2011, Aug 9-11, Chicago
>     http://www.cluecon.com 877-7-4ACLUE
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>     _______________________________________________
>
>     Join us at ClueCon 2011, Aug 9-11, Chicago
>
>     http://www.cluecon.com  877-7-4ACLUE
>
>       
>
>     FreeSWITCH-users mailing list
>
>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
>     http://www.freeswitch.org
>
>     No virus found in this message.
>     Checked by AVG - www.avg.com <http://www.avg.com>
>     Version: 10.0.1390 / Virus Database: 1518/3802 - Release Date:
>     08/01/11
>
> -- 
> Atentamente,
> *Dario Garc?a*
> Consultor.
>
> CCCT, Nivel C2, Sector Yarey, Mz,
> Ofc. MZ03a.
> Caracas-Venezuela.
> Tel?fono: +58 212 9081842
> Cel: +58 412 2221515
> dgarcia at anew.com.ve <mailto:dgarcia at anew.com.ve>
> http://www.anew.com.ve
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
With kind regards
Peter Steinbach

Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de


-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110808/b6957fd3/attachment-0001.html 

From frankie.k.yiu at gmail.com  Tue Aug  9 01:14:58 2011
From: frankie.k.yiu at gmail.com (Frankie Yiu)
Date: Mon, 8 Aug 2011 14:14:58 -0700
Subject: [Freeswitch-users] Problem on some of of the calls (not calling /
 no Hangupcompleted event / Not released)
Message-ID: <CAGV=Xw1f4p6DP=ufSoHuyGg6Z9VGnEDER_RtawDZvL9y++M1Lg@mail.gmail.com>

Hi there,

For testing purpose, I have my freeswitch calling itself.
Once in awhile, I have originated a call, but calls don't seem to go through
/ or get "stuck".  I have code that would create record the whole session,
and also report the call status by subscribing the HangupCompleted
event--None of these happens.  Also when I check the status, the session was
not released/destroyed while most of the calls went through OK.

Could someone please tell me what might go wrong and how to fix this?

My freeswitch is downloaded on 7/27/2011.

Thanks,
Frankie
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110808/4b8e409f/attachment.html 

From rzhang at gosilverplus.com  Tue Aug  9 03:12:42 2011
From: rzhang at gosilverplus.com (ran zhang)
Date: Mon, 08 Aug 2011 16:12:42 -0700
Subject: [Freeswitch-users] please help! Invalid file format [wav]
Message-ID: <4E406D6A.9040307@gosilverplus.com>

i'm trying to play a sound when entering the conference, FS gives an 
error that 'switch_core_file.c122 Invalid file format [wav] for 
[/sounds/welcomef.wav]

I put '/sounds' as the sound prefix and welcomef.wav as the wave file 
for entering conference.

please help!!!
-------------- next part --------------
A non-text attachment was scrubbed...
Name: welcomef.wav
Type: audio/wav
Size: 17552 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110808/e59a3d3e/attachment.bin 

From dujinfang at gmail.com  Tue Aug  9 03:49:54 2011
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 9 Aug 2011 07:49:54 +0800
Subject: [Freeswitch-users] please help! Invalid file format [wav]
In-Reply-To: <4E406D6A.9040307@gosilverplus.com>
References: <4E406D6A.9040307@gosilverplus.com>
Message-ID: <85F79F4B89784CC6A328604E477D20F5@gmail.com>

I think it will be helpful if you put your dialplan entry and the full log on pastebin.freeswitch.org (http://pastebin.freeswitch.org) helps others helping you 

-- 
Seven Du
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj: http://www.freeswitch.org.cn
Sent with Sparrow (http://www.sparrowmailapp.com)

On Tuesday, August 9, 2011 at 7:12 AM, ran zhang wrote:

> i'm trying to play a sound when entering the conference, FS gives an 
> error that 'switch_core_file.c122 Invalid file format [wav] for 
> [/sounds/welcomef.wav]
> 
> I put '/sounds' as the sound prefix and welcomef.wav as the wave file 
> for entering conference.
> 
> please help!!!
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org)
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> Attachments: 
> - welcomef.wav
> 
> 


-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110809/1f217cba/attachment-0001.html 

From covici at ccs.covici.com  Tue Aug  9 04:36:23 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Mon, 08 Aug 2011 20:36:23 -0400
Subject: [Freeswitch-users] how to have fs interrupt a file or tts if
	digit is pressed
In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD021FD8A4A6@NY1-EXMB-01.ip-soft.net>
References: <23058.1312823337@ccs.covici.com>
	<6A6B4C284AD15042B429EB9D904544AD021FD8A4A6@NY1-EXMB-01.ip-soft.net>
Message-ID: <16927.1312850183@ccs.covici.com>

OK, how do I send a break command?

Hector Geraldino <Hector.Geraldino at ip-soft.net> wrote:

> You can send a "break" command in the DTMF callback event. This will stop the audio playback or TTS.
> 
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of covici at ccs.covici.com
> Sent: Monday, August 08, 2011 1:09 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] how to have fs interrupt a file or tts if digit is pressed
> 
> Hi.  I am using Perl, if that makes any difference and I would like to
> arrange things in such a way that during a prompt, if the person presses
> a key, then the prompt will stop speaking and I can see what the key is
> and do something.  Here is an excerpt of the script I am using.
> 
> $session->setInputCallback('got_press',""); #listen for key presses in
> the background
> 
> 
> while($session->ready())
> {
> $session->streamFile("test_break.wav");
> 
>     if ($press_so_far != "")
>     {
>         $session->say($press_so_far,"EN", "NAME_SPELLED", "ITERATED");
>         $press_so_far="";
>             }
> }
>  
> Instead of streamFile, I tried speaking text through tts_command line,
> and a phrase macro which did the same.  I was sure that at least
> streamFile would break, but it did not.
> 
> Thanks in advance for any ideas.
> 
> -- 
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
> 
>          John Covici
>          covici at ccs.covici.com
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From steveayre at gmail.com  Tue Aug  9 04:54:38 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 9 Aug 2011 01:54:38 +0100
Subject: [Freeswitch-users] please help! Invalid file format [wav]
In-Reply-To: <4E406D6A.9040307@gosilverplus.com>
References: <4E406D6A.9040307@gosilverplus.com>
Message-ID: <EEB1CBB7-EE2B-4D82-A399-3A9987B4442F@gmail.com>

Have you loaded mod_sndfile? That module provides support for reading wav files.

Steve on iPhone



On 9 Aug 2011, at 00:12, ran zhang <rzhang at gosilverplus.com> wrote:

> i'm trying to play a sound when entering the conference, FS gives an error that 'switch_core_file.c122 Invalid file format [wav] for [/sounds/welcomef.wav]
> 
> I put '/sounds' as the sound prefix and welcomef.wav as the wave file for entering conference.
> 
> please help!!!
> <welcomef.wav>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From stephen_wilkey at hotmail.com  Tue Aug  9 03:44:25 2011
From: stephen_wilkey at hotmail.com (Stephen Wilkey)
Date: Mon, 8 Aug 2011 23:44:25 +0000
Subject: [Freeswitch-users] intermittent EXCHANGE_ROUTING_ERROR
In-Reply-To: <BLU152-W45140868DE9EDDE504AE68E4210@phx.gbl>
References: <BLU152-W45140868DE9EDDE504AE68E4210@phx.gbl>
Message-ID: <BLU152-W4470860E41B53A27022CDDE4210@phx.gbl>







Thanks for the discussion on this issue. 

I have discovered that my ISP is sending calls to me with Max-Forwards=4 which means that I can't do much with the call when it arrives.  Is there a way to ignore or override the Max-Forwards setting that is provided with the call when it arrives so that I am able to process it? 

Regards, 
Stephen  		 	   		  
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110808/16c4454f/attachment.html 

From govoiper at gmail.com  Tue Aug  9 11:11:36 2011
From: govoiper at gmail.com (Sam Govind)
Date: Tue, 9 Aug 2011 12:11:36 +0500
Subject: [Freeswitch-users] SIP proxy collect DTMF using FS
Message-ID: <CAJUJwtjqMyvVX1eteYBWFf41Mz+fZjnRZA_0BR2DZs0R5zh_tA@mail.gmail.com>

Hi guys,

I'm looking to establish a scenario like this, any idea how to do it, if its
possible.

1- SIP proxy send call to FS where DTMF will be collected. (I'm thinking of
using PlayAndGetDigits)
2- DTMF collected be sent back to SIP proxy while FS ends the call
3- Call at SIP proxy end keeps running for some other processing.

basically I just need FS to collect DTMF and send those back to SIP Proxy.

Any ideas are welcome.

Thanks.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110809/26727a20/attachment-0001.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: FS-DTMF-Collect.jpg
Type: image/jpeg
Size: 25583 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110809/26727a20/attachment-0001.jpg 

From hugh at irvine.com.au  Tue Aug  9 04:55:43 2011
From: hugh at irvine.com.au (Hugh Irvine)
Date: Tue, 9 Aug 2011 10:55:43 +1000
Subject: [Freeswitch-users] Fidelio
In-Reply-To: <4E404851.9030301@telefaks.de>
References: <CAHKqMb5dnE1mgdFaV65R=tk-zxvfz30=hi+3AyxezENQXVsrYw@mail.gmail.com>	<CAB2KLyfDvLjmqUU2coOJzG6ss6xzKvsGD5OOsS+bVwKaXx+P0g@mail.gmail.com>	<CAFiqYukzYyKZ_Tj8w8PFrM4FFpXzZiguRYNfqP8H=ah0iZLkhA@mail.gmail.com>	<CAB2KLycjzFtLc8jTBM9377CDi488K9nDmVD=Zz-13MZ7_w-cxA@mail.gmail.com>	<CABH8QX2y1mu1L=9-KZk2HH-Xh9MFOS98p4ctvvQrF7AW_RJV_A@mail.gmail.com>	<CAHKqMb5uAypVLHgqkt+GFNC_oj5wDPnFbzKHxjwr0QhcEhYJPw@mail.gmail.com>	<CABH8QX1HHvqL_YK0rbs-m1Ky1GtbZ6OQKyKMKkAne8XV5WBSig@mail.gmail.com>	<CAHKqMb51RbLBo+pmKwhJ82=ZWaGcyM6bd6y881Vh=4cAjztt5A@mail.gmail.com>	<CABH8QX0cgqbPcCtS=2Z0ron_vqX=5-YMrnkGJgqsOYXx81+x9Q@mail.gmail.com>	<4E37FED3.7080306@anew.com.ve>	<CABH8QX3+gukdj_wLvefbKd_YVEq2Rv2Y8FcDpQ+Xc_cGD+7UWA@mail.gmail.com>
	<062201cc512b$f308b720$d91a2560$@com>
	<4E404851.9030301@telefaks.de>
Message-ID: <B64A48AE-B83D-425B-A122-AC19BCE4DFDF@irvine.com.au>


Hello Everyone -

Mike and I have also been doing a bit of work in this area.

Here is a copy of a recent post to the Radiator mailing list.

?..

We have recently released some documentation and sample configuration files 
showing how to use Radiator and the AuthBy FIDELIO module to
handle authentication and accounting for the Freeswitch VOIP switch
(http://www.freeswitch.org). It can be used authenticate and to bill VOIP
calls to a Micros-Fidelio Opera Hotel Property Management System
(http://www.micros.com).

The goal of this sample configuration is to implement a user-pays VOIP system
in a hotel environment:
Before a user can make a call from a hotel room VOIP phone, there must be
someone checked into the room. When the call is completed, the call is billed
to the hotel room.

Documentation and sample configuration files are now in the latest Radiator 
patch set.

We welcome feedback and suggestions from Freeswitch/Fidelio implementers.

?..

Radiator is a well known and widely used RADIUS server than includes support for Fidelio.

See www.open.com.au/radiator for additional information.

regards

Hugh


On 9 Aug 2011, at 06:34, Peter Steinbach wrote:

> Hello
> 
> we have been developing a Fidelio interface which works with Freeswitch. It's actually based on an Asterisk interface, where we emulate the Asterisk side. If anybody is interested, just contact me.
> 
> Best regards
> Peter
>  
> -- 
> With kind regards
> Peter Steinbach 
> 
> Telefaks Services GmbH
> 
> mailto:lists
>  (att) telefaks.de
> Internet: 
> www.telefaks.de
> 
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mghicas at gmail.com  Mon Aug  8 22:38:23 2011
From: mghicas at gmail.com (Mike Ghicas)
Date: Mon, 8 Aug 2011 14:38:23 -0400
Subject: [Freeswitch-users] FS/MRCP/SIP issue....
Message-ID: <CAF__LHcVmYRWQJu+V5OBunGoawfgTGhtoy43etH6AgKeNxvqng@mail.gmail.com>

I have a  very odd one - what to see if anyone else has had a similar issue.

Randomly after starting freeswitch, we get timeouts talking to nuance
(different host via MRCP v2)  - at first we thought it  was an issue
with nuance - however after digging a little bit we saw that FS did
not even attempt to open a connection to the Nuance host (tcpdump
shows noting at all...) and the connection times out.

Have tried running with sqllite and mysql via odbc Have tied different
combinations of FS/OS/vmware/etc No common pattern.

When this fails I stop and restart FS and sometimes it works,
sometimes it does not.
Sometimes I have to restart 2-10 times, other times it will work right away.

Has been working find on my dev host for a while, and I can not
replicate the issue.
Have installed and updated vmware tools, kernel, etc Had is sync to
VMwarehost and not for time NTP is in use.
Tried FS with -norc and -nocal options
Nothing seems to make this behavior stop - or at least be consistent.
Have reserved CPU cycles on Vmware
Have run under strace - nothing interesting - just sits there sleeping...

Spinning up a physical box for testing - but has anyone seen anything like this?
Only seems to affect it when we are opening an MRCP connection

Valgrind is next on the list.....

Feels like a vmware/os/fs clocking/timing issue

Thoughts?


Some more details below including log - would love to hear if anyone
has experiences anything like this..


As i was writing this I  tried the Physical host - same issue - so it
does not looks like a timing issue.... RHEL 5.2


--------
We have FS running in a VM on Vmware Vsphere (4.1.0 320137, 4.1.0
208167, 4.1.0 260247) and on a physical machine.
OS that has been tried RHEL 5.2, 5.5, 5.6

Working: Vmware 4.1.0 260247 Guest 5.5

Freeswitch - pulled from Git on 2011-07-27 Sofia and uni_mrcp enabled


We have tried multiple combinations of everything - only similarity is VMware

Our extension that we test with looks like this:
<extension name="unimrcp">
     <condition field="destination_number" expression="5189">
       <action application="answer"/>
       <action application="set" data="tts_engine=unimrcp:nuance5-mrcp2-prod"/>
       <action application="set" data="tts_voice=tom"/>
       <action application="speak" data="The roof, the roof, the roof
is on fire"/>
       <action application="sleep" data="100"/>
     </condition>
   </extension>

This talks to Nuance Speech Server (via MRC v2) on another host.


--

Here is the log from the console with debug enabled:


freeswitch at devipc02.nj3.ip-soft.net> tport_wakeup_pri(0x6f996f0): events IN
tport_recv_event(0x6f996f0)
tport_recv_iovec(0x6f996f0) msg 0x2aaac000c420 from
(udp/10.140.20.182:5060) has 1171 bytes, veclen = 1
tport_deliver(0x6f996f0): msg 0x2aaac000c420 (1171 bytes) from
udp/192.168.8.11:5060/sip next=(nil)
nta: received INVITE sip:5189 at 10.140.20.182:5060 SIP/2.0 (CSeq 101)
nta: canonizing sip:5189 at 10.140.20.182:5060 with contact
nta: INVITE (101) going to a default leg
nta: timer set to 200 ms
nua: nua_stack_process_request: entering
nua: nh_create: entering
nua: nh_create_handle: entering
nua: nua_stack_set_params: entering
soa_clone(static::0x6f80bd0, 0x6f67ec0, 0x2aaac000c110) called
soa_set_params(static::0x2aaac000d300, ...) called
nta_leg_tcreate(0x2aaac000da30)
soa_init_offer_answer(static::0x2aaac000d300) called
soa_set_remote_sdp(static::0x2aaac000d300, (nil), 0x2aaac000a0af, 212)
called
nua(0x2aaac000c110): adding session usage
tport_tsend(0x6f996f0) tpn = UDP/192.168.8.11:5060 tport_resolve
addrinfo = 192.168.8.11:5060
tport_by_addrinfo(0x6f996f0): not found by name UDP/192.168.8.11:5060
tport_vsend(0x6f996f0): 350 bytes of 350 to udp/192.168.8.11:5060
tport_vsend returned 350
nta: sent 100 Trying for INVITE (101)
nua(0x2aaac000c110): event i_invite 100 Trying
nua(0x2aaac000c110): call state changed: init -> received, received
offer soa_get_remote_sdp(static::0x2aaac000d300, [0x400ea8c8],
[0x400ea8c0], [(nil)]) called
nua(0x2aaac000c110): event i_state 100 Trying
nua: nua_application_event: entering
2011-08-08 18:13:36.135540 [NOTICE] switch_channel.c:812 New Channel
sofia/internal/5502 at 192.168.8.11
[148a57a7-c2af-45f2-8539-462b21f77a87]
nua: nua_handle_bind: entering
2011-08-08 18:13:36.135540 [DEBUG] switch_core_state_machine.c:325
(sofia/internal/5502 at 192.168.8.11) Running State Change CS_NEW
2011-08-08 18:13:36.135540 [DEBUG] switch_core_state_machine.c:343
(sofia/internal/5502 at 192.168.8.11) State NEW
nua: nua_handle_magic: entering
nua: nua_application_event: entering
2011-08-08 18:13:36.141412 [DEBUG] sofia.c:4761 Channel
sofia/internal/5502 at 192.168.8.11 entering state [received][100]
2011-08-08 18:13:36.141412 [DEBUG] sofia.c:4772 Remote SDP:
v=0
o=CiscoSystemsCCM-SIP 2000 1 IN IP4 192.168.8.11 s=SIP Call c=IN IP4
10.150.25.10
t=0 0
m=audio 16388 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

2011-08-08 18:13:36.141412 [DEBUG] sofia_glue.c:4651 Audio Codec
Compare [PCMU:0:8000:20:64000]/[G7221:115:32000:20:48000]
2011-08-08 18:13:36.141412 [DEBUG] sofia_glue.c:4651 Audio Codec
Compare [PCMU:0:8000:20:64000]/[G7221:107:16000:20:32000]
2011-08-08 18:13:36.141412 [DEBUG] sofia_glue.c:4651 Audio Codec
Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000]
2011-08-08 18:13:36.141412 [DEBUG] sofia_glue.c:4651 Audio Codec
Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
2011-08-08 18:13:36.141412 [DEBUG] sofia_glue.c:2774 Set Codec
sofia/internal/5502 at 192.168.8.11 PCMU/8000 20 ms 160 samples 64000
bits
2011-08-08 18:13:36.141412 [DEBUG] sofia_glue.c:4765 Set 2833 dtmf
send/recv payload to 101
2011-08-08 18:13:36.141412 [DEBUG] sofia.c:4943
(sofia/internal/5502 at 192.168.8.11) State Change CS_NEW -> CS_INIT
2011-08-08 18:13:36.141412 [DEBUG] switch_core_session.c:1116 Send
signal sofia/internal/5502 at 192.168.8.11 [BREAK]
nua: nua_handle_magic: entering
2011-08-08 18:13:36.141412 [DEBUG] switch_core_state_machine.c:325
(sofia/internal/5502 at 192.168.8.11) Running State Change CS_INIT
2011-08-08 18:13:36.141412 [DEBUG] switch_core_state_machine.c:361
(sofia/internal/5502 at 192.168.8.11) State INIT
2011-08-08 18:13:36.141412 [DEBUG] mod_sofia.c:84
sofia/internal/5502 at 192.168.8.11 SOFIA INIT
2011-08-08 18:13:36.141412 [DEBUG] mod_sofia.c:124
(sofia/internal/5502 at 192.168.8.11) State Change CS_INIT -> CS_ROUTING
2011-08-08 18:13:36.141412 [DEBUG] switch_core_session.c:1116 Send
signal sofia/internal/5502 at 192.168.8.11 [BREAK]
2011-08-08 18:13:36.141412 [DEBUG] switch_core_state_machine.c:361
(sofia/internal/5502 at 192.168.8.11) State INIT going to sleep
2011-08-08 18:13:36.141412 [DEBUG] switch_core_state_machine.c:325
(sofia/internal/5502 at 192.168.8.11) Running State Change CS_ROUTING
2011-08-08 18:13:36.141412 [DEBUG] switch_channel.c:1668
(sofia/internal/5502 at 192.168.8.11) Callstate Change DOWN -> RINGING
2011-08-08 18:13:36.141412 [DEBUG] switch_core_state_machine.c:364
(sofia/internal/5502 at 192.168.8.11) State ROUTING
2011-08-08 18:13:36.141412 [DEBUG] mod_sofia.c:147
sofia/internal/5502 at 192.168.8.11 SOFIA ROUTING
2011-08-08 18:13:36.141412 [DEBUG] switch_core_state_machine.c:77
sofia/internal/5502 at 192.168.8.11 Standard ROUTING
2011-08-08 18:13:36.141412 [INFO] mod_dialplan_xml.c:331 Processing
Mike Ghicas <5502>->5189 in context public
Dialplan: sofia/internal/5502 at 192.168.8.11 parsing [public->unloop]
continue=false
Dialplan: sofia/internal/5502 at 192.168.8.11 Regex (PASS) [unloop]
${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/5502 at 192.168.8.11 Regex (FAIL) [unloop]
${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/5502 at 192.168.8.11 parsing
[public->outside_call] continue=true
Dialplan: sofia/internal/5502 at 192.168.8.11 Absolute Condition [outside_call]
Dialplan: sofia/internal/5502 at 192.168.8.11 Action set(outside_call=true)
Dialplan: sofia/internal/5502 at 192.168.8.11 Action
set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
Dialplan: sofia/internal/5502 at 192.168.8.11 parsing
[public->call_debug] continue=true
Dialplan: sofia/internal/5502 at 192.168.8.11 Regex (FAIL) [call_debug]
${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/internal/5502 at 192.168.8.11 parsing
[public->public_extensions] continue=false
Dialplan: sofia/internal/5502 at 192.168.8.11 Regex (FAIL)
[public_extensions] destination_number(5189) =~ /^(10[01][0-9])$/
break=on-false
Dialplan: sofia/internal/5502 at 192.168.8.11 parsing
[public->public_did] continue=false
Dialplan: sofia/internal/5502 at 192.168.8.11 Regex (FAIL) [public_did]
destination_number(5189) =~ /^(5551212)$/ break=on-false
Dialplan: sofia/internal/5502 at 192.168.8.11 parsing [public->unimrcp]
continue=false
Dialplan: sofia/internal/5502 at 192.168.8.11 Regex (PASS) [unimrcp]
destination_number(5189) =~ /5189/ break=on-false
Dialplan: sofia/internal/5502 at 192.168.8.11 Action answer()
Dialplan: sofia/internal/5502 at 192.168.8.11 Action
set(tts_engine=unimrcp:nuance5-mrcp2-prod)
Dialplan: sofia/internal/5502 at 192.168.8.11 Action set(tts_voice=tom)
Dialplan: sofia/internal/5502 at 192.168.8.11 Action speak(The roof, the
roof, the roof is on fire.  we don't need no water let that mother
fucker burn.)
Dialplan: sofia/internal/5502 at 192.168.8.11 Action sleep(100)
2011-08-08 18:13:36.141412 [DEBUG] switch_core_state_machine.c:119
(sofia/internal/5502 at 192.168.8.11) State Change CS_ROUTING ->
CS_EXECUTE
2011-08-08 18:13:36.141412 [DEBUG] switch_core_session.c:1116 Send
signal sofia/internal/5502 at 192.168.8.11 [BREAK]
2011-08-08 18:13:36.141412 [DEBUG] switch_core_state_machine.c:364
(sofia/internal/5502 at 192.168.8.11) State ROUTING going to sleep
2011-08-08 18:13:36.141412 [DEBUG] switch_core_state_machine.c:325
(sofia/internal/5502 at 192.168.8.11) Running State Change CS_EXECUTE
2011-08-08 18:13:36.141412 [DEBUG] switch_core_state_machine.c:371
(sofia/internal/5502 at 192.168.8.11) State EXECUTE
2011-08-08 18:13:36.141412 [DEBUG] mod_sofia.c:240
sofia/internal/5502 at 192.168.8.11 SOFIA EXECUTE
2011-08-08 18:13:36.141412 [DEBUG] switch_core_state_machine.c:157
sofia/internal/5502 at 192.168.8.11 Standard EXECUTE EXECUTE
sofia/internal/5502 at 192.168.8.11 set(outside_call=true)
2011-08-08 18:13:36.141412 [DEBUG] mod_dptools.c:1060
sofia/internal/5502 at 192.168.8.11 SET [outside_call]=[true] EXECUTE
sofia/internal/5502 at 192.168.8.11 set(RFC2822_DATE=Mon, 08 Aug 2011
18:13:36 +0000)
2011-08-08 18:13:36.143981 [DEBUG] mod_dptools.c:1060
sofia/internal/5502 at 192.168.8.11 SET [RFC2822_DATE]=[Mon, 08 Aug 2011
18:13:36 +0000] EXECUTE sofia/internal/5502 at 192.168.8.11 answer()
2011-08-08 18:13:36.152965 [DEBUG] sofia_glue.c:3015 AUDIO RTP
[sofia/internal/5502 at 192.168.8.11] 10.140.20.182 port 16858 ->
10.150.25.10 port 16388 codec: 0 ms: 20
2011-08-08 18:13:36.152965 [DEBUG] switch_rtp.c:1623 Starting timer
[soft] 160 bytes per 20ms
2011-08-08 18:13:36.154960 [DEBUG] sofia_glue.c:3277 Set 2833 dtmf
send payload to 101
2011-08-08 18:13:36.154960 [DEBUG] sofia_glue.c:3282 Set 2833 dtmf
receive payload to 101
2011-08-08 18:13:36.154960 [DEBUG] mod_sofia.c:681 Local SDP
sofia/internal/5502 at 192.168.8.11:
v=0
o=FreeSWITCH 1312810358 1312810359 IN IP4 10.140.20.182 s=FreeSWITCH
c=IN IP4 10.140.20.182
t=0 0
m=audio 16858 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

nua: nua_respond: entering
nua(0x2aaac000c110): sent signal r_respond
2011-08-08 18:13:36.154960 [DEBUG] switch_core_session.c:709 Send
signal sofia/internal/5502 at 192.168.8.11 [BREAK]
2011-08-08 18:13:36.154960 [DEBUG] switch_channel.c:2830
(sofia/internal/5502 at 192.168.8.11) Callstate Change RINGING -> ACTIVE
2011-08-08 18:13:36.154960 [NOTICE] mod_dptools.c:930 Channel
[sofia/internal/5502 at 192.168.8.11] has been answered EXECUTE
sofia/internal/5502 at 192.168.8.11
set(tts_engine=unimrcp:nuance5-mrcp2-prod)
2011-08-08 18:13:36.156012 [DEBUG] mod_dptools.c:1060
sofia/internal/5502 at 192.168.8.11 SET
[tts_engine]=[unimrcp:nuance5-mrcp2-prod]
EXECUTE sofia/internal/5502 at 192.168.8.11 set(tts_voice=tom)
2011-08-08 18:13:36.156012 [DEBUG] mod_dptools.c:1060
sofia/internal/5502 at 192.168.8.11 SET [tts_voice]=[tom] EXECUTE
sofia/internal/5502 at 192.168.8.11 speak(The roof, the roof, the roof is
on fire.  we don't need no water let that mother fucker burn.)
2011-08-08 18:13:36.156947 [INFO] mod_unimrcp.c:1567 speech_handle:
name = unimrcp, rate = 8000, speed = 0, samples = 160, voice = ,
engine = unimrcp, param = nuance5-mrcp2-prod
2011-08-08 18:13:36.156947 [INFO] mod_unimrcp.c:1570 voice = tom, rate = 8000
2011-08-08 18:13:36.156947 [DEBUG] mod_unimrcp.c:652 (TTS-1) audio queue created
2011-08-08 18:13:36.156947 [NOTICE] mrcp_client.c:549 Create MRCP
Handle 0x703f1f0 [nuance5-mrcp2-prod]
2011-08-08 18:13:36.156947 [INFO] mrcp_client_session.c:142 Create
Channel 0x703f1f0 <new>
2011-08-08 18:13:36.156947 [DEBUG] mrcp_client.c:1038 Signal
Application Task Message
nua(0x2aaac000c110): recv signal r_respond 200 OK
nua: nua_stack_set_params: entering
soa_set_params(static::0x2aaac000d300, ...) called
soa_set_user_sdp(static::0x2aaac000d300, (nil), 0x2aaac0010a32, -1)
called
2011-08-08 18:13:36.156947 [DEBUG] mrcp_client.c:1006 Receive
Application Task Message [0]
2011-08-08 18:13:36.156947 [INFO] mrcp_client_session.c:398 Receive
App Request 0x703f1f0 <new> [2]
2011-08-08 18:13:36.156947 [INFO] mrcp_client.c:901 Add MRCP Handle 0x703f1f0
2011-08-08 18:13:36.156947 [DEBUG] mrcp_client_session.c:1203 Dispatch
Application Request 0x703f1f0 <new> [2]
2011-08-08 18:13:36.156947 [NOTICE] mrcp_client_session.c:718 Add
Control Channel 0x703f1f0 <new at speechsynth>
soa_set_capability_sdp(static::0x2aaac000d300, (nil), 0x2aaac0010a32,
-1) called
2011-08-08 18:13:36.156947 [DEBUG] mrcp_client_session.c:762 Add RTP
Termination 0x703f1f0 <new>
2011-08-08 18:13:36.156947 [DEBUG] apt_consumer_task.c:90 Wait for
Task Messages [MRCP Client]
nua: nua_invite_server_respond: entering
soa_generate_answer(static::0x2aaac000d300) called
soa_static_offer_answer_action(0x2aaac000d300, soa_generate_answer):
called soa_static(0x2aaac000d300, soa_generate_answer): generating
local description soa_static(0x2aaac000d300, soa_generate_answer):
upgrade with remote description soa_sdp_mode_set(0x400eacd0,
0x2aaac000e990, ""): called soa_static(0x2aaac000d300,
soa_generate_answer): storing local description
soa_activate(static::0x2aaac000d300, (nil)) called
soa_get_local_sdp(static::0x2aaac000d300, [(nil)], [0x400eae38],
[0x400eae44]) called
tport_tsend(0x6f996f0) tpn = UDP/192.168.8.11:5060 tport_resolve
addrinfo = 192.168.8.11:5060
tport_by_addrinfo(0x6f996f0): not found by name UDP/192.168.8.11:5060
tport_vsend(0x6f996f0): 1175 bytes of 1175 to udp/192.168.8.11:5060
tport_vsend returned 1175
nta: sent 200 OK for INVITE (101)
nua(0x2aaac000c110): call state changed: received -> completed, sent
answer soa_get_local_sdp(static::0x2aaac000d300, [0x400eaef8],
[0x400eaef0], [(nil)]) called soa_get_params(static::0x2aaac000d300,
...) called
nua(0x2aaac000c110): event i_state 200 OK
2011-08-08 18:13:36.156947 [DEBUG] mrcp_client_connection.c:603
Process Control Message
2011-08-08 18:13:36.156947 [DEBUG] mrcp_client.c:1104 Signal
Connection Task Message
2011-08-08 18:13:36.156947 [DEBUG] mrcp_client.c:974 Receive
Connection Task Message [0]
nua: nua_application_event: entering
2011-08-08 18:13:36.156947 [DEBUG] mrcp_client_session.c:311 On
Control Channel Add 0x703f1f0 <new>
2011-08-08 18:13:36.156947 [DEBUG] apt_consumer_task.c:90 Wait for
Task Messages [MRCP Client]
2011-08-08 18:13:36.156947 [DEBUG] sofia.c:4761 Channel
sofia/internal/5502 at 192.168.8.11 entering state [completed][200]
nua: nua_handle_magic: entering
2011-08-08 18:13:36.159000 [DEBUG] mpf_engine.c:302 Process MPF Message
2011-08-08 18:13:36.159000 [DEBUG] mpf_context.c:172 Add Context
2011-08-08 18:13:36.159000 [DEBUG] mpf_context.c:176 Add Termination
2011-08-08 18:13:36.159000 [DEBUG] mpf_context.c:176 Add Termination
2011-08-08 18:13:36.159000 [DEBUG] mrcp_client.c:999 Receive Media Task Message
2011-08-08 18:13:36.159000 [DEBUG] mrcp_client_session.c:1039 On
Termination Add 0x703f1f0 <new>
2011-08-08 18:13:36.159000 [DEBUG] mrcp_client_session.c:1039 On
Termination Add 0x703f1f0 <new>
2011-08-08 18:13:36.159000 [INFO] mrcp_client_session.c:420 Send Offer
0x703f1f0 <new> [c:1 a:1 v:0]
2011-08-08 18:13:36.159000 [INFO] mrcp_sofiasip_client_agent.c:316
Local SDP 0x703f1f0 <new>
v=0
o=FreeSWITCH 0 0 IN IP4 10.140.20.182
s=-
c=IN IP4 10.140.20.182
t=0 0
m=application 9 TCP/MRCPv2 1
a=setup:active
a=connection:new
a=resource:speechsynth
a=cmid:1
m=audio 4002 RTP/AVP 0 8 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 L16/8000
a=recvonly
a=mid:1

2011-08-08 18:13:36.159000 [DEBUG] apt_consumer_task.c:90 Wait for
Task Messages [MRCP Client]
tport_wakeup_pri(0x6f996f0): events IN
tport_recv_event(0x6f996f0)
tport_recv_iovec(0x6f996f0) msg 0x2aaac000fb00 from
(udp/10.140.20.182:5060) has 431 bytes, veclen = 1
tport_deliver(0x6f996f0): msg 0x2aaac000fb00 (431 bytes) from
udp/192.168.8.11:5060/sip next=(nil)
nta: received ACK sip:5189 at 10.140.20.182:5060;transport=udp SIP/2.0 (CSeq 101)
nta: ACK (101) is going to INVITE (101)
nua: process_ack_or_cancel: entering
soa_clear_remote_sdp(static::0x2aaac000d300) called
nua(0x2aaac000c110): event i_ack 200 OK
nua(0x2aaac000c110): call state changed: completed -> ready
nua(0x2aaac000c110): event i_state 200 OK
nua(0x2aaac000c110): event i_active 200 Call active
nua(): refresh session after 1768 seconds (in [1768..1768])
nua: nua_application_event: entering
nua: nua_handle_magic: entering
nua: nua_application_event: entering
2011-08-08 18:13:36.161958 [DEBUG] sofia.c:4761 Channel
sofia/internal/5502 at 192.168.8.11 entering state [ready][200]
nua: nua_handle_magic: entering
nua: nua_application_event: entering
nua: nua_handle_magic: entering
nta: timer set next to 4827 ms
2011-08-08 18:13:41.158959 [WARNING] mod_unimrcp.c:1010 (TTS-1) MRCP
session has not opened after 5000 ms
nta: timer I fired, terminate 200 response incoming_reclaim_all((nil),
(nil), 0x400ead40)
nta_incoming_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free
nta: timer not set


From mghicas at gmail.com  Tue Aug  9 01:49:28 2011
From: mghicas at gmail.com (Mike Ghicas)
Date: Mon, 8 Aug 2011 17:49:28 -0400
Subject: [Freeswitch-users] FS/MRCP/SIP issue....
In-Reply-To: <CAF__LHcVmYRWQJu+V5OBunGoawfgTGhtoy43etH6AgKeNxvqng@mail.gmail.com>
References: <CAF__LHcVmYRWQJu+V5OBunGoawfgTGhtoy43etH6AgKeNxvqng@mail.gmail.com>
Message-ID: <CAF__LHfk4Gix9PRNp_P_1TjRPAiF1Wg8o5tK8dgHERer9yu8kQ@mail.gmail.com>

think this is resolved - just puled the latest from git and
recompiled.  Seems to have fixed whatever was the issue - have not yet
looked at the list of commits, but it is working consistently now

On Mon, Aug 8, 2011 at 2:38 PM, Mike Ghicas <mghicas at gmail.com> wrote:
> I have a ?very odd one - what to see if anyone else has had a similar issue.
>
> Randomly after starting freeswitch, we get timeouts talking to nuance
> (different host via MRCP v2) ?- at first we thought it ?was an issue
> with nuance - however after digging a little bit we saw that FS did
> not even attempt to open a connection to the Nuance host (tcpdump
> shows noting at all...) and the connection times out.
>
> Have tried running with sqllite and mysql via odbc Have tied different
> combinations of FS/OS/vmware/etc No common pattern.
>
> When this fails I stop and restart FS and sometimes it works,
> sometimes it does not.
> Sometimes I have to restart 2-10 times, other times it will work right away.
>
> Has been working find on my dev host for a while, and I can not
> replicate the issue.
> Have installed and updated vmware tools, kernel, etc Had is sync to
> VMwarehost and not for time NTP is in use.
> Tried FS with -norc and -nocal options
> Nothing seems to make this behavior stop - or at least be consistent.
> Have reserved CPU cycles on Vmware
> Have run under strace - nothing interesting - just sits there sleeping...
>
> Spinning up a physical box for testing - but has anyone seen anything like this?
> Only seems to affect it when we are opening an MRCP connection
>
> Valgrind is next on the list.....
>
> Feels like a vmware/os/fs clocking/timing issue
>
> Thoughts?
>
>
> Some more details below including log - would love to hear if anyone
> has experiences anything like this..
>
>
> As i was writing this I ?tried the Physical host - same issue - so it
> does not looks like a timing issue.... RHEL 5.2
>
>
> --------
> We have FS running in a VM on Vmware Vsphere (4.1.0 320137, 4.1.0
> 208167, 4.1.0 260247) and on a physical machine.
> OS that has been tried RHEL 5.2, 5.5, 5.6
>
> Working: Vmware 4.1.0 260247 Guest 5.5
>
> Freeswitch - pulled from Git on 2011-07-27 Sofia and uni_mrcp enabled
>
>
> We have tried multiple combinations of everything - only similarity is VMware
>
> Our extension that we test with looks like this:
> <extension name="unimrcp">
> ? ? <condition field="destination_number" expression="5189">
> ? ? ? <action application="answer"/>
> ? ? ? <action application="set" data="tts_engine=unimrcp:nuance5-mrcp2-prod"/>
> ? ? ? <action application="set" data="tts_voice=tom"/>
> ? ? ? <action application="speak" data="The roof, the roof, the roof
> is on fire"/>
> ? ? ? <action application="sleep" data="100"/>
> ? ? </condition>
> ? </extension>
>
> This talks to Nuance Speech Server (via MRC v2) on another host.
>
>
> --
>
> Here is the log from the console with debug enabled:
>
>
> freeswitch at devipc02.nj3.ip-soft.net> tport_wakeup_pri(0x6f996f0): events IN
> tport_recv_event(0x6f996f0)
> tport_recv_iovec(0x6f996f0) msg 0x2aaac000c420 from
> (udp/10.140.20.182:5060) has 1171 bytes, veclen = 1
> tport_deliver(0x6f996f0): msg 0x2aaac000c420 (1171 bytes) from
> udp/192.168.8.11:5060/sip next=(nil)
> nta: received INVITE sip:5189 at 10.140.20.182:5060 SIP/2.0 (CSeq 101)
> nta: canonizing sip:5189 at 10.140.20.182:5060 with contact
> nta: INVITE (101) going to a default leg
> nta: timer set to 200 ms
> nua: nua_stack_process_request: entering
> nua: nh_create: entering
> nua: nh_create_handle: entering
> nua: nua_stack_set_params: entering
> soa_clone(static::0x6f80bd0, 0x6f67ec0, 0x2aaac000c110) called
> soa_set_params(static::0x2aaac000d300, ...) called
> nta_leg_tcreate(0x2aaac000da30)
> soa_init_offer_answer(static::0x2aaac000d300) called
> soa_set_remote_sdp(static::0x2aaac000d300, (nil), 0x2aaac000a0af, 212)
> called
> nua(0x2aaac000c110): adding session usage
> tport_tsend(0x6f996f0) tpn = UDP/192.168.8.11:5060 tport_resolve
> addrinfo = 192.168.8.11:5060
> tport_by_addrinfo(0x6f996f0): not found by name UDP/192.168.8.11:5060
> tport_vsend(0x6f996f0): 350 bytes of 350 to udp/192.168.8.11:5060
> tport_vsend returned 350
> nta: sent 100 Trying for INVITE (101)
> nua(0x2aaac000c110): event i_invite 100 Trying
> nua(0x2aaac000c110): call state changed: init -> received, received
> offer soa_get_remote_sdp(static::0x2aaac000d300, [0x400ea8c8],
> [0x400ea8c0], [(nil)]) called
> nua(0x2aaac000c110): event i_state 100 Trying
> nua: nua_application_event: entering
> 2011-08-08 18:13:36.135540 [NOTICE] switch_channel.c:812 New Channel
> sofia/internal/5502 at 192.168.8.11
> [148a57a7-c2af-45f2-8539-462b21f77a87]
> nua: nua_handle_bind: entering
> 2011-08-08 18:13:36.135540 [DEBUG] switch_core_state_machine.c:325
> (sofia/internal/5502 at 192.168.8.11) Running State Change CS_NEW
> 2011-08-08 18:13:36.135540 [DEBUG] switch_core_state_machine.c:343
> (sofia/internal/5502 at 192.168.8.11) State NEW
> nua: nua_handle_magic: entering
> nua: nua_application_event: entering
> 2011-08-08 18:13:36.141412 [DEBUG] sofia.c:4761 Channel
> sofia/internal/5502 at 192.168.8.11 entering state [received][100]
> 2011-08-08 18:13:36.141412 [DEBUG] sofia.c:4772 Remote SDP:
> v=0
> o=CiscoSystemsCCM-SIP 2000 1 IN IP4 192.168.8.11 s=SIP Call c=IN IP4
> 10.150.25.10
> t=0 0
> m=audio 16388 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
>
> 2011-08-08 18:13:36.141412 [DEBUG] sofia_glue.c:4651 Audio Codec
> Compare [PCMU:0:8000:20:64000]/[G7221:115:32000:20:48000]
> 2011-08-08 18:13:36.141412 [DEBUG] sofia_glue.c:4651 Audio Codec
> Compare [PCMU:0:8000:20:64000]/[G7221:107:16000:20:32000]
> 2011-08-08 18:13:36.141412 [DEBUG] sofia_glue.c:4651 Audio Codec
> Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000]
> 2011-08-08 18:13:36.141412 [DEBUG] sofia_glue.c:4651 Audio Codec
> Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
> 2011-08-08 18:13:36.141412 [DEBUG] sofia_glue.c:2774 Set Codec
> sofia/internal/5502 at 192.168.8.11 PCMU/8000 20 ms 160 samples 64000
> bits
> 2011-08-08 18:13:36.141412 [DEBUG] sofia_glue.c:4765 Set 2833 dtmf
> send/recv payload to 101
> 2011-08-08 18:13:36.141412 [DEBUG] sofia.c:4943
> (sofia/internal/5502 at 192.168.8.11) State Change CS_NEW -> CS_INIT
> 2011-08-08 18:13:36.141412 [DEBUG] switch_core_session.c:1116 Send
> signal sofia/internal/5502 at 192.168.8.11 [BREAK]
> nua: nua_handle_magic: entering
> 2011-08-08 18:13:36.141412 [DEBUG] switch_core_state_machine.c:325
> (sofia/internal/5502 at 192.168.8.11) Running State Change CS_INIT
> 2011-08-08 18:13:36.141412 [DEBUG] switch_core_state_machine.c:361
> (sofia/internal/5502 at 192.168.8.11) State INIT
> 2011-08-08 18:13:36.141412 [DEBUG] mod_sofia.c:84
> sofia/internal/5502 at 192.168.8.11 SOFIA INIT
> 2011-08-08 18:13:36.141412 [DEBUG] mod_sofia.c:124
> (sofia/internal/5502 at 192.168.8.11) State Change CS_INIT -> CS_ROUTING
> 2011-08-08 18:13:36.141412 [DEBUG] switch_core_session.c:1116 Send
> signal sofia/internal/5502 at 192.168.8.11 [BREAK]
> 2011-08-08 18:13:36.141412 [DEBUG] switch_core_state_machine.c:361
> (sofia/internal/5502 at 192.168.8.11) State INIT going to sleep
> 2011-08-08 18:13:36.141412 [DEBUG] switch_core_state_machine.c:325
> (sofia/internal/5502 at 192.168.8.11) Running State Change CS_ROUTING
> 2011-08-08 18:13:36.141412 [DEBUG] switch_channel.c:1668
> (sofia/internal/5502 at 192.168.8.11) Callstate Change DOWN -> RINGING
> 2011-08-08 18:13:36.141412 [DEBUG] switch_core_state_machine.c:364
> (sofia/internal/5502 at 192.168.8.11) State ROUTING
> 2011-08-08 18:13:36.141412 [DEBUG] mod_sofia.c:147
> sofia/internal/5502 at 192.168.8.11 SOFIA ROUTING
> 2011-08-08 18:13:36.141412 [DEBUG] switch_core_state_machine.c:77
> sofia/internal/5502 at 192.168.8.11 Standard ROUTING
> 2011-08-08 18:13:36.141412 [INFO] mod_dialplan_xml.c:331 Processing
> Mike Ghicas <5502>->5189 in context public
> Dialplan: sofia/internal/5502 at 192.168.8.11 parsing [public->unloop]
> continue=false
> Dialplan: sofia/internal/5502 at 192.168.8.11 Regex (PASS) [unloop]
> ${unroll_loops}(true) =~ /^true$/ break=on-false
> Dialplan: sofia/internal/5502 at 192.168.8.11 Regex (FAIL) [unloop]
> ${sip_looped_call}() =~ /^true$/ break=on-false
> Dialplan: sofia/internal/5502 at 192.168.8.11 parsing
> [public->outside_call] continue=true
> Dialplan: sofia/internal/5502 at 192.168.8.11 Absolute Condition [outside_call]
> Dialplan: sofia/internal/5502 at 192.168.8.11 Action set(outside_call=true)
> Dialplan: sofia/internal/5502 at 192.168.8.11 Action
> set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
> Dialplan: sofia/internal/5502 at 192.168.8.11 parsing
> [public->call_debug] continue=true
> Dialplan: sofia/internal/5502 at 192.168.8.11 Regex (FAIL) [call_debug]
> ${call_debug}(false) =~ /^true$/ break=never
> Dialplan: sofia/internal/5502 at 192.168.8.11 parsing
> [public->public_extensions] continue=false
> Dialplan: sofia/internal/5502 at 192.168.8.11 Regex (FAIL)
> [public_extensions] destination_number(5189) =~ /^(10[01][0-9])$/
> break=on-false
> Dialplan: sofia/internal/5502 at 192.168.8.11 parsing
> [public->public_did] continue=false
> Dialplan: sofia/internal/5502 at 192.168.8.11 Regex (FAIL) [public_did]
> destination_number(5189) =~ /^(5551212)$/ break=on-false
> Dialplan: sofia/internal/5502 at 192.168.8.11 parsing [public->unimrcp]
> continue=false
> Dialplan: sofia/internal/5502 at 192.168.8.11 Regex (PASS) [unimrcp]
> destination_number(5189) =~ /5189/ break=on-false
> Dialplan: sofia/internal/5502 at 192.168.8.11 Action answer()
> Dialplan: sofia/internal/5502 at 192.168.8.11 Action
> set(tts_engine=unimrcp:nuance5-mrcp2-prod)
> Dialplan: sofia/internal/5502 at 192.168.8.11 Action set(tts_voice=tom)
> Dialplan: sofia/internal/5502 at 192.168.8.11 Action speak(The roof, the
> roof, the roof is on fire. ?we don't need no water let that mother
> fucker burn.)
> Dialplan: sofia/internal/5502 at 192.168.8.11 Action sleep(100)
> 2011-08-08 18:13:36.141412 [DEBUG] switch_core_state_machine.c:119
> (sofia/internal/5502 at 192.168.8.11) State Change CS_ROUTING ->
> CS_EXECUTE
> 2011-08-08 18:13:36.141412 [DEBUG] switch_core_session.c:1116 Send
> signal sofia/internal/5502 at 192.168.8.11 [BREAK]
> 2011-08-08 18:13:36.141412 [DEBUG] switch_core_state_machine.c:364
> (sofia/internal/5502 at 192.168.8.11) State ROUTING going to sleep
> 2011-08-08 18:13:36.141412 [DEBUG] switch_core_state_machine.c:325
> (sofia/internal/5502 at 192.168.8.11) Running State Change CS_EXECUTE
> 2011-08-08 18:13:36.141412 [DEBUG] switch_core_state_machine.c:371
> (sofia/internal/5502 at 192.168.8.11) State EXECUTE
> 2011-08-08 18:13:36.141412 [DEBUG] mod_sofia.c:240
> sofia/internal/5502 at 192.168.8.11 SOFIA EXECUTE
> 2011-08-08 18:13:36.141412 [DEBUG] switch_core_state_machine.c:157
> sofia/internal/5502 at 192.168.8.11 Standard EXECUTE EXECUTE
> sofia/internal/5502 at 192.168.8.11 set(outside_call=true)
> 2011-08-08 18:13:36.141412 [DEBUG] mod_dptools.c:1060
> sofia/internal/5502 at 192.168.8.11 SET [outside_call]=[true] EXECUTE
> sofia/internal/5502 at 192.168.8.11 set(RFC2822_DATE=Mon, 08 Aug 2011
> 18:13:36 +0000)
> 2011-08-08 18:13:36.143981 [DEBUG] mod_dptools.c:1060
> sofia/internal/5502 at 192.168.8.11 SET [RFC2822_DATE]=[Mon, 08 Aug 2011
> 18:13:36 +0000] EXECUTE sofia/internal/5502 at 192.168.8.11 answer()
> 2011-08-08 18:13:36.152965 [DEBUG] sofia_glue.c:3015 AUDIO RTP
> [sofia/internal/5502 at 192.168.8.11] 10.140.20.182 port 16858 ->
> 10.150.25.10 port 16388 codec: 0 ms: 20
> 2011-08-08 18:13:36.152965 [DEBUG] switch_rtp.c:1623 Starting timer
> [soft] 160 bytes per 20ms
> 2011-08-08 18:13:36.154960 [DEBUG] sofia_glue.c:3277 Set 2833 dtmf
> send payload to 101
> 2011-08-08 18:13:36.154960 [DEBUG] sofia_glue.c:3282 Set 2833 dtmf
> receive payload to 101
> 2011-08-08 18:13:36.154960 [DEBUG] mod_sofia.c:681 Local SDP
> sofia/internal/5502 at 192.168.8.11:
> v=0
> o=FreeSWITCH 1312810358 1312810359 IN IP4 10.140.20.182 s=FreeSWITCH
> c=IN IP4 10.140.20.182
> t=0 0
> m=audio 16858 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
> nua: nua_respond: entering
> nua(0x2aaac000c110): sent signal r_respond
> 2011-08-08 18:13:36.154960 [DEBUG] switch_core_session.c:709 Send
> signal sofia/internal/5502 at 192.168.8.11 [BREAK]
> 2011-08-08 18:13:36.154960 [DEBUG] switch_channel.c:2830
> (sofia/internal/5502 at 192.168.8.11) Callstate Change RINGING -> ACTIVE
> 2011-08-08 18:13:36.154960 [NOTICE] mod_dptools.c:930 Channel
> [sofia/internal/5502 at 192.168.8.11] has been answered EXECUTE
> sofia/internal/5502 at 192.168.8.11
> set(tts_engine=unimrcp:nuance5-mrcp2-prod)
> 2011-08-08 18:13:36.156012 [DEBUG] mod_dptools.c:1060
> sofia/internal/5502 at 192.168.8.11 SET
> [tts_engine]=[unimrcp:nuance5-mrcp2-prod]
> EXECUTE sofia/internal/5502 at 192.168.8.11 set(tts_voice=tom)
> 2011-08-08 18:13:36.156012 [DEBUG] mod_dptools.c:1060
> sofia/internal/5502 at 192.168.8.11 SET [tts_voice]=[tom] EXECUTE
> sofia/internal/5502 at 192.168.8.11 speak(The roof, the roof, the roof is
> on fire. ?we don't need no water let that mother fucker burn.)
> 2011-08-08 18:13:36.156947 [INFO] mod_unimrcp.c:1567 speech_handle:
> name = unimrcp, rate = 8000, speed = 0, samples = 160, voice = ,
> engine = unimrcp, param = nuance5-mrcp2-prod
> 2011-08-08 18:13:36.156947 [INFO] mod_unimrcp.c:1570 voice = tom, rate = 8000
> 2011-08-08 18:13:36.156947 [DEBUG] mod_unimrcp.c:652 (TTS-1) audio queue created
> 2011-08-08 18:13:36.156947 [NOTICE] mrcp_client.c:549 Create MRCP
> Handle 0x703f1f0 [nuance5-mrcp2-prod]
> 2011-08-08 18:13:36.156947 [INFO] mrcp_client_session.c:142 Create
> Channel 0x703f1f0 <new>
> 2011-08-08 18:13:36.156947 [DEBUG] mrcp_client.c:1038 Signal
> Application Task Message
> nua(0x2aaac000c110): recv signal r_respond 200 OK
> nua: nua_stack_set_params: entering
> soa_set_params(static::0x2aaac000d300, ...) called
> soa_set_user_sdp(static::0x2aaac000d300, (nil), 0x2aaac0010a32, -1)
> called
> 2011-08-08 18:13:36.156947 [DEBUG] mrcp_client.c:1006 Receive
> Application Task Message [0]
> 2011-08-08 18:13:36.156947 [INFO] mrcp_client_session.c:398 Receive
> App Request 0x703f1f0 <new> [2]
> 2011-08-08 18:13:36.156947 [INFO] mrcp_client.c:901 Add MRCP Handle 0x703f1f0
> 2011-08-08 18:13:36.156947 [DEBUG] mrcp_client_session.c:1203 Dispatch
> Application Request 0x703f1f0 <new> [2]
> 2011-08-08 18:13:36.156947 [NOTICE] mrcp_client_session.c:718 Add
> Control Channel 0x703f1f0 <new at speechsynth>
> soa_set_capability_sdp(static::0x2aaac000d300, (nil), 0x2aaac0010a32,
> -1) called
> 2011-08-08 18:13:36.156947 [DEBUG] mrcp_client_session.c:762 Add RTP
> Termination 0x703f1f0 <new>
> 2011-08-08 18:13:36.156947 [DEBUG] apt_consumer_task.c:90 Wait for
> Task Messages [MRCP Client]
> nua: nua_invite_server_respond: entering
> soa_generate_answer(static::0x2aaac000d300) called
> soa_static_offer_answer_action(0x2aaac000d300, soa_generate_answer):
> called soa_static(0x2aaac000d300, soa_generate_answer): generating
> local description soa_static(0x2aaac000d300, soa_generate_answer):
> upgrade with remote description soa_sdp_mode_set(0x400eacd0,
> 0x2aaac000e990, ""): called soa_static(0x2aaac000d300,
> soa_generate_answer): storing local description
> soa_activate(static::0x2aaac000d300, (nil)) called
> soa_get_local_sdp(static::0x2aaac000d300, [(nil)], [0x400eae38],
> [0x400eae44]) called
> tport_tsend(0x6f996f0) tpn = UDP/192.168.8.11:5060 tport_resolve
> addrinfo = 192.168.8.11:5060
> tport_by_addrinfo(0x6f996f0): not found by name UDP/192.168.8.11:5060
> tport_vsend(0x6f996f0): 1175 bytes of 1175 to udp/192.168.8.11:5060
> tport_vsend returned 1175
> nta: sent 200 OK for INVITE (101)
> nua(0x2aaac000c110): call state changed: received -> completed, sent
> answer soa_get_local_sdp(static::0x2aaac000d300, [0x400eaef8],
> [0x400eaef0], [(nil)]) called soa_get_params(static::0x2aaac000d300,
> ...) called
> nua(0x2aaac000c110): event i_state 200 OK
> 2011-08-08 18:13:36.156947 [DEBUG] mrcp_client_connection.c:603
> Process Control Message
> 2011-08-08 18:13:36.156947 [DEBUG] mrcp_client.c:1104 Signal
> Connection Task Message
> 2011-08-08 18:13:36.156947 [DEBUG] mrcp_client.c:974 Receive
> Connection Task Message [0]
> nua: nua_application_event: entering
> 2011-08-08 18:13:36.156947 [DEBUG] mrcp_client_session.c:311 On
> Control Channel Add 0x703f1f0 <new>
> 2011-08-08 18:13:36.156947 [DEBUG] apt_consumer_task.c:90 Wait for
> Task Messages [MRCP Client]
> 2011-08-08 18:13:36.156947 [DEBUG] sofia.c:4761 Channel
> sofia/internal/5502 at 192.168.8.11 entering state [completed][200]
> nua: nua_handle_magic: entering
> 2011-08-08 18:13:36.159000 [DEBUG] mpf_engine.c:302 Process MPF Message
> 2011-08-08 18:13:36.159000 [DEBUG] mpf_context.c:172 Add Context
> 2011-08-08 18:13:36.159000 [DEBUG] mpf_context.c:176 Add Termination
> 2011-08-08 18:13:36.159000 [DEBUG] mpf_context.c:176 Add Termination
> 2011-08-08 18:13:36.159000 [DEBUG] mrcp_client.c:999 Receive Media Task Message
> 2011-08-08 18:13:36.159000 [DEBUG] mrcp_client_session.c:1039 On
> Termination Add 0x703f1f0 <new>
> 2011-08-08 18:13:36.159000 [DEBUG] mrcp_client_session.c:1039 On
> Termination Add 0x703f1f0 <new>
> 2011-08-08 18:13:36.159000 [INFO] mrcp_client_session.c:420 Send Offer
> 0x703f1f0 <new> [c:1 a:1 v:0]
> 2011-08-08 18:13:36.159000 [INFO] mrcp_sofiasip_client_agent.c:316
> Local SDP 0x703f1f0 <new>
> v=0
> o=FreeSWITCH 0 0 IN IP4 10.140.20.182
> s=-
> c=IN IP4 10.140.20.182
> t=0 0
> m=application 9 TCP/MRCPv2 1
> a=setup:active
> a=connection:new
> a=resource:speechsynth
> a=cmid:1
> m=audio 4002 RTP/AVP 0 8 96
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:96 L16/8000
> a=recvonly
> a=mid:1
>
> 2011-08-08 18:13:36.159000 [DEBUG] apt_consumer_task.c:90 Wait for
> Task Messages [MRCP Client]
> tport_wakeup_pri(0x6f996f0): events IN
> tport_recv_event(0x6f996f0)
> tport_recv_iovec(0x6f996f0) msg 0x2aaac000fb00 from
> (udp/10.140.20.182:5060) has 431 bytes, veclen = 1
> tport_deliver(0x6f996f0): msg 0x2aaac000fb00 (431 bytes) from
> udp/192.168.8.11:5060/sip next=(nil)
> nta: received ACK sip:5189 at 10.140.20.182:5060;transport=udp SIP/2.0 (CSeq 101)
> nta: ACK (101) is going to INVITE (101)
> nua: process_ack_or_cancel: entering
> soa_clear_remote_sdp(static::0x2aaac000d300) called
> nua(0x2aaac000c110): event i_ack 200 OK
> nua(0x2aaac000c110): call state changed: completed -> ready
> nua(0x2aaac000c110): event i_state 200 OK
> nua(0x2aaac000c110): event i_active 200 Call active
> nua(): refresh session after 1768 seconds (in [1768..1768])
> nua: nua_application_event: entering
> nua: nua_handle_magic: entering
> nua: nua_application_event: entering
> 2011-08-08 18:13:36.161958 [DEBUG] sofia.c:4761 Channel
> sofia/internal/5502 at 192.168.8.11 entering state [ready][200]
> nua: nua_handle_magic: entering
> nua: nua_application_event: entering
> nua: nua_handle_magic: entering
> nta: timer set next to 4827 ms
> 2011-08-08 18:13:41.158959 [WARNING] mod_unimrcp.c:1010 (TTS-1) MRCP
> session has not opened after 5000 ms
> nta: timer I fired, terminate 200 response incoming_reclaim_all((nil),
> (nil), 0x400ead40)
> nta_incoming_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free
> nta: timer not set
>


From stephen_wilkey at hotmail.com  Tue Aug  9 08:12:11 2011
From: stephen_wilkey at hotmail.com (steve2010)
Date: Mon, 8 Aug 2011 21:12:11 -0700 (PDT)
Subject: [Freeswitch-users] intermittent EXCHANGE_ROUTING_ERROR
In-Reply-To: <1312857284679-6666783.post@n2.nabble.com>
References: <1QRrkF-0007cj-IK@mail.aastral.net>
	<BANLkTikc5bMokDY6__miju=rOUmsLc7kNg@mail.gmail.com>
	<1QRt7M-0002Zm-GE@mail.aastral.net>
	<BANLkTikXJrxuwUXAd2kYFCiDQOLGQgw9dg@mail.gmail.com>
	<1QS7yM-0005cW-8h@mail.aastral.net>
	<BANLkTi=hv5KZhMUfj1TKbucARH8gzADqpw@mail.gmail.com>
	<1QSAQM-0003GJ-6U@mail.aastral.net>
	<BANLkTink0R3qq96hQez786NJhFM_yAzrhg@mail.gmail.com>
	<1QTeTP-0003ff-Om@mail.aastral.net>
	<1312857284679-6666783.post@n2.nabble.com>
Message-ID: <BLU152-W6296972BEEE195CBE309C4E4200@phx.gbl>



 Yes, please post.



Date: Mon, 8 Aug 2011 19:34:44 -0700
From: ml-node+6666783-1527079717-350115 at n2.nabble.com
To: stephen_wilkey at hotmail.com
Subject: Re: intermittent EXCHANGE_ROUTING_ERROR

Thanks for the discussion on this issue. 

I have discovered that my ISP is sending calls to me with Max-Forwards=4 which means that I can't do much with the call when it arrives.  Is there a way to ignore or override the Max-Forwards setting that is provided with the call when it arrives so that I am able to process it? 

Regards, 
Stephen 



Bill W.-2-3 wrote:
Hey Anthony/Michael, 

After a lot of digging, it turns out the problem was the Max-Forwards on 
the incoming SIP traffic was too low for FreeSWITCH to complete the call. 

Thanks so much for all your help! 





If you reply to this email, your message will be added to the discussion below:http://freeswitch-users.2379917.n2.nabble.com/intermittent-EXCHANGE-ROUTING-ERROR-tp6428615p6666783.html 
To unsubscribe from intermittent EXCHANGE_ROUTING_ERROR, click here.  		 	   		  

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/intermittent-EXCHANGE-ROUTING-ERROR-tp6428615p6666911.html
Sent from the freeswitch-users mailing list archive at Nabble.com.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110808/57dae4d1/attachment.html 

From nsirugudi at gmail.com  Tue Aug  9 14:50:06 2011
From: nsirugudi at gmail.com (Narendra Sirugudi)
Date: Tue, 9 Aug 2011 16:20:06 +0530
Subject: [Freeswitch-users] rfc 5168 (picture fast update ) support in
	freeswitch.
Message-ID: <CAP18h9qN7v_KrerHdW+6re7j90-rQb62O7dXQ-b769W3P1B+ZQ@mail.gmail.com>

Hi,

I wanted to know whether freeswitch/sofia stack has support for rfc 5168
(picture fast update) ?

It looks like there is something related picture fast update in mod_sofia.
Is there any way to initiate a "picture fast update" SIP INFO request in
freeswitch ? Maybe through some application.

thanks,
--naren
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110809/a91e4c8f/attachment.html 

From steveayre at gmail.com  Tue Aug  9 14:51:46 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 9 Aug 2011 11:51:46 +0100
Subject: [Freeswitch-users] intermittent EXCHANGE_ROUTING_ERROR
In-Reply-To: <BLU152-W4470860E41B53A27022CDDE4210@phx.gbl>
References: <BLU152-W45140868DE9EDDE504AE68E4210@phx.gbl>
	<BLU152-W4470860E41B53A27022CDDE4210@phx.gbl>
Message-ID: <CAFiqYuke3z_dD6L=uRthfXGcGMXXdJ3t-MzZdSZ7+PFQtksd6Q@mail.gmail.com>

You can for an outgoing call, but you'll want to be very careful doing so,
as you may no longer be able to detect routing loops.

http://wiki.freeswitch.org/wiki/Variable_max_forwards
<action application="bridge"
data="{max_forwards=65}sofia/gateway/ipexchange/442920180123"/>

-Steve


On 9 August 2011 00:44, Stephen Wilkey <stephen_wilkey at hotmail.com> wrote:

>
>  Thanks for the discussion on this issue.
>
> I have discovered that my ISP is sending calls to me with *Max*-*Forwards*=4
> which means that I can't do much with the call when it arrives.  Is there a
> way to ignore or override the *Max*-*Forwards* setting that is provided
> with the call when it arrives so that I am able to process it?
>
> Regards,
> Stephen
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110809/03308bef/attachment.html 

From max.asterisk at gmail.com  Tue Aug  9 14:39:36 2011
From: max.asterisk at gmail.com (Max Alex)
Date: Tue, 9 Aug 2011 16:09:36 +0530
Subject: [Freeswitch-users] Look n8nrcuv
Message-ID: <CAJ7RAxCfqoFYFO=HeF5_g-zwrZ5qjDXmmJxgP5mN0RwcEui-Hg@mail.gmail.com>

2qhealspczpb.
 http://aqabpsychology.co.uk/wp-content/plugins/wp-filemanager/incl/img/yptm.html
cevvqi eerhlt5es17 57ty43lzfqn, q5kbmgldfmln tavkldcqk8. ziy19sq30
9l2c8hm i9ueccmem3.

-- 
Thanks,
Max Alex
Voip Developer


From nagalenoj at gmail.com  Tue Aug  9 16:01:25 2011
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Tue, 9 Aug 2011 17:31:25 +0530
Subject: [Freeswitch-users] Core dump - When using user_recurse_variables in
	bridge
Message-ID: <CAA1Qw2vfhhDEMYV7L-S8AZqDh5MfKEEYfcmNXB21pARA_wfZ5Q@mail.gmail.com>

Hi Friends,
      I've got segfault in yesterday's git, when I execute the below from
nc,
sendmsg
call-command: execute
execute-app-name: bridge
execute-app-arg:
{user_recurse_variables=false,group_confirm_cancel_timeout=true,group_confirm_key=exec,group_confirm_file='perl
/FMS-FS/bin/bridge_confirm.pl'}[leg_timeout=10]user/1001

It works fine when I bridge without any of those variables.
execute-app-arg: [leg_timeout=10]user/1001

Here is the backtrace, http://pastebin.freeswitch.org/17004.
-- 
Regards,
Nagalenoj H.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110809/747abe7b/attachment.html 

From fieldpeak at gmail.com  Tue Aug  9 16:58:22 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Tue, 9 Aug 2011 20:58:22 +0800
Subject: [Freeswitch-users] Mod_rad_auth issue for FS working with
 FreeRadius server
In-Reply-To: <CAJoKKdWqjt46Gvbj=4Tz-apQb+EV2UHJTsRBTE_bJZ95ogo2DA@mail.gmail.com>
References: <CAOTmbMzvd5OHL8z=_8vu6Orp2_prGxaw1gmXYCM2=2+pgzs_WA@mail.gmail.com>
	<CAJoKKdX01Qc8ibs25PUELi8GrzMDNOrHb_bQvYdq3ZqA4XLy4w@mail.gmail.com>
	<CAOTmbMxL3pLOuw655fyT5VzAON_N0PrDN+-i-++=oZoYU6+Fmg@mail.gmail.com>
	<CAOTmbMyvyDA-CRpaMC=WRGVkiyTG=Rs+n4hyeQWdyTZpqysSzw@mail.gmail.com>
	<CAJoKKdXSwF02S9tBkorGwOjj57AtkRpU0c177qXASEpodKVk=w@mail.gmail.com>
	<CAOTmbMzu2qDYh2T24JSpCYn1xH2J4iGiK7UkUB4Kq=-KdcDWGQ@mail.gmail.com>
	<CAJoKKdU66jK3f+mOEV90Y3zrmzpTSdWnwGsWimZxir42khiJ-w@mail.gmail.com>
	<CAOTmbMxW=EDr3-Ayo2R_MsCCTUZ4PnDkkRUuR2FisH9tK5xkWA@mail.gmail.com>
	<CAJoKKdVSkQOV3Kou_ro5AQZTyaqH6r0WRE-hZdrA=ye8ARE_vQ@mail.gmail.com>
	<CAOTmbMyg5_q-pFFtpzOooUcSZ8Gkb38y4sk1pYeJLYqguA=Jbg@mail.gmail.com>
	<CAJoKKdWqjt46Gvbj=4Tz-apQb+EV2UHJTsRBTE_bJZ95ogo2DA@mail.gmail.com>
Message-ID: <CAOTmbMx8_aDoB2Q9ocuVtuXoQf0hbTJ0=VL+RtaWiZQgAw=YTg@mail.gmail.com>

Hi Tihomir,

As my understanding, when using mod_rad_auth, we have to send both username
and password to FreeRadius, like the example in wiki below (marked in
yellow), the example is for a fixed password, however in real world, we have
to dynamically inject the password as per user on-the-fly, e.g. user 1001 's
password is 1234, user 1002's password is 2345 etc. in other word, we have
to dynamically get the specific user's password and inject to the dial plan.
Can you please advise how we should write the dial plan for the real case?
Thanks in avdvance.

P.S. What I'm concerning are both REGISTERATON and INVITE...how can we do
the auth by Freeradius...

<extension name="unitest_rad-ANI-balance">
    <condition field="destination_number" expression="^602$">
      <action application="log" data="INFO  PRIJE RAD_AUTH "/>

      <action inline="true" application="set"
data="CALLID=h323-conf-id=${uuid}"/>
      <action inline="true" application="set"
data="SERVICENUM=h323-prompt-id=${destination_number}"/>
      <action inline="true" application="set"
data="TRANSACTIONID=h323-ivr-out=transactionID:1234"/>
 <!--      <action inline="true" application="set"
data="CALLINGNUMBER=${caller_id_number}"/> -->
      <action inline="true" application="set" data="CALLINGNUMBER=38516060333"/>
      <action inline="true" application="set" data="USERNAME=38516060333"/>
 <!--      <action inline="true" application="set" data="USERNAME=209354"/> -->
      <action inline="true" application="set" data="PASSWD=003282"/>
      <action inline="true" application="set" data="DIALED_NUMBER=16094191500"/>



Regards,
Charles

2011/8/9 Tihomir Culjaga <tculjaga at gmail.com>

> im glad it works :=)
>
> T.
>
>
> On Mon, Aug 8, 2011 at 8:18 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>
>> Hi Tihomir,
>>
>> The issue has been resolved by correcting the client secrect, appreciated
>> very much for your kindly help!
>>
>> Regards,
>> Charles
>>
>> 2011/8/7 Tihomir Culjaga <tculjaga at gmail.com>
>>
>>> are u sure you are using the correct secret on both client and server ?
>>>
>>>
>>> On Fri, Aug 5, 2011 at 10:12 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>>>
>>>> Hi Tihomir,
>>>>
>>>> Thanks for your advise, i've added below to rad_auth.conf.xml (vsas
>>>> section), as well as tried auth-type to 0(local) and 1(system), however, the
>>>> issue still exist.
>>>>
>>>>
>>>> <param name="NAS-Port-Type" id="61" value="0" pec="0" expr="0"
>>>> direction="in"/>
>>>> <param name="Login-User" id="6" value="1" pec="0" expr="0"
>>>> direction="in"/>
>>>> <param name="Auth-Type" id="1000" value="0" pec="0" expr="0"
>>>> direction="in"/>
>>>>
>>>> FreeRadius output:
>>>>
>>>> Found Auth-Type = PAP
>>>> # Executing group from file /usr/local/etc/raddb/sites-enabled/default
>>>> +- entering group PAP {...}
>>>> [pap] login attempt with password "Q?????? ??????p???F?+??a"
>>>> [pap] Using clear text password "1111"
>>>> [pap] Passwords don't match
>>>> ++[pap] returns reject
>>>> Failed to authenticate the user.
>>>>   WARNING: Unprintable characters in the password.        Double-check the shared secret on the server and the NAS!
>>>>
>>>> Using Post-Auth-Type Reject
>>>> # Executing group from file /usr/local/etc/raddb/sites-enabled/default
>>>> +- entering group REJECT {...}
>>>> [attr_filter.access_reject]     expand: %{User-Name} -> 1001
>>>>  attr_filter: Matched entry DEFAULT at line 11
>>>> ++[attr_filter.access_reject] returns updated
>>>> Delaying reject of request 38 for 1 seconds
>>>>
>>>> Regards,
>>>> Charles
>>>>
>>>>
>>>> 2011/8/5 Tihomir Culjaga <tculjaga at gmail.com>
>>>>
>>>>> add to rad_auth.conf.xml
>>>>>
>>>>> <param name="NAS-Port-Type" id="61" value="0" pec="0" expr="0"
>>>>> direction="in"/>
>>>>> <param name="Login-User" id="6" value="1" pec="0" expr="0"
>>>>> direction="in"/>
>>>>>
>>>>>
>>>>>
>>>>> as for Auth Type im not sure if you need it ... this is up to your
>>>>> server.
>>>>> According to dictionary file you need to set it as follows:
>>>>>
>>>>> <param name="Auth-Type" id="1000" value="?" pec="0" expr="0"
>>>>> direction="in"/>
>>>>>
>>>>> the value (set as ?) is one of the folowing. Again, not sure what is
>>>>> required by your server.
>>>>>
>>>>> VALUE           Auth-Type               Local                   0
>>>>> VALUE           Auth-Type               System                  1
>>>>> VALUE           Auth-Type               SecurID                 2
>>>>> VALUE           Auth-Type               Crypt-Local             3
>>>>> VALUE           Auth-Type               Reject                  4
>>>>>
>>>>> #
>>>>> #       Cistron extensions
>>>>> #
>>>>> VALUE           Auth-Type               Pam                     253
>>>>> VALUE           Auth-Type               Accept                  254
>>>>>
>>>>>
>>>>>
>>>>> regards,
>>>>> Tihomir.
>>>>>
>>>>>
>>>>>
>>>>> On Wed, Aug 3, 2011 at 6:32 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>>>>>
>>>>>> Hi Tihomir,
>>>>>>
>>>>>> Sorry, i missed your mail in gmail before, just now saw it, and after
>>>>>> using your dictionary.all, the dictionary issue was resolved, very
>>>>>> appreciated for your kindly help! however, it did not fully functional yet,
>>>>>>
>>>>>> Attached are configuration files that i used, when i dial 601 to
>>>>>> trigger to auth, the freeradius server shows log below, the supecious log is
>>>>>> the value User-Password, it should be '1111' that i've set in the mysql db
>>>>>> of freeradisu server for the user 1001 .
>>>>>>
>>>>>> i searched in google, for "known good" password issue, i suggest
>>>>>> change user-password to cleartext-password, however, i did not find where it
>>>>>> is.
>>>>>> and also the Auth-Type, where to configure it...
>>>>>>
>>>>>> Freeradius server log:
>>>>>>
>>>>>> rad_recv: Access-Request packet from host 127.0.0.1 port 52684, id=49,
>>>>>> length=111
>>>>>>         User-Name = "1001"
>>>>>>         User-Password = "?\210\365@\263\t\306\343\243iT?\311C\t\002"
>>>>>>         Called-Station-Id = "888"
>>>>>>         h323-conf-id = "749d2b5a-16ad-48e4-af58-24011949d1b5"
>>>>>>         Calling-Station-Id = "1001"
>>>>>>         NAS-Port = 0
>>>>>>         NAS-IP-Address = 127.0.0.1
>>>>>> # Executing section authorize from file
>>>>>> /usr/local/etc/raddb/sites-enabled/default
>>>>>> +- entering group authorize {...}
>>>>>> ++[preprocess] returns ok
>>>>>> [auth_log]      expand:
>>>>>> /usr/local/var/log/radius/radacct/%{Client-IP-Address}/auth-detail-%Y%m%d ->
>>>>>> /usr/local/var/log/radius/radacct/127.0.0.1/auth-detail-20110803
>>>>>> [auth_log]
>>>>>> /usr/local/var/log/radius/radacct/%{Client-IP-Address}/auth-detail-%Y%m%d
>>>>>> expands to /usr/local/var/log/radius/radacct/
>>>>>> 127.0.0.1/auth-detail-20110803
>>>>>> [auth_log]      expand: %t -> Wed Aug  3 12:06:33 2011
>>>>>> ++[auth_log] returns ok
>>>>>> ++[chap] returns noop
>>>>>> ++[mschap] returns noop
>>>>>> ++[digest] returns noop
>>>>>> [suffix] No '@' in User-Name = "1001", looking up realm NULL
>>>>>> [suffix] No such realm "NULL"
>>>>>> ++[suffix] returns noop
>>>>>> [eap] No EAP-Message, not doing EAP
>>>>>> ++[eap] returns noop
>>>>>> ++[unix] returns notfound
>>>>>> ++[files] returns noop
>>>>>> [sql]   expand: %{User-Name} -> 1001
>>>>>> [sql] sql_set_user escaped user --> '1001'
>>>>>> rlm_sql (sql): Reserving sql socket id: 4
>>>>>> [sql]   expand: SELECT id, username, attribute, value, op
>>>>>> FROM radcheck           WHERE username = '%{SQL-User-Name}'           ORDER
>>>>>> BY id -> SELECT id, username, attribute, value, op           FROM
>>>>>> radcheck           WHERE username = '1001'           ORDER BY id
>>>>>> [sql]   expand: SELECT groupname           FROM radusergroup
>>>>>> WHERE username = '%{SQL-User-Name}'           ORDER BY priority -> SELECT
>>>>>> groupname           FROM radusergroup           WHERE username =
>>>>>> '1001'           ORDER BY priority
>>>>>> rlm_sql (sql): Released sql socket id: 4
>>>>>> [sql] User 1001 not found
>>>>>> ++[sql] returns notfound
>>>>>> ++[expiration] returns noop
>>>>>> ++[logintime] returns noop
>>>>>> [pap] WARNING! No "known good" password found for the user.
>>>>>> Authentication may fail because of this.
>>>>>> ++[pap] returns noop
>>>>>> ERROR: No authenticate method (Auth-Type) found for the request:
>>>>>> Rejecting the user
>>>>>> Failed to authenticate the user.
>>>>>>   WARNING: Unprintable characters in the password.        Double-check
>>>>>> the shared secret on the server and the NAS!
>>>>>> Using Post-Auth-Type Reject
>>>>>> # Executing group from file /usr/local/etc/raddb/sites-enabled/default
>>>>>> +- entering group REJECT {...}
>>>>>> [attr_filter.access_reject]     expand: %{User-Name} -> 1001
>>>>>>  attr_filter: Matched entry DEFAULT at line 11
>>>>>> ++[attr_filter.access_reject] returns updated
>>>>>> Delaying reject of request 8 for 1 seconds
>>>>>> Going to the next request
>>>>>> Waking up in 0.9 seconds.
>>>>>> Sending delayed reject for request 8
>>>>>> Sending Access-Reject of id 49 to 127.0.0.1 port 52684
>>>>>> Waking up in 4.9 seconds.
>>>>>> Cleaning up request 8 ID 49 with timestamp +7674
>>>>>> Ready to process requests.
>>>>>> WARNING! No "known good" password found for the user
>>>>>>
>>>>>> Regards,
>>>>>> Charles
>>>>>>
>>>>>>
>>>>>> 2011/8/3 Tihomir Culjaga <tculjaga at gmail.com>
>>>>>>
>>>>>>> did u use the dictionary i have attached ?
>>>>>>>
>>>>>>>
>>>>>>> On Tue, Aug 2, 2011 at 10:08 AM, fieldpeak <fieldpeak at gmail.com>wrote:
>>>>>>>
>>>>>>>> i tried change to 'h323-conf-id' to 'h323-call-origin' in
>>>>>>>> 02_unitest_rad-ANI-auth.xml, rad_auth.conf.xml, however, it still prompt
>>>>>>>> '[ERR] mod_rad_auth.c:428 Unknown attribute: key:h323-conf-id, not
>>>>>>>> found in dictionary', so where the mod_rad_auth read out the 'h323-conf-id'?
>>>>>>>> very very strange, which dictionary it was using...
>>>>>>>>
>>>>>>>> Regards,
>>>>>>>> Charles
>>>>>>>>
>>>>>>>>
>>>>>>>> 2011/8/2 fieldpeak <fieldpeak at gmail.com>
>>>>>>>>
>>>>>>>>> Hi Tihomir,
>>>>>>>>>
>>>>>>>>> Finally the answer coming, i see the hope, thanks for your reply,
>>>>>>>>> :)
>>>>>>>>>
>>>>>>>>> As your advise, i only use one attribute(h323-conf-id) in my
>>>>>>>>> dialplan, and only one attribute(h323-conf-id) in rad_auth.conf.xml, and
>>>>>>>>> using the attached dictionary (from ciso) which contains this attribute,
>>>>>>>>> however, it still prompt 'unknown attribute', so i suspected if it was
>>>>>>>>> reading /usr/local/etc/radiusclient/dictionary, so i copy the same
>>>>>>>>> dictionary to /usr/local/freeswitch/radius/, it did not any help at all...
>>>>>>>>> very strange...
>>>>>>>>>
>>>>>>>>> Log:
>>>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>> default_realm := .
>>>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>> radius_timeout := 3.
>>>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>> radius_retries := 2.
>>>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>> radius_deadtime := 0.
>>>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set bindaddr
>>>>>>>>> := *.
>>>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:371 ... radius:
>>>>>>>>> User-Name: 38516060333
>>>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:380 ... radius:
>>>>>>>>> User-Password: 003282
>>>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:396 ... radius:
>>>>>>>>> Called-station-Id: 16094191500
>>>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:413 Handle
>>>>>>>>> attribute: h323-conf-id
>>>>>>>>> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:428 Unknown
>>>>>>>>> attribute: key:h323-conf-id, not found in dictionary
>>>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:538 abort sending
>>>>>>>>> radius packet.
>>>>>>>>> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:546 An error
>>>>>>>>> occured during RADIUS Authentication(RC=-1)
>>>>>>>>> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:702 An error
>>>>>>>>> occured during radius authorization.
>>>>>>>>>
>>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO
>>>>>>>>> AUTH_RESULT=)
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>   <extension name="unitest_rad-ANI-auth">
>>>>>>>>>     <condition field="destination_number" expression="^601$">
>>>>>>>>>       <!-- <action application="log" data="INFO  Before Auth "/>
>>>>>>>>> -->
>>>>>>>>>
>>>>>>>>>       <action inline="true" application="set" data="CALLID=
>>>>>>>>> h323-conf-id=${uuid}"/>
>>>>>>>>>
>>>>>>>>>       <action inline="true" application="set"
>>>>>>>>> data="USERNAME=1001"/>
>>>>>>>>>       <action inline="true" application="set" data="PASSWD=1111"/>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>       <action application="sleep" data="2000"/>
>>>>>>>>>       <action application="auth_function" data="in
>>>>>>>>> ${DIALED_NUMBER}, in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>
>>>>>>>>>
>>>>>>>>>     </condition>
>>>>>>>>>   </extension>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> <configuration name="rad_auth.conf" description="radius
>>>>>>>>> authentification module">
>>>>>>>>>   <settings>
>>>>>>>>>
>>>>>>>>>   </settings>
>>>>>>>>>
>>>>>>>>>   <client>
>>>>>>>>>     <param name="authserver" value="127.0.0.1:1812:gateway"/>
>>>>>>>>>     <param name="dictionary"
>>>>>>>>> value="/usr/local/etc/radiusclient/dictionary"/>
>>>>>>>>>     <param name="seqfile" value="/var/run/radius.seq"/>
>>>>>>>>>     <param name="mapfile"
>>>>>>>>> value="/usr/local/etc/radiusclient/port-id-map"/>
>>>>>>>>>     <param name="default_realm" value=""/>
>>>>>>>>>     <param name="radius_timeout" value="3"/>
>>>>>>>>>     <param name="radius_retries" value="2"/>
>>>>>>>>>     <param name="radius_deadtime" value="0"/>
>>>>>>>>>     <param name="bindaddr" value="*"/>
>>>>>>>>>   </client>
>>>>>>>>>
>>>>>>>>>   <vsas>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>     <param name="h323-conf-id" id="24" value="CALLID" pec="9"
>>>>>>>>> expr="1" direction="in"/>
>>>>>>>>>
>>>>>>>>>   </vsas>
>>>>>>>>>  </configuration>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> 2011/8/2 Tihomir Culjaga <tculjaga at gmail.com>
>>>>>>>>>
>>>>>>>>>> hi,
>>>>>>>>>>
>>>>>>>>>> dictionary.all is just the name of a file containing all
>>>>>>>>>> attributes i needed at that time.
>>>>>>>>>>
>>>>>>>>>> you can include other dictionaries by putting #INCLUDE <pathname>
>>>>>>>>>> at the end of the dictionary file you reference in rad_auth.conf.xml.
>>>>>>>>>> if the INCLUDE doesn't work, just append dictionary.cisco to your
>>>>>>>>>> dictionary file... and make your own file.
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> check inline comments down below...
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> T.
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> On Sun, Jul 31, 2011 at 10:46 AM, fieldpeak <fieldpeak at gmail.com>wrote:
>>>>>>>>>>
>>>>>>>>>>> Hello Gurus,
>>>>>>>>>>>
>>>>>>>>>>> i met a issue when using
>>>>>>>>>>> mod_rad_auth(http://wiki.freeswitch.org/wiki/Mod_rad_auth) to
>>>>>>>>>>> works
>>>>>>>>>>> with freeradius server+mysql for AAA, the details is below, Could
>>>>>>>>>>> anyone give any hints, Thanks in advance.
>>>>>>>>>>>
>>>>>>>>>>> i setup a dial plan "unitest_rad-ANI-auth" as wiki above,
>>>>>>>>>>> however,
>>>>>>>>>>> when i dialed 601 to trigger the dial plan, the console show
>>>>>>>>>>> errors,
>>>>>>>>>>> it looks "h323-conf-id" is not in the directory, then i tried to
>>>>>>>>>>> add
>>>>>>>>>>> this attribute to the dictionary, however, it does not help, in
>>>>>>>>>>> the
>>>>>>>>>>> wiki, it mentioned the rad_auth.conf.xml contains <param
>>>>>>>>>>> name="dictionary"
>>>>>>>>>>> value="/usr/local/etc/radiusclient/dictionary.all"/>, however i
>>>>>>>>>>> did
>>>>>>>>>>> not find the file "dictionary.all" at that directory, so i use
>>>>>>>>>>> dictionary. BTW, the freeradius server + mysql works well.
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> i just appended the information needed into dictionary.all file...
>>>>>>>>>> (vendor and attribute definition).
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> console errors:
>>>>>>>>>>>
>>>>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 auth_function(in ,
>>>>>>>>>>> in
>>>>>>>>>>> 38516060333, in 003282, out AUTH_RESULT)
>>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:301 allocate
>>>>>>>>>>> initial
>>>>>>>>>>> structure.
>>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:313 initialzed
>>>>>>>>>>> configuration.
>>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>>>> authserver
>>>>>>>>>>> := 127.0.0.1:1812:gateway.
>>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>>>> dictionary
>>>>>>>>>>> := /usr/local/etc/radiusclient/dictionary.
>>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set seqfile
>>>>>>>>>>> :=
>>>>>>>>>>> /var/run/radius.seq.
>>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set mapfile
>>>>>>>>>>> :=
>>>>>>>>>>> /usr/local/etc/radiusclient/port-id-map.
>>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>>>> default_realm := .
>>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>>>> radius_timeout := 3.
>>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>>>> radius_retries := 2.
>>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>>>> radius_deadtime := 0.
>>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>>>> bindaddr := *.
>>>>>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:371 ... radius:
>>>>>>>>>>> User-Name: 38516060333
>>>>>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:380 ... radius:
>>>>>>>>>>> User-Password: 003282
>>>>>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:391 ... radius:
>>>>>>>>>>> Called-station-Id is empty, ignoring...
>>>>>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:413 Handle
>>>>>>>>>>> attribute: h323-conf-id
>>>>>>>>>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:428 Unknown
>>>>>>>>>>> attribute:
>>>>>>>>>>> key:h323-conf-id, not found in dictionary
>>>>>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:538 abort
>>>>>>>>>>> sending
>>>>>>>>>>> radius packet.
>>>>>>>>>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:546 An error
>>>>>>>>>>> occured
>>>>>>>>>>> during RADIUS Authentication(RC=-1)
>>>>>>>>>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:702 An error
>>>>>>>>>>> occured
>>>>>>>>>>> during radius authorization.
>>>>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO
>>>>>>>>>>>  AUTH_RESULT=)
>>>>>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202
>>>>>>>>>>>  AUTH_RESULT=
>>>>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO
>>>>>>>>>>>  billing_model=)
>>>>>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202
>>>>>>>>>>>  billing_model=
>>>>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO
>>>>>>>>>>>  credit_amount=)
>>>>>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202
>>>>>>>>>>>  credit_amount=
>>>>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO  currency=)
>>>>>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  currency=
>>>>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO
>>>>>>>>>>>  preffered_lang=)
>>>>>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202
>>>>>>>>>>>  preffered_lang=
>>>>>>>>>>>
>>>>>>>>>>> added below in the
>>>>>>>>>>> dictionary(/usr/local/etc/radiusclient/dictionary):
>>>>>>>>>>>
>>>>>>>>>>> ATTRIBUTE       h323-conf-id            1008    string
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> you need the vendor definition as well
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> dial plan:
>>>>>>>>>>> <extension name="unitest_rad-ANI-auth">
>>>>>>>>>>>    <condition field="destination_number" expression="^601$">
>>>>>>>>>>>      <action application="log" data="INFO  Before Auth "/>
>>>>>>>>>>>
>>>>>>>>>>>      <action inline="true" application="set"
>>>>>>>>>>> data="CALLID=h323-conf-id=${uuid}"/>
>>>>>>>>>>>      <action inline="true" application="set"
>>>>>>>>>>> data="SERVICENUM=h323-prompt-id=${destination_number}"/>
>>>>>>>>>>>      <action inline="true" application="set"
>>>>>>>>>>> data="TRANSACTIONID=h323-ivr-out=transactionID:1234"/>
>>>>>>>>>>>  <!--      <action inline="true" application="set"
>>>>>>>>>>> data="CALLINGNUMBER=${caller_id_number}"/> -->
>>>>>>>>>>>      <action inline="true" application="set"
>>>>>>>>>>> data="CALLINGNUMBER=38516060333"/>
>>>>>>>>>>>      <action inline="true" application="set"
>>>>>>>>>>> data="USERNAME=38516060333"/>
>>>>>>>>>>>  <!--      <action inline="true" application="set"
>>>>>>>>>>> data="USERNAME=209354"/> -->
>>>>>>>>>>>      <action inline="true" application="set"
>>>>>>>>>>> data="PASSWD=003282"/>
>>>>>>>>>>>  <!--      <action inline="true" application="set"
>>>>>>>>>>> data="DIALED_NUMBER=16094191500"/>  -->
>>>>>>>>>>>
>>>>>>>>>>>      <action application="sleep" data="2000"/>
>>>>>>>>>>>      <action application="auth_function" data="in
>>>>>>>>>>> ${DIALED_NUMBER},
>>>>>>>>>>> in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>>>>  AUTH_RESULT=${AUTH_RESULT}"/>
>>>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>>>>  billing_model=${billing_model}"/>
>>>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>>>>  credit_amount=${credit_amount}"/>
>>>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>>>>  currency=${currency}"/>
>>>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>>>>  preffered_lang=${preffered_lang}"/>
>>>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>>>>  credit_time=${credit_time}"/>
>>>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>>>> h323_ivr_duration=${h323_ivr_duration}"/>
>>>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>>>>  return_code=${return_code}"/>
>>>>>>>>>>>      <!-- <action application="execute_extension" data="AUTH XML
>>>>>>>>>>> default"/> -->
>>>>>>>>>>>    </condition>
>>>>>>>>>>>  </extension>
>>>>>>>>>>>
>>>>>>>>>>>  radius_cdr.conf.xml:
>>>>>>>>>>>  <configuration name="radius_cdr.conf" description="RADIUS CDR
>>>>>>>>>>> Configuration">
>>>>>>>>>>>
>>>>>>>>>>>        <settings>
>>>>>>>>>>>
>>>>>>>>>>>                <!-- location of the radius dictionary files -->
>>>>>>>>>>>
>>>>>>>>>>>                <param name="dictionary"
>>>>>>>>>>> value="/usr/local/freeswitch/conf/radius/dictionary"/>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>> your dictionary file need to contain all the attributes you are
>>>>>>>>>> trying to use or to include other dictionaries (In this case
>>>>>>>>>> dictionary.cisco) from the dictionary file you are referencing here.
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>>                <!-- number of retries for each server -->
>>>>>>>>>>>
>>>>>>>>>>>                <param name="radius_retries" value="3"/>
>>>>>>>>>>>
>>>>>>>>>>>                <!-- number of seconds to wait between retries -->
>>>>>>>>>>>
>>>>>>>>>>>                <param name="radius_timeout" value="5"/>
>>>>>>>>>>>
>>>>>>>>>>>                <!-- accounting servers, up to 8 allowed -->
>>>>>>>>>>>
>>>>>>>>>>>                <!-- value is "host:port:secret", port is optional
>>>>>>>>>>> -->
>>>>>>>>>>>
>>>>>>>>>>>                <!-- use IP ADDRESSES, not hostnames -->
>>>>>>>>>>>
>>>>>>>>>>>                <param name="acct_server" value="127.0.0.1:1813
>>>>>>>>>>> :testing123"/>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>        </settings>
>>>>>>>>>>>
>>>>>>>>>>> </configuration>
>>>>>>>>>>>
>>>>>>>>>>>  the FS version:
>>>>>>>>>>>  FreeSWITCH Version 1.0.head (git-492bc6b 2011-07-23 12-53-04
>>>>>>>>>>> -0400)
>>>>>>>>>>>
>>>>>>>>>>>  Regards,
>>>>>>>>>>>  Charles
>>>>>>>>>>>
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>>
>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>
>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110809/2055cc1f/attachment-0001.html 

From Hector.Geraldino at ip-soft.net  Tue Aug  9 17:12:50 2011
From: Hector.Geraldino at ip-soft.net (Hector Geraldino)
Date: Tue, 9 Aug 2011 09:12:50 -0400
Subject: [Freeswitch-users] how to have fs interrupt a file or tts
	if	digit is pressed
In-Reply-To: <16927.1312850183@ccs.covici.com>
References: <23058.1312823337@ccs.covici.com>
	<6A6B4C284AD15042B429EB9D904544AD021FD8A4A6@NY1-EXMB-01.ip-soft.net>
	<16927.1312850183@ccs.covici.com>
Message-ID: <6A6B4C284AD15042B429EB9D904544AD021FD8A505@NY1-EXMB-01.ip-soft.net>

Well, I'm not a perl developer so I don't have an accurate response for your question. My guess is that you can call:

$session->call_command("break");

But I would recommend you to take a look at the wiki page http://wiki.freeswitch.org/wiki/Mod_perl to see some examples and get a better idea of how to do it. 


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of covici at ccs.covici.com
Sent: Monday, August 08, 2011 8:36 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] how to have fs interrupt a file or tts if digit is pressed

OK, how do I send a break command?

Hector Geraldino <Hector.Geraldino at ip-soft.net> wrote:

> You can send a "break" command in the DTMF callback event. This will stop the audio playback or TTS.
> 
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of covici at ccs.covici.com
> Sent: Monday, August 08, 2011 1:09 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] how to have fs interrupt a file or tts if digit is pressed
> 
> Hi.  I am using Perl, if that makes any difference and I would like to
> arrange things in such a way that during a prompt, if the person presses
> a key, then the prompt will stop speaking and I can see what the key is
> and do something.  Here is an excerpt of the script I am using.
> 
> $session->setInputCallback('got_press',""); #listen for key presses in
> the background
> 
> 
> while($session->ready())
> {
> $session->streamFile("test_break.wav");
> 
>     if ($press_so_far != "")
>     {
>         $session->say($press_so_far,"EN", "NAME_SPELLED", "ITERATED");
>         $press_so_far="";
>             }
> }
>  
> Instead of streamFile, I tried speaking text through tts_command line,
> and a phrase macro which did the same.  I was sure that at least
> streamFile would break, but it did not.
> 
> Thanks in advance for any ideas.
> 
> -- 
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
> 
>          John Covici
>          covici at ccs.covici.com
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com

_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

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


From anthony.minessale at gmail.com  Tue Aug  9 17:59:35 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 9 Aug 2011 08:59:35 -0500
Subject: [Freeswitch-users] Core dump - When using
 user_recurse_variables in bridge
In-Reply-To: <CAA1Qw2vfhhDEMYV7L-S8AZqDh5MfKEEYfcmNXB21pARA_wfZ5Q@mail.gmail.com>
References: <CAA1Qw2vfhhDEMYV7L-S8AZqDh5MfKEEYfcmNXB21pARA_wfZ5Q@mail.gmail.com>
Message-ID: <CAKbxfG_0KA2UB_=Fz=3yaxKGWgo7y0DWCkwdgEZ_bXYSJ1-E9A@mail.gmail.com>

fixed

On Tue, Aug 9, 2011 at 7:01 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:
> Hi Friends,
> ????? I've got segfault in yesterday's git, when I execute the below from
> nc,
> sendmsg
> call-command: execute
> execute-app-name: bridge
> execute-app-arg:
> {user_recurse_variables=false,group_confirm_cancel_timeout=true,group_confirm_key=exec,group_confirm_file='perl
> /FMS-FS/bin/bridge_confirm.pl'}[leg_timeout=10]user/1001
> It works fine when I bridge without any of those variables.
> execute-app-arg: [leg_timeout=10]user/1001
>
> Here is the backtrace, http://pastebin.freeswitch.org/17004.
> --
> Regards,
> Nagalenoj H.
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From covici at ccs.covici.com  Tue Aug  9 18:44:46 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Tue, 09 Aug 2011 10:44:46 -0400
Subject: [Freeswitch-users] how to have fs interrupt a file or tts if
	digit is pressed
In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD021FD8A505@NY1-EXMB-01.ip-soft.net>
References: <23058.1312823337@ccs.covici.com>
	<6A6B4C284AD15042B429EB9D904544AD021FD8A4A6@NY1-EXMB-01.ip-soft.net>
	<16927.1312850183@ccs.covici.com>
	<6A6B4C284AD15042B429EB9D904544AD021FD8A505@NY1-EXMB-01.ip-soft.net>
Message-ID: <14507.1312901086@ccs.covici.com>

Well, I can try that -- I had never used call_command and it seems not
to be available in the .pm.

I did a search and found sock.pl was using such a thing, so this
requires further investigation.

Thanks for the tip.

Hector Geraldino <Hector.Geraldino at ip-soft.net> wrote:

> Well, I'm not a perl developer so I don't have an accurate response for your question. My guess is that you can call:
> 
> $session->call_command("break");
> 
> But I would recommend you to take a look at the wiki page http://wiki.freeswitch.org/wiki/Mod_perl to see some examples and get a better idea of how to do it. 
> 
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of covici at ccs.covici.com
> Sent: Monday, August 08, 2011 8:36 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] how to have fs interrupt a file or tts if digit is pressed
> 
> OK, how do I send a break command?
> 
> Hector Geraldino <Hector.Geraldino at ip-soft.net> wrote:
> 
> > You can send a "break" command in the DTMF callback event. This will stop the audio playback or TTS.
> > 
> > 
> > -----Original Message-----
> > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of covici at ccs.covici.com
> > Sent: Monday, August 08, 2011 1:09 PM
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: [Freeswitch-users] how to have fs interrupt a file or tts if digit is pressed
> > 
> > Hi.  I am using Perl, if that makes any difference and I would like to
> > arrange things in such a way that during a prompt, if the person presses
> > a key, then the prompt will stop speaking and I can see what the key is
> > and do something.  Here is an excerpt of the script I am using.
> > 
> > $session->setInputCallback('got_press',""); #listen for key presses in
> > the background
> > 
> > 
> > while($session->ready())
> > {
> > $session->streamFile("test_break.wav");
> > 
> >     if ($press_so_far != "")
> >     {
> >         $session->say($press_so_far,"EN", "NAME_SPELLED", "ITERATED");
> >         $press_so_far="";
> >             }
> > }
> >  
> > Instead of streamFile, I tried speaking text through tts_command line,
> > and a phrase macro which did the same.  I was sure that at least
> > streamFile would break, but it did not.
> > 
> > Thanks in advance for any ideas.
> > 
> > -- 
> > Your life is like a penny.  You're going to lose it.  The question is:
> > How do
> > you spend it?
> > 
> >          John Covici
> >          covici at ccs.covici.com
> > 
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > 
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> -- 
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
> 
>          John Covici
>          covici at ccs.covici.com
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From nagalenoj at gmail.com  Tue Aug  9 19:07:36 2011
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Tue, 9 Aug 2011 20:37:36 +0530
Subject: [Freeswitch-users] DTMF issue when using execute_extension with
	play_and_get_digits
Message-ID: <CAA1Qw2sLHtu8ZHQn7GJFOs=f2k9H2pLa+mgu6Kt+FEtGcwWBOw@mail.gmail.com>

Hi Friends,
     Facing an issue when using bind_meta_app and execute_extension(with
play_and_get_digits) combined.

Here is my dialplan,
<condition field="destination_number" expression="^4567$" break="on-true">
<action application="play_and_get_digits" data="1 5 3 7000 #
'/voice_files/A.wav' '/voice_files/failed.wav' dtmf_received \d+"/>
</condition>
<condition field="destination_number" expression="^.*$" break="on-true">
<action application="answer"/>
<action application="bind_meta_app" data="5 b o execute_extension::4567 XML
default"/>
<action application="bridge" data="freetdm/1/a/9952248266"/>
<action application="park"/>
</condition>

So, when callee enters *5,  I want the caller to enter a number. I get the
extension executed as expected. The caller is able to hear the voice file
played and when he enters the digits, it is not received. Digits are not
even present in FS log.

In the normal cases, there is no issues in getting DTMFs. I don't know, what
am I doing wrong here. Kindly, help me to resolve this.

-- 
Regards,
Nagalenoj H.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110809/74e4c3cf/attachment.html 

From Hector.Geraldino at ip-soft.net  Tue Aug  9 19:47:37 2011
From: Hector.Geraldino at ip-soft.net (Hector Geraldino)
Date: Tue, 9 Aug 2011 11:47:37 -0400
Subject: [Freeswitch-users] how to have fs interrupt a file or tts
	if	digit is pressed
In-Reply-To: <14507.1312901086@ccs.covici.com>
References: <23058.1312823337@ccs.covici.com>
	<6A6B4C284AD15042B429EB9D904544AD021FD8A4A6@NY1-EXMB-01.ip-soft.net>
	<16927.1312850183@ccs.covici.com>
	<6A6B4C284AD15042B429EB9D904544AD021FD8A505@NY1-EXMB-01.ip-soft.net>
	<14507.1312901086@ccs.covici.com>
Message-ID: <6A6B4C284AD15042B429EB9D904544AD021FD8A551@NY1-EXMB-01.ip-soft.net>

There's a session->execute("command") in the perl API as I can see on the wiki. 

You can try to replace the 'call_command' method name for 'execute' [$session->execute("hangup")] and see if this does the trick.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of covici at ccs.covici.com
Sent: Tuesday, August 09, 2011 10:45 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] how to have fs interrupt a file or tts if digit is pressed

Well, I can try that -- I had never used call_command and it seems not
to be available in the .pm.

I did a search and found sock.pl was using such a thing, so this
requires further investigation.

Thanks for the tip.

Hector Geraldino <Hector.Geraldino at ip-soft.net> wrote:

> Well, I'm not a perl developer so I don't have an accurate response for your question. My guess is that you can call:
> 
> $session->call_command("break");
> 
> But I would recommend you to take a look at the wiki page http://wiki.freeswitch.org/wiki/Mod_perl to see some examples and get a better idea of how to do it. 
> 
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of covici at ccs.covici.com
> Sent: Monday, August 08, 2011 8:36 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] how to have fs interrupt a file or tts if digit is pressed
> 
> OK, how do I send a break command?
> 
> Hector Geraldino <Hector.Geraldino at ip-soft.net> wrote:
> 
> > You can send a "break" command in the DTMF callback event. This will stop the audio playback or TTS.
> > 
> > 
> > -----Original Message-----
> > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of covici at ccs.covici.com
> > Sent: Monday, August 08, 2011 1:09 PM
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: [Freeswitch-users] how to have fs interrupt a file or tts if digit is pressed
> > 
> > Hi.  I am using Perl, if that makes any difference and I would like to
> > arrange things in such a way that during a prompt, if the person presses
> > a key, then the prompt will stop speaking and I can see what the key is
> > and do something.  Here is an excerpt of the script I am using.
> > 
> > $session->setInputCallback('got_press',""); #listen for key presses in
> > the background
> > 
> > 
> > while($session->ready())
> > {
> > $session->streamFile("test_break.wav");
> > 
> >     if ($press_so_far != "")
> >     {
> >         $session->say($press_so_far,"EN", "NAME_SPELLED", "ITERATED");
> >         $press_so_far="";
> >             }
> > }
> >  
> > Instead of streamFile, I tried speaking text through tts_command line,
> > and a phrase macro which did the same.  I was sure that at least
> > streamFile would break, but it did not.
> > 
> > Thanks in advance for any ideas.
> > 
> > -- 
> > Your life is like a penny.  You're going to lose it.  The question is:
> > How do
> > you spend it?
> > 
> >          John Covici
> >          covici at ccs.covici.com
> > 
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > 
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> -- 
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
> 
>          John Covici
>          covici at ccs.covici.com
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com

_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

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


From covici at ccs.covici.com  Tue Aug  9 20:10:16 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Tue, 09 Aug 2011 12:10:16 -0400
Subject: [Freeswitch-users] how to have fs interrupt a file or tts if
	digit is pressed
In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD021FD8A551@NY1-EXMB-01.ip-soft.net>
References: <23058.1312823337@ccs.covici.com>
	<6A6B4C284AD15042B429EB9D904544AD021FD8A4A6@NY1-EXMB-01.ip-soft.net>
	<16927.1312850183@ccs.covici.com>
	<6A6B4C284AD15042B429EB9D904544AD021FD8A505@NY1-EXMB-01.ip-soft.net>
	<14507.1312901086@ccs.covici.com>
	<6A6B4C284AD15042B429EB9D904544AD021FD8A551@NY1-EXMB-01.ip-soft.net>
Message-ID: <28538.1312906216@ccs.covici.com>

I don't want to hang up, I just want the file to stop and process the
dtmf.  Also, I saw something in streamfile where I could process the
dtmf while the file was still playing -- maybe I could do rewind and
fast forward, if there are calls for that purpose.

Thanks for your suggestions.

Hector Geraldino <Hector.Geraldino at ip-soft.net> wrote:

> There's a session->execute("command") in the perl API as I can see on the wiki. 
> 
> You can try to replace the 'call_command' method name for 'execute' [$session->execute("hangup")] and see if this does the trick.
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of covici at ccs.covici.com
> Sent: Tuesday, August 09, 2011 10:45 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] how to have fs interrupt a file or tts if digit is pressed
> 
> Well, I can try that -- I had never used call_command and it seems not
> to be available in the .pm.
> 
> I did a search and found sock.pl was using such a thing, so this
> requires further investigation.
> 
> Thanks for the tip.
> 
> Hector Geraldino <Hector.Geraldino at ip-soft.net> wrote:
> 
> > Well, I'm not a perl developer so I don't have an accurate response for your question. My guess is that you can call:
> > 
> > $session->call_command("break");
> > 
> > But I would recommend you to take a look at the wiki page http://wiki.freeswitch.org/wiki/Mod_perl to see some examples and get a better idea of how to do it. 
> > 
> > 
> > -----Original Message-----
> > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of covici at ccs.covici.com
> > Sent: Monday, August 08, 2011 8:36 PM
> > To: FreeSWITCH Users Help
> > Subject: Re: [Freeswitch-users] how to have fs interrupt a file or tts if digit is pressed
> > 
> > OK, how do I send a break command?
> > 
> > Hector Geraldino <Hector.Geraldino at ip-soft.net> wrote:
> > 
> > > You can send a "break" command in the DTMF callback event. This will stop the audio playback or TTS.
> > > 
> > > 
> > > -----Original Message-----
> > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of covici at ccs.covici.com
> > > Sent: Monday, August 08, 2011 1:09 PM
> > > To: freeswitch-users at lists.freeswitch.org
> > > Subject: [Freeswitch-users] how to have fs interrupt a file or tts if digit is pressed
> > > 
> > > Hi.  I am using Perl, if that makes any difference and I would like to
> > > arrange things in such a way that during a prompt, if the person presses
> > > a key, then the prompt will stop speaking and I can see what the key is
> > > and do something.  Here is an excerpt of the script I am using.
> > > 
> > > $session->setInputCallback('got_press',""); #listen for key presses in
> > > the background
> > > 
> > > 
> > > while($session->ready())
> > > {
> > > $session->streamFile("test_break.wav");
> > > 
> > >     if ($press_so_far != "")
> > >     {
> > >         $session->say($press_so_far,"EN", "NAME_SPELLED", "ITERATED");
> > >         $press_so_far="";
> > >             }
> > > }
> > >  
> > > Instead of streamFile, I tried speaking text through tts_command line,
> > > and a phrase macro which did the same.  I was sure that at least
> > > streamFile would break, but it did not.
> > > 
> > > Thanks in advance for any ideas.
> > > 
> > > -- 
> > > Your life is like a penny.  You're going to lose it.  The question is:
> > > How do
> > > you spend it?
> > > 
> > >          John Covici
> > >          covici at ccs.covici.com
> > > 
> > > _______________________________________________
> > > Join us at ClueCon 2011, Aug 9-11, Chicago
> > > http://www.cluecon.com 877-7-4ACLUE
> > > 
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > > 
> > > _______________________________________________
> > > Join us at ClueCon 2011, Aug 9-11, Chicago
> > > http://www.cluecon.com 877-7-4ACLUE
> > > 
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > 
> > -- 
> > Your life is like a penny.  You're going to lose it.  The question is:
> > How do
> > you spend it?
> > 
> >          John Covici
> >          covici at ccs.covici.com
> > 
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > 
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> -- 
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
> 
>          John Covici
>          covici at ccs.covici.com
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From covici at ccs.covici.com  Tue Aug  9 20:30:14 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Tue, 09 Aug 2011 12:30:14 -0400
Subject: [Freeswitch-users] how to have fs interrupt a file or tts if
	digit is pressed
In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD021FD8A551@NY1-EXMB-01.ip-soft.net>
References: <23058.1312823337@ccs.covici.com>
	<6A6B4C284AD15042B429EB9D904544AD021FD8A4A6@NY1-EXMB-01.ip-soft.net>
	<16927.1312850183@ccs.covici.com>
	<6A6B4C284AD15042B429EB9D904544AD021FD8A505@NY1-EXMB-01.ip-soft.net>
	<14507.1312901086@ccs.covici.com>
	<6A6B4C284AD15042B429EB9D904544AD021FD8A551@NY1-EXMB-01.ip-soft.net>
Message-ID: <31329.1312907414@ccs.covici.com>

I have found that returning 0 from my inputcallback will do the break --
its not perfect, but about as good as I could get.  Now how to do rewind
and forward during a stream file as some of them are long.

Hector Geraldino <Hector.Geraldino at ip-soft.net> wrote:

> There's a session->execute("command") in the perl API as I can see on the wiki. 
> 
> You can try to replace the 'call_command' method name for 'execute' [$session->execute("hangup")] and see if this does the trick.
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of covici at ccs.covici.com
> Sent: Tuesday, August 09, 2011 10:45 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] how to have fs interrupt a file or tts if digit is pressed
> 
> Well, I can try that -- I had never used call_command and it seems not
> to be available in the .pm.
> 
> I did a search and found sock.pl was using such a thing, so this
> requires further investigation.
> 
> Thanks for the tip.
> 
> Hector Geraldino <Hector.Geraldino at ip-soft.net> wrote:
> 
> > Well, I'm not a perl developer so I don't have an accurate response for your question. My guess is that you can call:
> > 
> > $session->call_command("break");
> > 
> > But I would recommend you to take a look at the wiki page http://wiki.freeswitch.org/wiki/Mod_perl to see some examples and get a better idea of how to do it. 
> > 
> > 
> > -----Original Message-----
> > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of covici at ccs.covici.com
> > Sent: Monday, August 08, 2011 8:36 PM
> > To: FreeSWITCH Users Help
> > Subject: Re: [Freeswitch-users] how to have fs interrupt a file or tts if digit is pressed
> > 
> > OK, how do I send a break command?
> > 
> > Hector Geraldino <Hector.Geraldino at ip-soft.net> wrote:
> > 
> > > You can send a "break" command in the DTMF callback event. This will stop the audio playback or TTS.
> > > 
> > > 
> > > -----Original Message-----
> > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of covici at ccs.covici.com
> > > Sent: Monday, August 08, 2011 1:09 PM
> > > To: freeswitch-users at lists.freeswitch.org
> > > Subject: [Freeswitch-users] how to have fs interrupt a file or tts if digit is pressed
> > > 
> > > Hi.  I am using Perl, if that makes any difference and I would like to
> > > arrange things in such a way that during a prompt, if the person presses
> > > a key, then the prompt will stop speaking and I can see what the key is
> > > and do something.  Here is an excerpt of the script I am using.
> > > 
> > > $session->setInputCallback('got_press',""); #listen for key presses in
> > > the background
> > > 
> > > 
> > > while($session->ready())
> > > {
> > > $session->streamFile("test_break.wav");
> > > 
> > >     if ($press_so_far != "")
> > >     {
> > >         $session->say($press_so_far,"EN", "NAME_SPELLED", "ITERATED");
> > >         $press_so_far="";
> > >             }
> > > }
> > >  
> > > Instead of streamFile, I tried speaking text through tts_command line,
> > > and a phrase macro which did the same.  I was sure that at least
> > > streamFile would break, but it did not.
> > > 
> > > Thanks in advance for any ideas.
> > > 
> > > -- 
> > > Your life is like a penny.  You're going to lose it.  The question is:
> > > How do
> > > you spend it?
> > > 
> > >          John Covici
> > >          covici at ccs.covici.com
> > > 
> > > _______________________________________________
> > > Join us at ClueCon 2011, Aug 9-11, Chicago
> > > http://www.cluecon.com 877-7-4ACLUE
> > > 
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > > 
> > > _______________________________________________
> > > Join us at ClueCon 2011, Aug 9-11, Chicago
> > > http://www.cluecon.com 877-7-4ACLUE
> > > 
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > 
> > -- 
> > Your life is like a penny.  You're going to lose it.  The question is:
> > How do
> > you spend it?
> > 
> >          John Covici
> >          covici at ccs.covici.com
> > 
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > 
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> -- 
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
> 
>          John Covici
>          covici at ccs.covici.com
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From nicevoip at googlemail.com  Tue Aug  9 21:39:14 2011
From: nicevoip at googlemail.com (Nice Voip)
Date: Tue, 9 Aug 2011 19:39:14 +0200
Subject: [Freeswitch-users] Problem with XML CDR
In-Reply-To: <CAKzWOxWJe85gThLNsEjPNo2cVgoxxv_wkjWgOeNWxZagd4SjSQ@mail.gmail.com>
References: <CA+7EGbmk6WhgC8tCfFtSqkrQDosd4j87x7x1Bei0J+Km=HWHpw@mail.gmail.com>
	<CALNZuwnK9g83RoGF4CnM0Ou8qAabYPB-nWQ7_uAuYL_Cr=1zLw@mail.gmail.com>
	<CA+7EGb=nj_XKFkPMijtD_MSj9XurFG0THDZVq+mr+AExcqrEaA@mail.gmail.com>
	<CAKzWOxWJe85gThLNsEjPNo2cVgoxxv_wkjWgOeNWxZagd4SjSQ@mail.gmail.com>
Message-ID: <CA+7EGbnZD+uGrtEC+QH4gmH8zGS83jjLQoFOCEY2SvzFPRBSxw@mail.gmail.com>

Hi Michael,

Yes correct, the carrier sent basically German characters for example when
they send "B?sch" then it shows in XML as strange character, but when i open
xml cdr in notepad++ then it looks ok

2ndly originate_early_media has nothing to do with special characters, and
FS sometimes writes it has orginate_*uarly*_media

br

On Sat, Aug 6, 2011 at 12:30 AM, Michael Collins <msc at freeswitch.org> wrote:

> Can you open the XML CDR in a hex editor and see if there are goofy
> characters? I wonder if there's an encoding issue with the info sent over
> from the carrier.
>
> -MC
>
>
> On Thu, Aug 4, 2011 at 3:03 AM, Nice Voip <nicevoip at googlemail.com> wrote:
>
>> Hmm i've latest version too, but there is not much traffic, and also on
>> this two month older version its rarely happen, in any case i'll try to move
>> to latest version and will try to reproduce.... but when it will be
>> reproduced my version will not be latest anymore :)
>>
>>
>>
>>
>> On Thu, Aug 4, 2011 at 11:56 AM, Avi Marcus <avi at avimarcus.net> wrote:
>>
>>> The usual response is.. can you update from your nearly two month old
>>> version and reproduce the same issue?
>>> -Avi
>>>
>>>
>>> On Thu, Aug 4, 2011 at 12:48 PM, Nice Voip <nicevoip at googlemail.com>wrote:
>>>
>>>> Dear All,
>>>>
>>>> This problem is very hard to reproduce and i really don't know when it
>>>> would happen, i don't have sip log or other traces but only the CDR file and
>>>> its looks like this:
>>>>
>>>>
>>>> <caller_id_name>TELESAT TRAJKOVI?
>>>>       <ani>+3xxxxxxx</ani>          ? MILE</caller_id_name>
>>>>       <aniii></aniii>
>>>>
>>>> and sometimes
>>>>
>>>> <oritinate_early_media> is writted <originate_uarly_media> (insead of
>>>> early its "uarly")
>>>>
>>>> i noted down this issue on: FreeSWITCH Version 1.0.head (git-1d3417a
>>>> 2011-06-07 17-35-49 -0400)
>>>>
>>>> Thanks.
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110809/6132250a/attachment.html 

From matt at hellohunter.com  Tue Aug  9 21:54:17 2011
From: matt at hellohunter.com (Matt Hunter)
Date: Tue, 9 Aug 2011 10:54:17 -0700
Subject: [Freeswitch-users] 400 Bad Record-Route Header
Message-ID: <CAA5vsdMjDTu_04r4TXH2bVWG1Te2Za7dQz1Ns4JgUO-L0xK-8w@mail.gmail.com>

FreeSWITCH 1.0.6 is complaining about a 400 Bad Record-Route Header. The
header is

Record-Route:
<sip:209.216.2.211;lr=on;ftag=SDgbvq702-3521894321-133800%uukeghgcjiudgieijkcuu0u>

is the %uukeghgcjiudgieijkcuu0u the problem?

Thanks.

--matt
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110809/09488868/attachment.html 

From mi.ke at null.net  Tue Aug  9 22:55:34 2011
From: mi.ke at null.net (Mi Ke)
Date: Tue, 09 Aug 2011 18:55:34 +0000
Subject: [Freeswitch-users] playing value with mod_say & getting DTMF input
	at the same time
Message-ID: <20110809185534.167940@gmx.com>

Hi All,

 Using mod_say and read function at once - is that possible with Lua ? I need to play a numeric value (like credit amount or time) to Leg A and catch user's input at the same time. As a temporary solution, I wrote a function translating numeric values into a string "prompt1.wav&prompt2.wav....etc" and then suppliy is to the read function. Is there anyway I could use say for that ?

 Thanks / Mike
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110809/818ac7d2/attachment.html 

From avi at avimarcus.net  Tue Aug  9 22:59:57 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 9 Aug 2011 21:59:57 +0300
Subject: [Freeswitch-users] playing value with mod_say & getting DTMF
 input at the same time
In-Reply-To: <20110809185534.167940@gmx.com>
References: <20110809185534.167940@gmx.com>
Message-ID: <CALNZuwkP7hd6v1fGFkqjHkaLg4V4ncgTX8_uoCdJKs5pVon1kw@mail.gmail.com>

Create a phrase macro for that and call it for your play_and_get_digits

-Avi


On Tue, Aug 9, 2011 at 9:55 PM, Mi Ke <mi.ke at null.net> wrote:

> Hi All,
>
> Using mod_say and read function at once - is that possible with Lua ? I
> need to play a numeric value (like credit amount or time) to Leg A and catch
> user's input at the same time. As a temporary solution, I wrote a function
> translating numeric values into a string "prompt1.wav&prompt2.wav....etc"
> and then suppliy is to the read function. Is there anyway I could use say
> for that ?
>
> Thanks / Mike
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110809/c5a82c57/attachment-0001.html 

From freeswitch-list at puzzled.xs4all.nl  Tue Aug  9 23:07:56 2011
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Tue, 09 Aug 2011 21:07:56 +0200
Subject: [Freeswitch-users] 400 Bad Record-Route Header
In-Reply-To: <CAA5vsdMjDTu_04r4TXH2bVWG1Te2Za7dQz1Ns4JgUO-L0xK-8w@mail.gmail.com>
References: <CAA5vsdMjDTu_04r4TXH2bVWG1Te2Za7dQz1Ns4JgUO-L0xK-8w@mail.gmail.com>
Message-ID: <4E41858C.5050104@puzzled.xs4all.nl>

On 08/09/2011 07:54 PM, Matt Hunter wrote:
> FreeSWITCH 1.0.6 is complaining about a 400 Bad Record-Route Header. The
> header is
>
> Record-Route:
> <sip:209.216.2.211;lr=on;ftag=SDgbvq702-3521894321-133800%uukeghgcjiudgieijkcuu0u>
>
> is the %uukeghgcjiudgieijkcuu0u the problem?

I'd say version 1.0.6 is the problem. It's ancient and compared to 
recent releases a Pretty Bad Idea. Please get latest git and try again. 
If there still is a problem with latest git then file a bug.

Regards,
Patrick


From mi.ke at null.net  Tue Aug  9 23:54:09 2011
From: mi.ke at null.net (Mi Ke)
Date: Tue, 09 Aug 2011 19:54:09 +0000
Subject: [Freeswitch-users] playing value with mod_say & getting DTMF
 input at the same time
Message-ID: <20110809195409.167940@gmx.com>

can I then use phrase:macro_name as a part of the played prompt in play_and_get_digits i.e. "prompt1.wav&phrase:macro_name&prompt2.wav ... etc" ? will such construction really work ?



----- Original Message -----
From: Avi Marcus
Sent: 08/09/11 09:59 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] playing value with mod_say & getting DTMF input at the same time

 Create a phrase macro for that and call it for your play_and_get_digits 

-Avi

 On Tue, Aug 9, 2011 at 9:55 PM, Mi Ke < mi.ke at null.net > wrote:
Hi All,

 Using mod_say and read function at once - is that possible with Lua ? I need to play a numeric value (like credit amount or time) to Leg A and catch user's input at the same time. As a temporary solution, I wrote a function translating numeric values into a string "prompt1.wav&prompt2.wav....etc" and then suppliy is to the read function. Is there anyway I could use say for that ?

 Thanks / Mike

 _______________________________________________
 Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com  877-7-4ACLUE

 FreeSWITCH-users mailing list
 FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110809/7c14f52f/attachment.html 

From nestor at tiendalinux.com  Tue Aug  9 20:48:00 2011
From: nestor at tiendalinux.com (Nestor A Diaz)
Date: Tue, 09 Aug 2011 11:48:00 -0500
Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
Message-ID: <4E4164C0.8030507@tiendalinux.com>

Hi Guys.

I am starting to use FreeSWITCH, i am an asterisk user since the 1.0.7 
release appears as a package on the debian distribution, at the 
beginning i was amazed by the fact i can build a PBX for my own business 
and i did, later i began to install this system for my customers and 
sooner i meet the problems, however being the software open source i 
always find a way to fix things using patchs from others, sometimes i 
felt how my life was at risk when the system stops working and that 
usually happens when i have to use queues and dealing with digium hardware.

Fixing those problems either by applying patches or by changing the 
hardware where the digium cards were supposed to be installed helps me, 
but that was to much stress for me and seeking for a balance that will 
let me invest more time on services, configuration and hoping to have 
better hardware options brings me to freeswitch.

I agree with freeswitch philosophy that instead of having thousands of 
modules that don't work fine i prefer just a few that works the way it 
should be, a rock solid system for a corporate pbx and a call center is 
what i want.

So here i am trying to begin the conversion, and i hope the information 
we can transcript in this list will help others that want to try another 
alternative to asterisk.

First of all i think the saner for a migration is to have the two 
systems running either on the same machine or different and use the 
stable features of each one.

So could you please freeswitch users help me with this rosetta stone 
migration guide in order to post it to voip-info.org or freeswitch wiki 
(i list only the ones i currently use ):


*Technology* 	*Asterisk* 	*Freeswitch*
PSTN Connectivity (Digium / Sangoma) 	dahdi 	freetdm
IAX2 	mod_iax 	?? none stable yet.
Use Asterisk to forward traffic via SIP.
Enable Hardware HPET for IAX2 trunk if card not available for Asterisk
Bluetooth Channel 	chan_mobile 	??
Use asterisk via SIP
Skype 	Skypeforasterisk (no longer for sale) 	mod_skypeopen
CDR Stadistics 	Arternic cdr-stats 	??
Queue Statistics 	Asteriskguru queue-stats 	??
Web Management 	Freepbx 	??
IVR 	AGI / AMI 	Event Socket
Codec G.729 	Transcodind Cards
G.729 licenses
Free G.729 (Intel IPP) 	Transcodind Cards
G.729 licenses
fsg729 Intel IPP(any experience with it ? )
Fax Handling 	Iaxmodem with Hylafax 	??
Iaxmodem via asterisk to FS via SIP ?
SIP 	chan_sip 	sofia
ACD 	app_queue 	mod_callcenter



Thank you all


-- 
Nestor A. Diaz
Ingeniero de Sistemas
Tel. +57 1-485-3020 x 211
Cel. +57 316-227-3593
Tel. SIP: sip:211 at tiendalinux.com
Email/MSN: nestor at tiendalinux.com
http://www.tiendalinux.com/
Bogota, Colombia

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110809/f353dcbf/attachment.html 

From amjad.soomro at philips.com  Tue Aug  9 21:14:13 2011
From: amjad.soomro at philips.com (Soomro, Amjad)
Date: Tue, 9 Aug 2011 19:14:13 +0200
Subject: [Freeswitch-users] SIP MESSAGE delivery via Freeswitch
Message-ID: <E85A9947AF2FB64AB2E506AE3657E11F602B3062B3@NLCLUEXM12.connect1.local>

Hello,
I am using PJSUA to send SIP MESSAGE to another SIP URL. The message is received at FreeSwitch server and 200 OK response is sent back to the sender.
However, I see no message being forwarded to the receiver at the server. Both wireshark and siptrace reveal no attempt to deliver the message to the receipient. The SIP message and its response is:

------------------------------------------------------------------------

recv 572 bytes from udp/[96.57.214.157]:1024 at 16:16:19.536837:

------------------------------------------------------------------------

MESSAGE sip:1001 at wimed.philips.com SIP/2.0

Via: SIP/2.0/UDP 96.57.214.157:1024;rport;branch=z9hG4bKPjf03b25a7-3d05-46b1-bc16-347cbbd97c07

Max-Forwards: 70

From: <sip:1000 at wimed.philips.com>;tag=9ef1d5f6-d19c-459c-87cf-eaab7ca9dfe2

To: <sip:1001 at wimed.philips.com>

Call-ID: 907e1671-1b1f-495e-9808-5802e7670328

CSeq: 29555 MESSAGE

Accept: text/plain, application/im-iscomposing+xml

Contact: <sip:1000 at 96.57.214.157:1024;transport=UDP;ob>

User-Agent: PJSUA v1.10.0 Linux-2.6.35.13/i686/glibc-2.13

Content-Type: text/plain

Content-Length: 16

SIP REGISTRATION

------------------------------------------------------------------------

send 541 bytes to udp/[96.57.214.157]:1024 at 16:16:19.537837:

------------------------------------------------------------------------

SIP/2.0 200 OK

Via: SIP/2.0/UDP 96.57.214.157:1024;rport=1024;branch=z9hG4bKPjf03b25a7-3d05-46b1-bc16-347cbbd97c07

From: <sip:1000 at wimed.philips.com>;tag=9ef1d5f6-d19c-459c-87cf-eaab7ca9dfe2

To: <sip:1001 at wimed.philips.com>;tag=00tQ14HpaSDUg

Call-ID: 907e1671-1b1f-495e-9808-5802e7670328

CSeq: 29555 MESSAGE

User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE

Supported: timer, precondition, path, replaces

Content-Length: 0



----------------------------------------------------------------



The registration messages from both the clients at Freeswitch are:



------------------------------------------------------------------------

recv 841 bytes from udp/[96.57.214.157]:1024 at 16:15:58.158614:

------------------------------------------------------------------------

REGISTER sip:96.57.214.156 SIP/2.0

Via: SIP/2.0/UDP 96.57.214.157:1024;rport;branch=z9hG4bKPj4b9b314a-e7d7-4e51-9b58-2aab22e78633

Max-Forwards: 70

From: <sip:1000 at wimed.philips.com>;tag=738d0018-6309-47df-81a8-0efbcb12a1b7

To: <sip:1000 at wimed.philips.com>

Call-ID: 5887dcaf-c8b5-4837-a708-ea8c75704f79

CSeq: 14535 REGISTER

User-Agent: PJSUA v1.10.0 Linux-2.6.35.13/i686/glibc-2.13

Contact: <sip:1000 at 96.57.214.157:1024;transport=UDP;ob>

Expires: 300

Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS

Authorization: Digest username="1000", realm="wimed.philips.com", nonce="bba2277d-66b8-4d56-93bf-df343dc7fe52", uri="sip:96.57.214.156", response="b2a31bef17c12d8c0c6c3961ae958932", algorithm=MD5,

cnonce="eb828e16-22b4-4a24-ac21-9b96bc0ad765", qop=auth, nc=00000001

Content-Length: 0

------------------------------------------------------------------------

send 647 bytes to udp/[96.57.214.157]:1024 at 16:15:58.185616:

------------------------------------------------------------------------

SIP/2.0 200 OK

Via: SIP/2.0/UDP 96.57.214.157:1024;rport=1024;branch=z9hG4bKPj4b9b314a-e7d7-4e51-9b58-2aab22e78633

From: <sip:1000 at wimed.philips.com>;tag=738d0018-6309-47df-81a8-0efbcb12a1b7

To: <sip:1000 at wimed.philips.com>;tag=X5eDvKZBKya3D

Call-ID: 5887dcaf-c8b5-4837-a708-ea8c75704f79

CSeq: 14535 REGISTER

Contact: <sip:1000 at 96.57.214.157:1024;transport=UDP;ob>;expires=30

Date: Tue, 09 Aug 2011 16:15:58 GMT

User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE

Supported: timer, precondition, path, replaces

Content-Length: 0

------------------------------------------------------------------------



------------------------------------------------------------------------

recv 841 bytes from udp/[96.57.214.157]:5060 at 16:15:56.372512:

------------------------------------------------------------------------

REGISTER sip:96.57.214.156 SIP/2.0

Via: SIP/2.0/UDP 96.57.214.157:5060;rport;branch=z9hG4bKPj53492c33-27ad-42a5-a5ea-6b0a7fb6760c

Max-Forwards: 70

From: <sip:1001 at wimed.philips.com>;tag=76e9c9d5-7f52-4f22-a17c-11a8c287c227

To: <sip:1001 at wimed.philips.com>

Call-ID: 71251231-581e-4664-85b9-a9cd74877ca5

CSeq: 52991 REGISTER

User-Agent: PJSUA v1.10.0 Linux-2.6.35.13/i686/glibc-2.13

Contact: <sip:1001 at 96.57.214.157:5060;transport=UDP;ob>

Expires: 300

Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS

Authorization: Digest username="1001", realm="wimed.philips.com", nonce="aac8bc16-1644-4986-a4e2-a5749d1ea911", uri="sip:96.57.214.156", response="a5a19227c03a4d053ccb0dd65af23aaf", algorithm=MD5,

cnonce="33e8968a-5f8f-45d3-9ccf-a3e20b50de66", qop=auth, nc=00000001

Content-Length: 0

------------------------------------------------------------------------

send 647 bytes to udp/[96.57.214.157]:5060 at 16:15:56.396514:

------------------------------------------------------------------------

SIP/2.0 200 OK

Via: SIP/2.0/UDP 96.57.214.157:5060;rport=5060;branch=z9hG4bKPj53492c33-27ad-42a5-a5ea-6b0a7fb6760c

From: <sip:1001 at wimed.philips.com>;tag=76e9c9d5-7f52-4f22-a17c-11a8c287c227

To: <sip:1001 at wimed.philips.com>;tag=ta32p2c1U37aB

Call-ID: 71251231-581e-4664-85b9-a9cd74877ca5

CSeq: 52991 REGISTER

Contact: <sip:1001 at 96.57.214.157:5060;transport=UDP;ob>;expires=30

Date: Tue, 09 Aug 2011 16:15:56 GMT

User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE

Supported: timer, precondition, path, replaces

Content-Length: 0

------------------------------------------------------------------



Are there any additional settings (other than default) at Freeswitch server for messages to be actually forwarded to the receipient?



Thanks in advance



Amjad



________________________________
The information contained in this message may be confidential and legally protected under applicable law. The message is intended solely for the addressee(s). If you are not the intended recipient, you are hereby notified that any use, forwarding, dissemination, or reproduction of this message is strictly prohibited and may be unlawful. If you are not the intended recipient, please contact the sender by return e-mail and destroy all copies of the original message.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110809/7563d7f1/attachment-0001.html 

From avi at avimarcus.net  Wed Aug 10 03:11:12 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 10 Aug 2011 02:11:12 +0300
Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
In-Reply-To: <4E4164C0.8030507@tiendalinux.com>
References: <4E4164C0.8030507@tiendalinux.com>
Message-ID: <CALNZuwkWenmeQFh6U93fh5V1bF+b1Sn7VNkRQhX_pQydHQ2Axw@mail.gmail.com>

A lot of this information is on the freeswitch wiki here:
http://wiki.freeswitch.org/wiki/Rosetta_stone
CDR management: http://wiki.freeswitch.org/wiki/Cdr
Web GUIs:
http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ#Q:_Is_there_a_GUI_for_configuring_FreeSWITCH.3F
Queues: mod_fifo or mod_callcenter


-Avi Marcus


On Tue, Aug 9, 2011 at 7:48 PM, Nestor A Diaz <nestor at tiendalinux.com>wrote:

> **
> Hi Guys.
>
> I am starting to use FreeSWITCH, i am an asterisk user since the 1.0.7
> release appears as a package on the debian distribution, at the beginning i
> was amazed by the fact i can build a PBX for my own business and i did,
> later i began to install this system for my customers and sooner i meet the
> problems, however being the software open source i always find a way to fix
> things using patchs from others, sometimes i felt how my life was at risk
> when the system stops working and that usually happens when i have to use
> queues and dealing with digium hardware.
>
> Fixing those problems either by applying patches or by changing the
> hardware where the digium cards were supposed to be installed helps me, but
> that was to much stress for me and seeking for a balance that will let me
> invest more time on services, configuration and hoping to have better
> hardware options brings me to freeswitch.
>
> I agree with freeswitch philosophy that instead of having thousands of
> modules that don't work fine i prefer just a few that works the way it
> should be, a rock solid system for a corporate pbx and a call center is what
> i want.
>
> So here i am trying to begin the conversion, and i hope the information we
> can transcript in this list will help others that want to try another
> alternative to asterisk.
>
> First of all i think the saner for a migration is to have the two systems
> running either on the same machine or different and use the stable features
> of each one.
>
> So could you please freeswitch users help me with this rosetta stone
> migration guide in order to post it to voip-info.org or freeswitch wiki (i
> list only the ones i currently use ):
>
>
>   *Technology* *Asterisk* *Freeswitch*  PSTN Connectivity (Digium /
> Sangoma) dahdi freetdm  IAX2 mod_iax ?? none stable yet.
> Use Asterisk to forward traffic via SIP.
> Enable Hardware HPET for IAX2 trunk if card not available for Asterisk  Bluetooth
> Channel chan_mobile ??
> Use asterisk via SIP
>   Skype Skypeforasterisk (no longer for sale) mod_skypeopen  CDR
> Stadistics Arternic cdr-stats ??  Queue Statistics  Asteriskguru
> queue-stats ??  Web Management Freepbx ??  IVR AGI / AMI Event Socket  Codec
> G.729 Transcodind Cards
> G.729 licenses
> Free G.729 (Intel IPP) Transcodind Cards
> G.729 licenses
> fsg729 Intel IPP(any experience with it ? )  Fax Handling Iaxmodem with
> Hylafax ??
> Iaxmodem via asterisk to FS via SIP ?
>   SIP chan_sip sofia  ACD app_queue mod_callcenter
>
> Thank you all
>
>
> --
> Nestor A. Diaz
> Ingeniero de Sistemas
> Tel. +57 1-485-3020 x 211
> Cel. +57 316-227-3593
> Tel. SIP: sip:211 at tiendalinux.com
> Email/MSN: nestor at tiendalinux.com
> http://www.tiendalinux.com/
> Bogota, Colombia
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110810/167a75ce/attachment.html 

From lakersman2006 at yahoo.com  Wed Aug 10 04:54:09 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Tue, 9 Aug 2011 17:54:09 -0700 (PDT)
Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
In-Reply-To: <4E4164C0.8030507@tiendalinux.com>
References: <4E4164C0.8030507@tiendalinux.com>
Message-ID: <1312937649.7702.YahooMailNeo@web161011.mail.bf1.yahoo.com>

I find that Asterisk's AGI is much easier to use, they allow you to retrieve the dial status much easier than freeswitch's api's. Come on freeswitch, if you want to be better than asterisk, you should make it easier to get the dialstatus, etc. At this point asterisk is still defacto.



________________________________
From: Nestor A Diaz <nestor at tiendalinux.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Tuesday, August 9, 2011 9:48 AM
Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide


Hi Guys.

I am starting to use FreeSWITCH, i am an asterisk user since the
1.0.7
release appears as a package on the debian distribution, at the
beginning i was amazed by the fact i can build a PBX for my own
business
and i did, later i began to install this system for my customers and
sooner i meet the problems, however being the software open source i
always find a way to fix things using patchs from others, sometimes i
felt how my life was at risk when the system stops working and that
usually happens when i have to use queues and dealing with digium
hardware.

Fixing those problems either by applying patches or by changing
the
hardware where the digium cards were supposed to be installed helps me,
but that was to much stress for me and seeking for a balance that will
let me invest more time on services, configuration and hoping to have
better hardware
options brings me to freeswitch.

I agree with freeswitch philosophy that instead of having
thousands of
modules
that don't work fine i prefer just a few that works the way it should
be, a rock solid system for a corporate pbx and a call center is what i
want.

So here i am trying to begin the conversion, and i hope the
information
we can transcript in this list will help others that want to try
another
alternative to asterisk.

First of all i think the saner for a migration is to have the two
systems
running either on the same machine or different and use the stable
features of each one.

So could you please freeswitch users help me with this rosetta
stone migration guide in order to post it to voip-info.org or
freeswitch wiki (i list only the ones i currently use ):


 
Technology Asterisk Freeswitch 
PSTN Connectivity (Digium /
Sangoma) dahdi freetdm 
IAX2 mod_iax ?? none stable yet.
Use Asterisk to forward traffic via SIP.
Enable Hardware HPET for IAX2 trunk if card not available for Asterisk  
Bluetooth Channel chan_mobile ??
Use asterisk via SIP
 
Skype Skypeforasterisk (no longer for sale) mod_skypeopen 
CDR Stadistics Arternic cdr-stats  ?? 
Queue Statistics  Asteriskguru queue-stats ?? 
Web Management Freepbx ?? 
IVR AGI / AMI Event Socket 
Codec G.729 Transcodind Cards
G.729 licenses
Free G.729 (Intel IPP) Transcodind Cards
G.729 licenses
fsg729 Intel IPP(any experience with it ? ) 
Fax Handling Iaxmodem with Hylafax ??
Iaxmodem via asterisk to FS via SIP ?
 
SIP chan_sip sofia 
ACD app_queue mod_callcenter 

Thank you all


-- 
Nestor A. Diaz
Ingeniero de Sistemas
Tel. +57 1-485-3020 x 211
Cel. +57 316-227-3593
Tel. SIP: sip:211 at tiendalinux.com
Email/MSN: nestor at tiendalinux.com
http://www.tiendalinux.com/
Bogota, Colombia 
 

_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110809/aee303c2/attachment-0001.html 

From rzhang at gosilverplus.com  Wed Aug 10 05:35:23 2011
From: rzhang at gosilverplus.com (ran zhang)
Date: Tue, 09 Aug 2011 18:35:23 -0700
Subject: [Freeswitch-users] Sofia not taking new ipaddress from local ip
	'127.0.0.1'
Message-ID: <4E41E05B.5050209@gosilverplus.com>

when I do 'sofia status' after network ipaddress is changed from the 
local ipaddress of '127.0.0.1', its not updated,
has anyone have any idea why???  the new ipaddress is 192.168.0.2



Name                    Type                                       
Data                                 State
================================================================================================ 
internal                 profile               
sip:mod_sofia at 127.0.0.1:5062      RUNNING (0)
192.168.0.2         alias                                   
internal                               ALIASED
================================================================================================= 



From gcd at i.ph  Wed Aug 10 06:07:47 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Wed, 10 Aug 2011 10:07:47 +0800
Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
In-Reply-To: <1312937649.7702.YahooMailNeo@web161011.mail.bf1.yahoo.com>
References: <4E4164C0.8030507@tiendalinux.com>
	<1312937649.7702.YahooMailNeo@web161011.mail.bf1.yahoo.com>
Message-ID: <CAHKqMb7AhJ=9tSDDjYS+K87tjp5g4UMMgaTraNveLb+_KP+hUA@mail.gmail.com>

hi nestor,

you'll find more information here:
http://wiki.freeswitch.org/wiki/Specsheet

for  Web management: fusionpbx,  bluebox

-nandy


On Wed, Aug 10, 2011 at 8:54 AM, Sam <lakersman2006 at yahoo.com> wrote:

> I find that Asterisk's AGI is much easier to use, they allow you to
> retrieve the dial status much easier than freeswitch's api's. Come on
> freeswitch, if you want to be better than asterisk, you should make it
> easier to get the dialstatus, etc. At this point asterisk is still defacto.
>
> ------------------------------
> *From:* Nestor A Diaz <nestor at tiendalinux.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Tuesday, August 9, 2011 9:48 AM
> *Subject:* [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
> Hi Guys.
>
> I am starting to use FreeSWITCH, i am an asterisk user since the 1.0.7
> release appears as a package on the debian distribution, at the beginning i
> was amazed by the fact i can build a PBX for my own business and i did,
> later i began to install this system for my customers and sooner i meet the
> problems, however being the software open source i always find a way to fix
> things using patchs from others, sometimes i felt how my life was at risk
> when the system stops working and that usually happens when i have to use
> queues and dealing with digium hardware.
>
> Fixing those problems either by applying patches or by changing the
> hardware where the digium cards were supposed to be installed helps me, but
> that was to much stress for me and seeking for a balance that will let me
> invest more time on services, configuration and hoping to have better
> hardware options brings me to freeswitch.
>
> I agree with freeswitch philosophy that instead of having thousands of
> modules that don't work fine i prefer just a few that works the way it
> should be, a rock solid system for a corporate pbx and a call center is what
> i want.
>
> So here i am trying to begin the conversion, and i hope the information we
> can transcript in this list will help others that want to try another
> alternative to asterisk.
>
> First of all i think the saner for a migration is to have the two systems
> running either on the same machine or different and use the stable features
> of each one.
>
> So could you please freeswitch users help me with this rosetta stone
> migration guide in order to post it to voip-info.org or freeswitch wiki (i
> list only the ones i currently use ):
>
>
>   *Technology* *Asterisk* *Freeswitch*  PSTN Connectivity (Digium /
> Sangoma) dahdi freetdm  IAX2 mod_iax ?? none stable yet.
> Use Asterisk to forward traffic via SIP.
> Enable Hardware HPET for IAX2 trunk if card not available for Asterisk  Bluetooth
> Channel chan_mobile ??
> Use asterisk via SIP
>   Skype Skypeforasterisk (no longer for sale) mod_skypeopen  CDR
> Stadistics Arternic cdr-stats ??  Queue Statistics  Asteriskguru
> queue-stats ??  Web Management Freepbx ??  IVR AGI / AMI Event Socket  Codec
> G.729 Transcodind Cards
> G.729 licenses
> Free G.729 (Intel IPP) Transcodind Cards
> G.729 licenses
> fsg729 Intel IPP(any experience with it ? )  Fax Handling Iaxmodem with
> Hylafax ??
> Iaxmodem via asterisk to FS via SIP ?
>   SIP chan_sip sofia  ACD app_queue mod_callcenter
>
> Thank you all
>
>
> --
> Nestor A. Diaz
> Ingeniero de Sistemas
> Tel. +57 1-485-3020 x 211
> Cel. +57 316-227-3593
> Tel. SIP: sip:211 at tiendalinux.com
> Email/MSN: nestor at tiendalinux.com
> http://www.tiendalinux.com/
> Bogota, Colombia
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110810/3d838cc3/attachment.html 

From ovvenkatesan at gmail.com  Wed Aug 10 10:33:30 2011
From: ovvenkatesan at gmail.com (ovvenkat)
Date: Wed, 10 Aug 2011 12:03:30 +0530
Subject: [Freeswitch-users] error loading mod_xml_cdr
Message-ID: <CAABykDoopsM7FeEH9s1AUptSD9dJNNYfDJG-Ea66sjmxGkBnag@mail.gmail.com>

Hi to all,


I am using FC13 and latest freeswitch

freeswitch at 192.168.1.110@internal> version
FreeSWITCH Version 1.0.head (git-6d1d4a9 2011-08-09 16-48-58 -0500)

I have installed mod_xml_cdr and verified following files

/usr/local/freeswitch/mod/mod_xml_cdr.la
/usr/local/freeswitch/mod/mod_xml_cdr.so

When I am trying to load xml cdr module, I am getting below error message.
I tried with googling to find out the issue but, no luck.

2011-08-10 11:50:40.088429 [CRIT] switch_loadable_module.c:929 Error Loading
module /usr/local/freeswitch/mod/mod_xml_cdr.so
**/usr/lib/libnssutil3.so: undefined symbol: PR_GetDirectorySeparator**

Any one plz help me to find out what is going wrong here?
-- 

Regards
Venkatesan OV.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110810/326a2590/attachment.html 

From avi at avimarcus.net  Wed Aug 10 12:15:24 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 10 Aug 2011 11:15:24 +0300
Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
In-Reply-To: <1312937649.7702.YahooMailNeo@web161011.mail.bf1.yahoo.com>
References: <4E4164C0.8030507@tiendalinux.com>
	<1312937649.7702.YahooMailNeo@web161011.mail.bf1.yahoo.com>
Message-ID: <CALNZuw=B7+-7DC1RyVakeT+R-1M-=Fbg8tV1zTHWkkBCiVnyMw@mail.gmail.com>

Easier to do what? fs_cli -x "sofia status" or "show channels" or whatever
and you can do "as xml" so you can parse it easier.
But better is to just pick up a library for your language to make the ESL
stuff much easier.

-Avi

On Wed, Aug 10, 2011 at 3:54 AM, Sam <lakersman2006 at yahoo.com> wrote:

> I find that Asterisk's AGI is much easier to use, they allow you to
> retrieve the dial status much easier than freeswitch's api's. Come on
> freeswitch, if you want to be better than asterisk, you should make it
> easier to get the dialstatus, etc. At this point asterisk is still defacto.
>
> ------------------------------
> *From:* Nestor A Diaz <nestor at tiendalinux.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Tuesday, August 9, 2011 9:48 AM
> *Subject:* [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
> Hi Guys.
>
> I am starting to use FreeSWITCH, i am an asterisk user since the 1.0.7
> release appears as a package on the debian distribution, at the beginning i
> was amazed by the fact i can build a PBX for my own business and i did,
> later i began to install this system for my customers and sooner i meet the
> problems, however being the software open source i always find a way to fix
> things using patchs from others, sometimes i felt how my life was at risk
> when the system stops working and that usually happens when i have to use
> queues and dealing with digium hardware.
>
> Fixing those problems either by applying patches or by changing the
> hardware where the digium cards were supposed to be installed helps me, but
> that was to much stress for me and seeking for a balance that will let me
> invest more time on services, configuration and hoping to have better
> hardware options brings me to freeswitch.
>
> I agree with freeswitch philosophy that instead of having thousands of
> modules that don't work fine i prefer just a few that works the way it
> should be, a rock solid system for a corporate pbx and a call center is what
> i want.
>
> So here i am trying to begin the conversion, and i hope the information we
> can transcript in this list will help others that want to try another
> alternative to asterisk.
>
> First of all i think the saner for a migration is to have the two systems
> running either on the same machine or different and use the stable features
> of each one.
>
> So could you please freeswitch users help me with this rosetta stone
> migration guide in order to post it to voip-info.org or freeswitch wiki (i
> list only the ones i currently use ):
>
>
>   *Technology* *Asterisk* *Freeswitch*  PSTN Connectivity (Digium /
> Sangoma) dahdi freetdm  IAX2 mod_iax ?? none stable yet.
> Use Asterisk to forward traffic via SIP.
> Enable Hardware HPET for IAX2 trunk if card not available for Asterisk  Bluetooth
> Channel chan_mobile ??
> Use asterisk via SIP
>   Skype Skypeforasterisk (no longer for sale) mod_skypeopen  CDR
> Stadistics Arternic cdr-stats ??  Queue Statistics  Asteriskguru
> queue-stats ??  Web Management Freepbx ??  IVR AGI / AMI Event Socket  Codec
> G.729 Transcodind Cards
> G.729 licenses
> Free G.729 (Intel IPP) Transcodind Cards
> G.729 licenses
> fsg729 Intel IPP(any experience with it ? )  Fax Handling Iaxmodem with
> Hylafax ??
> Iaxmodem via asterisk to FS via SIP ?
>   SIP chan_sip sofia  ACD app_queue mod_callcenter
>
> Thank you all
>
>
> --
> Nestor A. Diaz
> Ingeniero de Sistemas
> Tel. +57 1-485-3020 x 211
> Cel. +57 316-227-3593
> Tel. SIP: sip:211 at tiendalinux.com
> Email/MSN: nestor at tiendalinux.com
> http://www.tiendalinux.com/
> Bogota, Colombia
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110810/79aef0e5/attachment-0001.html 

From covici at ccs.covici.com  Wed Aug 10 12:26:07 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Wed, 10 Aug 2011 04:26:07 -0400
Subject: [Freeswitch-users] commit broke one of my trunks
Message-ID: <8940.1312964767@ccs.covici.com>

Hi.  I am having a problem where I am  using an asterisk box as a
gateway (asterisk 1.4) and the following commit breaks this trunk -- I
get a 491 response  for any revision after that commit and it works
correctly before this commit.  Here is what git says:
commit 56d67eadf66c1b22652ff1f77002a8d024a93fca
Author: Anthony Minessale <anthm at freeswitch.org>
Date:   Mon Aug 1 10:22:55 2011 -0500

    sdp_m_per_ptime is now implied to be true, if you don't like this
    set it to false but its going to be undefined behaviour.  This
    basically means if you call in with ptime 30 then you have a bunch
    of ptime 20 codecs in your outbound list that there will be one m=
    line with 30 and the original inbound codec and more
m= lines for each discinct ptime in your list.  This is, of course, will
    depend on disable_trancoding or absolute_codec_string as well
 
So, I am not sure what is the matter, I am just using g711 ulaw, but in
any case, is this a bug, or is there a way to configure for this gateway
onlyto get the old behavior back?

Thanks in advance for any suggestions.

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From manavid at gmail.com  Wed Aug 10 12:44:18 2011
From: manavid at gmail.com (Moe Navid)
Date: Wed, 10 Aug 2011 01:44:18 -0700
Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
In-Reply-To: <1312937649.7702.YahooMailNeo@web161011.mail.bf1.yahoo.com>
References: <4E4164C0.8030507@tiendalinux.com>
	<1312937649.7702.YahooMailNeo@web161011.mail.bf1.yahoo.com>
Message-ID: <757CA8EF-25EA-4372-AD39-0853551F4399@gmail.com>

There is no way by any means to compare Asterisk's AGI with the different facilities FreeSWITCH offers you in terms of controlling your call flow.

For almost 3 years I managed a cluster of Asterisk + AGI + AMI with tones of channel locks and core dumps? Asterisk's dial status might seem compelling when you want to do simple things like calling cards etc? but when it comes to complex accounting and routing sky is limitless with the power of FreeSWITCH.

I found FreeSWITCH's learning curve to be like vim, initially it may seem a bit difficult but in long run it pays of very well.  

If you know the difference between Dial command in Asterisk and Bridge in FreeSWITCH you would never go back to Asterisk. I give you just 3 simple examples:
1) Bridge command (via the channel variables) gives you the ability to control PDD on calls. Asterisk does not have such facility nonetheless it does not even bother to give you any useful information about your "Dial Status"! To control the PDD I had to tweak my kamailio.

2) If you want to implement a simple rate engine + fail over routing with asterisk + agi for failover you have to have a loop and watch for CONGESTIONs to select your next route/carrier where as in FreeSWITCH you can just simply define your fail overs in your bridge args.

3) If you are in a cluster, have multiple gateways acting as proxy and you want to define outbound proxy for your carriers/endpoints you either have to define bunch of sip peers with outbound proxies or do it in dirty way which I did, I used to add a header in my outgoing calls X-Carrier-IP and had my kamailio to take care of the rest. In FreeSWITCH you just simply add ;fspath= to your bridge args.

List can go on and on and on?

Asterisk's dial status was the most annoying part of asterisk in my opinion :)

On Aug 9, 2011, at 5:54 PM, Sam wrote:

> I find that Asterisk's AGI is much easier to use, they allow you to retrieve the dial status much easier than freeswitch's api's. Come on freeswitch, if you want to be better than asterisk, you should make it easier to get the dialstatus, etc. At this point asterisk is still defacto.
> 
> From: Nestor A Diaz <nestor at tiendalinux.com>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Tuesday, August 9, 2011 9:48 AM
> Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
> 
> Hi Guys.
> 
> I am starting to use FreeSWITCH, i am an asterisk user since the 1.0.7 release appears as a package on the debian distribution, at the beginning i was amazed by the fact i can build a PBX for my own business and i did, later i began to install this system for my customers and sooner i meet the problems, however being the software open source i always find a way to fix things using patchs from others, sometimes i felt how my life was at risk when the system stops working and that usually happens when i have to use queues and dealing with digium hardware.
> 
> Fixing those problems either by applying patches or by changing the hardware where the digium cards were supposed to be installed helps me, but that was to much stress for me and seeking for a balance that will let me invest more time on services, configuration and hoping to have better hardware options brings me to freeswitch.
> 
> I agree with freeswitch philosophy that instead of having thousands of modules that don't work fine i prefer just a few that works the way it should be, a rock solid system for a corporate pbx and a call center is what i want.
> 
> So here i am trying to begin the conversion, and i hope the information we can transcript in this list will help others that want to try another alternative to asterisk.
> 
> First of all i think the saner for a migration is to have the two systems running either on the same machine or different and use the stable features of each one.
> 
> So could you please freeswitch users help me with this rosetta stone migration guide in order to post it to voip-info.org or freeswitch wiki (i list only the ones i currently use ):
> 
> 
> Technology	Asterisk	Freeswitch
> PSTN Connectivity (Digium / Sangoma)	dahdi	freetdm
> IAX2	mod_iax	?? none stable yet.
> Use Asterisk to forward traffic via SIP.
> Enable Hardware HPET for IAX2 trunk if card not available for Asterisk
> Bluetooth Channel	chan_mobile	??
> Use asterisk via SIP
> Skype	Skypeforasterisk (no longer for sale)	mod_skypeopen
> CDR Stadistics	 Arternic cdr-stats	??
> Queue Statistics	 Asteriskguru queue-stats	??
> Web Management	Freepbx	??
> IVR	AGI / AMI	Event Socket
> Codec G.729	Transcodind Cards
> G.729 licenses
> Free G.729 (Intel IPP)	Transcodind Cards
> G.729 licenses
> fsg729 Intel IPP(any experience with it ? )
> Fax Handling	 Iaxmodem with Hylafax	??
> Iaxmodem via asterisk to FS via SIP ?
> SIP	chan_sip	sofia
> ACD	app_queue	mod_callcenter
> 
> Thank you all
> 
> 
> -- 
> Nestor A. Diaz
> Ingeniero de Sistemas
> Tel. +57 1-485-3020 x 211
> Cel. +57 316-227-3593
> Tel. SIP: sip:211 at tiendalinux.com
> Email/MSN: nestor at tiendalinux.com
> http://www.tiendalinux.com/
> Bogota, Colombia 
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110810/0eef788a/attachment.html 

From steveayre at gmail.com  Wed Aug 10 13:51:44 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 10 Aug 2011 10:51:44 +0100
Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
In-Reply-To: <4E4164C0.8030507@tiendalinux.com>
References: <4E4164C0.8030507@tiendalinux.com>
Message-ID: <CAFiqYu=092K=O5xExqfmW2ZvBeWrxsRC_YAoNjkXSbMhe_zMEQ@mail.gmail.com>

I see Avi Marcus has already directed you to the Wiki page.

As with Asterisk there's no official GUI, but there are several open source
projects providing one. Or you can build your own. A few include blue.box
(2800hz project) and FusionPBX.
http://wiki.freeswitch.org/wiki/Freeswitch_Gui

There are 2 ACD modules: mod_fifo and mod_callcentre. mod_fifo is the older
one. That doesn't mean it's not as good, they just approach the problem in
different ways.

As for G729 which you mentioned... do NOT use the Intel IPP codec. It is
ILLEGAL unless you have purchased a valid licence for it, which is extremely
unlikely. You can support it using a hardware transcoding card (Sangoma),
mod_com_g729 (version licensed by FreeSWITCH) and mod_g729 (which is
passthrough only, no transcoding but fine for bridging calls).

chan_mobile's closest match is probably mod_gsmopen. I believe it uses a
cable rather than bluetooth though, and is faily new so probably
'experimental'.

IAX2 is supported by mod_opal. As you noted though, it isn't as stable as
say mod_sofia.

For fax handling check the T38 functionality provided by mod_spandsp.

-Steve


On 9 August 2011 17:48, Nestor A Diaz <nestor at tiendalinux.com> wrote:

> **
> Hi Guys.
>
> I am starting to use FreeSWITCH, i am an asterisk user since the 1.0.7
> release appears as a package on the debian distribution, at the beginning i
> was amazed by the fact i can build a PBX for my own business and i did,
> later i began to install this system for my customers and sooner i meet the
> problems, however being the software open source i always find a way to fix
> things using patchs from others, sometimes i felt how my life was at risk
> when the system stops working and that usually happens when i have to use
> queues and dealing with digium hardware.
>
> Fixing those problems either by applying patches or by changing the
> hardware where the digium cards were supposed to be installed helps me, but
> that was to much stress for me and seeking for a balance that will let me
> invest more time on services, configuration and hoping to have better
> hardware options brings me to freeswitch.
>
> I agree with freeswitch philosophy that instead of having thousands of
> modules that don't work fine i prefer just a few that works the way it
> should be, a rock solid system for a corporate pbx and a call center is what
> i want.
>
> So here i am trying to begin the conversion, and i hope the information we
> can transcript in this list will help others that want to try another
> alternative to asterisk.
>
> First of all i think the saner for a migration is to have the two systems
> running either on the same machine or different and use the stable features
> of each one.
>
> So could you please freeswitch users help me with this rosetta stone
> migration guide in order to post it to voip-info.org or freeswitch wiki (i
> list only the ones i currently use ):
>
>
>   *Technology* *Asterisk* *Freeswitch*  PSTN Connectivity (Digium /
> Sangoma) dahdi freetdm  IAX2 mod_iax ?? none stable yet.
> Use Asterisk to forward traffic via SIP.
> Enable Hardware HPET for IAX2 trunk if card not available for Asterisk  Bluetooth
> Channel chan_mobile ??
> Use asterisk via SIP
>   Skype Skypeforasterisk (no longer for sale) mod_skypeopen  CDR
> Stadistics Arternic cdr-stats ??  Queue Statistics  Asteriskguru
> queue-stats ??  Web Management Freepbx ??  IVR AGI / AMI Event Socket  Codec
> G.729 Transcodind Cards
> G.729 licenses
> Free G.729 (Intel IPP) Transcodind Cards
> G.729 licenses
> fsg729 Intel IPP(any experience with it ? )  Fax Handling Iaxmodem with
> Hylafax ??
> Iaxmodem via asterisk to FS via SIP ?
>   SIP chan_sip sofia  ACD app_queue mod_callcenter
>
> Thank you all
>
>
> --
> Nestor A. Diaz
> Ingeniero de Sistemas
> Tel. +57 1-485-3020 x 211
> Cel. +57 316-227-3593
> Tel. SIP: sip:211 at tiendalinux.com
> Email/MSN: nestor at tiendalinux.com
> http://www.tiendalinux.com/
> Bogota, Colombia
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110810/4cafae62/attachment-0001.html 

From steveayre at gmail.com  Wed Aug 10 13:54:00 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 10 Aug 2011 10:54:00 +0100
Subject: [Freeswitch-users] error loading mod_xml_cdr
In-Reply-To: <CAABykDoopsM7FeEH9s1AUptSD9dJNNYfDJG-Ea66sjmxGkBnag@mail.gmail.com>
References: <CAABykDoopsM7FeEH9s1AUptSD9dJNNYfDJG-Ea66sjmxGkBnag@mail.gmail.com>
Message-ID: <CAFiqYu=o_6nyN+0XR0DLR25mWqDeRL4pm1NpJm2FFQ90cyMp5Q@mail.gmail.com>

Was it a fresh git checkout or did you git pull? Perhaps you've got a
mixture of 2 different versions...

-Steve


On 10 August 2011 07:33, ovvenkat <ovvenkatesan at gmail.com> wrote:

>
> Hi to all,
>
>
> I am using FC13 and latest freeswitch
>
> freeswitch at 192.168.1.110@internal> version
> FreeSWITCH Version 1.0.head (git-6d1d4a9 2011-08-09 16-48-58 -0500)
>
> I have installed mod_xml_cdr and verified following files
>
> /usr/local/freeswitch/mod/mod_xml_cdr.la
> /usr/local/freeswitch/mod/mod_xml_cdr.so
>
> When I am trying to load xml cdr module, I am getting below error message.
> I tried with googling to find out the issue but, no luck.
>
> 2011-08-10 11:50:40.088429 [CRIT] switch_loadable_module.c:929 Error
> Loading module /usr/local/freeswitch/mod/mod_xml_cdr.so
> **/usr/lib/libnssutil3.so: undefined symbol: PR_GetDirectorySeparator**
>
> Any one plz help me to find out what is going wrong here?
> --
>
> Regards
> Venkatesan OV.
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110810/8c77e655/attachment.html 

From cyril.zlachevsky at gmail.com  Wed Aug 10 14:26:12 2011
From: cyril.zlachevsky at gmail.com (Cyril Zlachevsky)
Date: Wed, 10 Aug 2011 13:26:12 +0300
Subject: [Freeswitch-users] FreeSWITCH CLI in Red Hat Linux
Message-ID: <4E425CC4.2070808@gmail.com>

Hi!
I have RHEL 5 server and FreeSWITCH installed from RPM.
I'm start FreeSWITCH daemon by command
/sbin/service/freeswitch start
Process owner is freeswitch:daemon
I have to use CLI, but don't know how get FreeSWITCH CLI prompt if daeomn is already started.



From gcd at i.ph  Wed Aug 10 14:42:54 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Wed, 10 Aug 2011 18:42:54 +0800
Subject: [Freeswitch-users] FreeSWITCH CLI in Red Hat Linux
In-Reply-To: <4E425CC4.2070808@gmail.com>
References: <4E425CC4.2070808@gmail.com>
Message-ID: <CAHKqMb61WSjAzJHnaPMjaF3GbUY88hn1ZU8UuLt5r-h4ycxiyw@mail.gmail.com>

the CLI command is: fs_cli


On Wed, Aug 10, 2011 at 6:26 PM, Cyril Zlachevsky <
cyril.zlachevsky at gmail.com> wrote:

> Hi!
> I have RHEL 5 server and FreeSWITCH installed from RPM.
> I'm start FreeSWITCH daemon by command
> /sbin/service/freeswitch start
> Process owner is freeswitch:daemon
> I have to use CLI, but don't know how get FreeSWITCH CLI prompt if daeomn
> is already started.
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110810/ec138ca4/attachment.html 

From avi at avimarcus.net  Wed Aug 10 14:43:45 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 10 Aug 2011 13:43:45 +0300
Subject: [Freeswitch-users] FreeSWITCH CLI in Red Hat Linux
In-Reply-To: <4E425CC4.2070808@gmail.com>
References: <4E425CC4.2070808@gmail.com>
Message-ID: <CALNZuwnG8M+v+e32inU+A7se39SoB5k10=irCn-6ouVQmggSkw@mail.gmail.com>

the fs_cli is in the same folder as freeswitch: freeswitch/bin/fs_cli

Read more:
http://wiki.freeswitch.org/wiki/Fs_cli


-Avi Marcus


On Wed, Aug 10, 2011 at 1:26 PM, Cyril Zlachevsky <
cyril.zlachevsky at gmail.com> wrote:

> Hi!
> I have RHEL 5 server and FreeSWITCH installed from RPM.
> I'm start FreeSWITCH daemon by command
> /sbin/service/freeswitch start
> Process owner is freeswitch:daemon
> I have to use CLI, but don't know how get FreeSWITCH CLI prompt if daeomn
> is already started.
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110810/6d2ded8c/attachment.html 

From cyril.zlachevsky at gmail.com  Wed Aug 10 14:37:41 2011
From: cyril.zlachevsky at gmail.com (Cyril Zlachevsky)
Date: Wed, 10 Aug 2011 13:37:41 +0300
Subject: [Freeswitch-users] FreeSWITCH CLI in Red Hat Linux
In-Reply-To: <CALNZuwnG8M+v+e32inU+A7se39SoB5k10=irCn-6ouVQmggSkw@mail.gmail.com>
References: <4E425CC4.2070808@gmail.com>
	<CALNZuwnG8M+v+e32inU+A7se39SoB5k10=irCn-6ouVQmggSkw@mail.gmail.com>
Message-ID: <4E425F75.7040503@gmail.com>

I got error:
# /opt/freeswitch/bin/fs_cli
[ERROR] fs_cli.c:1261 main() Error Connecting [Socket Connection Error]
May be I need to enable CLI in freeswitch.xml?


10.08.2011 13:43, Avi Marcus ?????:
> the fs_cli is in the same folder as freeswitch: freeswitch/bin/fs_cli
>
> Read more:
> http://wiki.freeswitch.org/wiki/Fs_cli
>
>
> -Avi Marcus
>
>
> On Wed, Aug 10, 2011 at 1:26 PM, Cyril Zlachevsky <cyril.zlachevsky at gmail.com
> <mailto:cyril.zlachevsky at gmail.com>> wrote:
>
>     Hi!
>     I have RHEL 5 server and FreeSWITCH installed from RPM.
>     I'm start FreeSWITCH daemon by command
>     /sbin/service/freeswitch start
>     Process owner is freeswitch:daemon
>     I have to use CLI, but don't know how get FreeSWITCH CLI prompt if daeomn is already started.
>
>
>     _______________________________________________
>     Join us at ClueCon 2011, Aug 9-11, Chicago
>     http://www.cluecon.com 877-7-4ACLUE
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From gcd at i.ph  Wed Aug 10 15:03:58 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Wed, 10 Aug 2011 19:03:58 +0800
Subject: [Freeswitch-users] FreeSWITCH CLI in Red Hat Linux
In-Reply-To: <4E425F75.7040503@gmail.com>
References: <4E425CC4.2070808@gmail.com>
	<CALNZuwnG8M+v+e32inU+A7se39SoB5k10=irCn-6ouVQmggSkw@mail.gmail.com>
	<4E425F75.7040503@gmail.com>
Message-ID: <CAHKqMb6RoE+U1w-n7XHTZG14ODnkzynyqA_JKwAJcjab4MOE+w@mail.gmail.com>

you need to  supply the host address: fs_cli -H 127.0.0.1

On Wed, Aug 10, 2011 at 6:37 PM, Cyril Zlachevsky <
cyril.zlachevsky at gmail.com> wrote:

> I got error:
> # /opt/freeswitch/bin/fs_cli
> [ERROR] fs_cli.c:1261 main() Error Connecting [Socket Connection Error]
> May be I need to enable CLI in freeswitch.xml?
>
>
> 10.08.2011 13:43, Avi Marcus ?????:
> > the fs_cli is in the same folder as freeswitch: freeswitch/bin/fs_cli
> >
> > Read more:
> > http://wiki.freeswitch.org/wiki/Fs_cli
> >
> >
> > -Avi Marcus
> >
> >
> > On Wed, Aug 10, 2011 at 1:26 PM, Cyril Zlachevsky <
> cyril.zlachevsky at gmail.com
> > <mailto:cyril.zlachevsky at gmail.com>> wrote:
> >
> >     Hi!
> >     I have RHEL 5 server and FreeSWITCH installed from RPM.
> >     I'm start FreeSWITCH daemon by command
> >     /sbin/service/freeswitch start
> >     Process owner is freeswitch:daemon
> >     I have to use CLI, but don't know how get FreeSWITCH CLI prompt if
> daeomn is already started.
> >
> >
> >     _______________________________________________
> >     Join us at ClueCon 2011, Aug 9-11, Chicago
> >     http://www.cluecon.com 877-7-4ACLUE
> >
> >     FreeSWITCH-users mailing list
> >     FreeSWITCH-users at lists.freeswitch.org <mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     http://www.freeswitch.org
> >
> >
> >
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110810/2f906966/attachment-0001.html 

From cyril.zlachevsky at gmail.com  Wed Aug 10 14:50:18 2011
From: cyril.zlachevsky at gmail.com (Cyril Zlachevsky)
Date: Wed, 10 Aug 2011 13:50:18 +0300
Subject: [Freeswitch-users] FreeSWITCH CLI in Red Hat Linux
In-Reply-To: <CAHKqMb6RoE+U1w-n7XHTZG14ODnkzynyqA_JKwAJcjab4MOE+w@mail.gmail.com>
References: <4E425CC4.2070808@gmail.com>
	<CALNZuwnG8M+v+e32inU+A7se39SoB5k10=irCn-6ouVQmggSkw@mail.gmail.com>
	<4E425F75.7040503@gmail.com>
	<CAHKqMb6RoE+U1w-n7XHTZG14ODnkzynyqA_JKwAJcjab4MOE+w@mail.gmail.com>
Message-ID: <4E42626A.5000801@gmail.com>

same behaviour:
/opt/freeswitch/bin/fs_cli -H 127.0.0.1
[ERROR] fs_cli.c:1261 main() Error Connecting [Socket Connection Error]

10.08.2011 14:03, Nandy Dagondon ?????:
> you need to  supply the host address: fs_cli -H 127.0.0.1
>
> On Wed, Aug 10, 2011 at 6:37 PM, Cyril Zlachevsky <cyril.zlachevsky at gmail.com
> <mailto:cyril.zlachevsky at gmail.com>> wrote:
>
>     I got error:
>     # /opt/freeswitch/bin/fs_cli
>     [ERROR] fs_cli.c:1261 main() Error Connecting [Socket Connection Error]
>     May be I need to enable CLI in freeswitch.xml?
>
>
>     10.08.2011 13:43, Avi Marcus ?????:
>      > the fs_cli is in the same folder as freeswitch: freeswitch/bin/fs_cli
>      >
>      > Read more:
>      > http://wiki.freeswitch.org/wiki/Fs_cli
>      >
>      >
>      > -Avi Marcus
>      >
>      >
>      > On Wed, Aug 10, 2011 at 1:26 PM, Cyril Zlachevsky <cyril.zlachevsky at gmail.com
>     <mailto:cyril.zlachevsky at gmail.com>
>      > <mailto:cyril.zlachevsky at gmail.com <mailto:cyril.zlachevsky at gmail.com>>> wrote:
>      >
>      >     Hi!
>      >     I have RHEL 5 server and FreeSWITCH installed from RPM.
>      >     I'm start FreeSWITCH daemon by command
>      >     /sbin/service/freeswitch start
>      >     Process owner is freeswitch:daemon
>      >     I have to use CLI, but don't know how get FreeSWITCH CLI prompt if daeomn is already started.
>      >


From avi at avimarcus.net  Wed Aug 10 15:34:08 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 10 Aug 2011 14:34:08 +0300
Subject: [Freeswitch-users] FreeSWITCH CLI in Red Hat Linux
In-Reply-To: <4E42626A.5000801@gmail.com>
References: <4E425CC4.2070808@gmail.com>
	<CALNZuwnG8M+v+e32inU+A7se39SoB5k10=irCn-6ouVQmggSkw@mail.gmail.com>
	<4E425F75.7040503@gmail.com>
	<CAHKqMb6RoE+U1w-n7XHTZG14ODnkzynyqA_JKwAJcjab4MOE+w@mail.gmail.com>
	<4E42626A.5000801@gmail.com>
Message-ID: <CALNZuwmc1Y=vEGGtLPv2rvdSoXdvUVHkUKUHPKcqtqjv3wSQQw@mail.gmail.com>

No, you don't need to specify the host when it's localhost. Did you change
the password?
But first.. are you sure freeswitch is currently running? Try a ps aux |
grep freeswitch

-Avi


On Wed, Aug 10, 2011 at 1:50 PM, Cyril Zlachevsky <
cyril.zlachevsky at gmail.com> wrote:

> same behaviour:
> /opt/freeswitch/bin/fs_cli -H 127.0.0.1
> [ERROR] fs_cli.c:1261 main() Error Connecting [Socket Connection Error]
>
> 10.08.2011 14:03, Nandy Dagondon ?????:
> > you need to  supply the host address: fs_cli -H 127.0.0.1
> >
> > On Wed, Aug 10, 2011 at 6:37 PM, Cyril Zlachevsky <
> cyril.zlachevsky at gmail.com
> > <mailto:cyril.zlachevsky at gmail.com>> wrote:
> >
> >     I got error:
> >     # /opt/freeswitch/bin/fs_cli
> >     [ERROR] fs_cli.c:1261 main() Error Connecting [Socket Connection
> Error]
> >     May be I need to enable CLI in freeswitch.xml?
> >
> >
> >     10.08.2011 13:43, Avi Marcus ?????:
> >      > the fs_cli is in the same folder as freeswitch:
> freeswitch/bin/fs_cli
> >      >
> >      > Read more:
> >      > http://wiki.freeswitch.org/wiki/Fs_cli
> >      >
> >      >
> >      > -Avi Marcus
> >      >
> >      >
> >      > On Wed, Aug 10, 2011 at 1:26 PM, Cyril Zlachevsky <
> cyril.zlachevsky at gmail.com
> >     <mailto:cyril.zlachevsky at gmail.com>
> >      > <mailto:cyril.zlachevsky at gmail.com <mailto:
> cyril.zlachevsky at gmail.com>>> wrote:
> >      >
> >      >     Hi!
> >      >     I have RHEL 5 server and FreeSWITCH installed from RPM.
> >      >     I'm start FreeSWITCH daemon by command
> >      >     /sbin/service/freeswitch start
> >      >     Process owner is freeswitch:daemon
> >      >     I have to use CLI, but don't know how get FreeSWITCH CLI
> prompt if daeomn is already started.
> >      >
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110810/ec69605c/attachment.html 

From justlikeef at gmail.com  Wed Aug 10 15:35:08 2011
From: justlikeef at gmail.com (Rob Hutton)
Date: Wed, 10 Aug 2011 07:35:08 -0400
Subject: [Freeswitch-users] FreeSWITCH CLI in Red Hat Linux
In-Reply-To: <4E42626A.5000801@gmail.com>
References: <4E425CC4.2070808@gmail.com>
	<CAHKqMb6RoE+U1w-n7XHTZG14ODnkzynyqA_JKwAJcjab4MOE+w@mail.gmail.com>
	<4E42626A.5000801@gmail.com>
Message-ID: <201108100735.09072.justlikeef@gmail.com>

fs_cli uses mod_event_socket:

http://wiki.freeswitch.org/wiki/Mod_event_socket

Make sure your password matches, the port that you have defined in the event socket config is not being blocked by a firewall, etc.  It is a network service just like any other.

On Wednesday 10 August 2011 06:50:18 Cyril Zlachevsky wrote:
> same behaviour:
> /opt/freeswitch/bin/fs_cli -H 127.0.0.1
> [ERROR] fs_cli.c:1261 main() Error Connecting [Socket Connection Error]
> 
> 10.08.2011 14:03, Nandy Dagondon ?????:
> > you need to  supply the host address: fs_cli -H 127.0.0.1
> >
> > On Wed, Aug 10, 2011 at 6:37 PM, Cyril Zlachevsky <cyril.zlachevsky at gmail.com
> > <mailto:cyril.zlachevsky at gmail.com>> wrote:
> >
> >     I got error:
> >     # /opt/freeswitch/bin/fs_cli
> >     [ERROR] fs_cli.c:1261 main() Error Connecting [Socket Connection Error]
> >     May be I need to enable CLI in freeswitch.xml?
> >
> >
> >     10.08.2011 13:43, Avi Marcus ?????:
> >      > the fs_cli is in the same folder as freeswitch: freeswitch/bin/fs_cli
> >      >
> >      > Read more:
> >      > http://wiki.freeswitch.org/wiki/Fs_cli
> >      >
> >      >
> >      > -Avi Marcus
> >      >
> >      >
> >      > On Wed, Aug 10, 2011 at 1:26 PM, Cyril Zlachevsky <cyril.zlachevsky at gmail.com
> >     <mailto:cyril.zlachevsky at gmail.com>
> >      > <mailto:cyril.zlachevsky at gmail.com <mailto:cyril.zlachevsky at gmail.com>>> wrote:
> >      >
> >      >     Hi!
> >      >     I have RHEL 5 server and FreeSWITCH installed from RPM.
> >      >     I'm start FreeSWITCH daemon by command
> >      >     /sbin/service/freeswitch start
> >      >     Process owner is freeswitch:daemon
> >      >     I have to use CLI, but don't know how get FreeSWITCH CLI prompt if daeomn is already started.
> >      >
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110810/bd7c2758/attachment.html 

From anthony.minessale at gmail.com  Wed Aug 10 17:01:40 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 10 Aug 2011 08:01:40 -0500
Subject: [Freeswitch-users] commit broke one of my trunks
In-Reply-To: <8940.1312964767@ccs.covici.com>
References: <8940.1312964767@ccs.covici.com>
Message-ID: <CAKbxfG-4eRhPFYJ1qXr1mzeb0XJPZMw5QtOurgHikUZp5GwkXA@mail.gmail.com>

Like it says set the variable sdp_m_per_ptime=false before calling or set it
it in vars.xml to make it permanent.
On Aug 10, 2011 3:27 AM, <covici at ccs.covici.com> wrote:
> Hi. I am having a problem where I am using an asterisk box as a
> gateway (asterisk 1.4) and the following commit breaks this trunk -- I
> get a 491 response for any revision after that commit and it works
> correctly before this commit. Here is what git says:
> commit 56d67eadf66c1b22652ff1f77002a8d024a93fca
> Author: Anthony Minessale <anthm at freeswitch.org>
> Date: Mon Aug 1 10:22:55 2011 -0500
>
> sdp_m_per_ptime is now implied to be true, if you don't like this
> set it to false but its going to be undefined behaviour. This
> basically means if you call in with ptime 30 then you have a bunch
> of ptime 20 codecs in your outbound list that there will be one m=
> line with 30 and the original inbound codec and more
> m= lines for each discinct ptime in your list. This is, of course, will
> depend on disable_trancoding or absolute_codec_string as well
>
> So, I am not sure what is the matter, I am just using g711 ulaw, but in
> any case, is this a bug, or is there a way to configure for this gateway
> onlyto get the old behavior back?
>
> Thanks in advance for any suggestions.
>
> --
> Your life is like a penny. You're going to lose it. The question is:
> How do
> you spend it?
>
> John Covici
> covici at ccs.covici.com
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110810/91d67285/attachment-0001.html 

From covici at ccs.covici.com  Wed Aug 10 17:38:12 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Wed, 10 Aug 2011 09:38:12 -0400
Subject: [Freeswitch-users] commit broke one of my trunks
In-Reply-To: <CAKbxfG-4eRhPFYJ1qXr1mzeb0XJPZMw5QtOurgHikUZp5GwkXA@mail.gmail.com>
References: <8940.1312964767@ccs.covici.com>
	<CAKbxfG-4eRhPFYJ1qXr1mzeb0XJPZMw5QtOurgHikUZp5GwkXA@mail.gmail.com>
Message-ID: <17510.1312983492@ccs.covici.com>

Yep, but it also said the behavior is undefined, so I could not figure
out why this would happen -- no strange codecs or anything that I know
of -- any asterisk setting I can change instead?

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

> Like it says set the variable sdp_m_per_ptime=false before calling or set it
> it in vars.xml to make it permanent.
> On Aug 10, 2011 3:27 AM, <covici at ccs.covici.com> wrote:
> > Hi. I am having a problem where I am using an asterisk box as a
> > gateway (asterisk 1.4) and the following commit breaks this trunk -- I
> > get a 491 response for any revision after that commit and it works
> > correctly before this commit. Here is what git says:
> > commit 56d67eadf66c1b22652ff1f77002a8d024a93fca
> > Author: Anthony Minessale <anthm at freeswitch.org>
> > Date: Mon Aug 1 10:22:55 2011 -0500
> >
> > sdp_m_per_ptime is now implied to be true, if you don't like this
> > set it to false but its going to be undefined behaviour. This
> > basically means if you call in with ptime 30 then you have a bunch
> > of ptime 20 codecs in your outbound list that there will be one m=
> > line with 30 and the original inbound codec and more
> > m= lines for each discinct ptime in your list. This is, of course, will
> > depend on disable_trancoding or absolute_codec_string as well
> >
> > So, I am not sure what is the matter, I am just using g711 ulaw, but in
> > any case, is this a bug, or is there a way to configure for this gateway
> > onlyto get the old behavior back?
> >
> > Thanks in advance for any suggestions.
> >
> > --
> > Your life is like a penny. You're going to lose it. The question is:
> > How do
> > you spend it?
> >
> > John Covici
> > covici at ccs.covici.com
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From anthony.minessale at gmail.com  Wed Aug 10 19:52:43 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 10 Aug 2011 10:52:43 -0500
Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
In-Reply-To: <1312937649.7702.YahooMailNeo@web161011.mail.bf1.yahoo.com>
References: <4E4164C0.8030507@tiendalinux.com>
	<1312937649.7702.YahooMailNeo@web161011.mail.bf1.yahoo.com>
Message-ID: <CAKbxfG_802p+G1a9HtLEh_i-K1C06fpZgyR4_NP=jZJR68YK8g@mail.gmail.com>

=D

ok, sure.  If that's your only complaint.... see
commit 9d98d49f0556fb79656c8403f285ae0a615439d3

Some caveats

1) There is actually less specific, more generalized data in this DIALSTATUS
variable than what we already report, when you're ready to move on see the
originate_disposition variable:  It's kind of like going from reporting the
precise geo-location of a cafe in Paris to generalizing it to "EUROPE"

We follow the Q.850 standard for call cause codes and follow the SIP RFC to
map sip response codes to/from the Q.850 equivalent.  Also each module has
its own version "sip_hangup_disposition" for sip so you have both the real
sip response code AND the official Q.850 equiv variables set on each call.


2) We don't have a torture feature so we never return that code.


3) Since our originate can return before a call is answered I added "EARLY"
which means the originate succeeded but its still not answered.

4) For any others that do not map directly to FreeSWITCH, I just installed a
copy of originate_disposition for good measure.

P.S

This email took longer to compose than the patch did while sitting in the
middle of a crowded room so you probably could have simply parsed the
originate originate_disposition yourself but if it helps people get over
being stuck in a paradigm, it's worth it for me to write........


On Tue, Aug 9, 2011 at 7:54 PM, Sam <lakersman2006 at yahoo.com> wrote:

> I find that Asterisk's AGI is much easier to use, they allow you to
> retrieve the dial status much easier than freeswitch's api's. Come on
> freeswitch, if you want to be better than asterisk, you should make it
> easier to get the dialstatus, etc. At this point asterisk is still defacto.
>
> ------------------------------
> *From:* Nestor A Diaz <nestor at tiendalinux.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Tuesday, August 9, 2011 9:48 AM
> *Subject:* [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
> Hi Guys.
>
> I am starting to use FreeSWITCH, i am an asterisk user since the 1.0.7
> release appears as a package on the debian distribution, at the beginning i
> was amazed by the fact i can build a PBX for my own business and i did,
> later i began to install this system for my customers and sooner i meet the
> problems, however being the software open source i always find a way to fix
> things using patchs from others, sometimes i felt how my life was at risk
> when the system stops working and that usually happens when i have to use
> queues and dealing with digium hardware.
>
> Fixing those problems either by applying patches or by changing the
> hardware where the digium cards were supposed to be installed helps me, but
> that was to much stress for me and seeking for a balance that will let me
> invest more time on services, configuration and hoping to have better
> hardware options brings me to freeswitch.
>
> I agree with freeswitch philosophy that instead of having thousands of
> modules that don't work fine i prefer just a few that works the way it
> should be, a rock solid system for a corporate pbx and a call center is what
> i want.
>
> So here i am trying to begin the conversion, and i hope the information we
> can transcript in this list will help others that want to try another
> alternative to asterisk.
>
> First of all i think the saner for a migration is to have the two systems
> running either on the same machine or different and use the stable features
> of each one.
>
> So could you please freeswitch users help me with this rosetta stone
> migration guide in order to post it to voip-info.org or freeswitch wiki (i
> list only the ones i currently use ):
>
>
>   *Technology* *Asterisk* *Freeswitch*  PSTN Connectivity (Digium /
> Sangoma) dahdi freetdm  IAX2 mod_iax ?? none stable yet.
> Use Asterisk to forward traffic via SIP.
> Enable Hardware HPET for IAX2 trunk if card not available for Asterisk  Bluetooth
> Channel chan_mobile ??
> Use asterisk via SIP
>   Skype Skypeforasterisk (no longer for sale) mod_skypeopen  CDR
> Stadistics Arternic cdr-stats ??  Queue Statistics  Asteriskguru
> queue-stats ??  Web Management Freepbx ??  IVR AGI / AMI Event Socket  Codec
> G.729 Transcodind Cards
> G.729 licenses
> Free G.729 (Intel IPP) Transcodind Cards
> G.729 licenses
> fsg729 Intel IPP(any experience with it ? )  Fax Handling Iaxmodem with
> Hylafax ??
> Iaxmodem via asterisk to FS via SIP ?
>   SIP chan_sip sofia  ACD app_queue mod_callcenter
>
> Thank you all
>
>
> --
> Nestor A. Diaz
> Ingeniero de Sistemas
> Tel. +57 1-485-3020 x 211
> Cel. +57 316-227-3593
> Tel. SIP: sip:211 at tiendalinux.com
> Email/MSN: nestor at tiendalinux.com
> http://www.tiendalinux.com/
> Bogota, Colombia
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110810/75e23dc5/attachment.html 

From hmkias at gmail.com  Thu Aug 11 03:37:40 2011
From: hmkias at gmail.com (HM Kias)
Date: Thu, 11 Aug 2011 05:07:40 +0530
Subject: [Freeswitch-users] Custom SIP Profile
Message-ID: <CAMXUpRy=ft+puH8T9YZabiffPMC=cZSJGV-adKEoa64f0=oBzQ@mail.gmail.com>

Hi All,

I m trying to create a SIP profile equivalent to the below asterisk config,
please advise. Domain & outboundproxy are very important in the
registration.

register =>
XXXXXXXX at sia-nas01ca146.srg.com.bs:yyyyy:XXXXXXXX at nas-sbc-01.srg.com.bs:5060/XXXXXXXX
[XXXXXXXX]
host=sia-nas01ca146.srg.com.bs
outboundproxy=nas-sbc-01.srg.com.bs
type=friend
username=XXXXXXXX
fromuser=XXXXXXXX
fromdomain=sia-nas01ca146.srg.com.bs
secret=yyyyy
dtmfmode=rfc2833
disallow=all
allow=ulaw&gsm
context=users
usereqphone=yes
canreinvite=no
insecure=port,invite

Thanks in advance.


Regards,




-- 
HM Kias
91-9443467600
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110811/9513eb53/attachment-0001.html 

From yungwei at resolvity.com  Wed Aug 10 22:44:34 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Wed, 10 Aug 2011 14:44:34 -0400
Subject: [Freeswitch-users] freeswitch.org is down
Message-ID: <33095823FD21DF429B481B5163264B7950FF060640@VMBX102.ihostexchange.net>

Hi,

I noticed that freeswitch.org is down. Any idea when it will be back up?
Thanks.


From cyril.zlachevsky at gmail.com  Wed Aug 10 20:52:23 2011
From: cyril.zlachevsky at gmail.com (Cyril Zlachevsky)
Date: Wed, 10 Aug 2011 19:52:23 +0300
Subject: [Freeswitch-users] FreeSWITCH CLI in Red Hat Linux
In-Reply-To: <CALNZuwmc1Y=vEGGtLPv2rvdSoXdvUVHkUKUHPKcqtqjv3wSQQw@mail.gmail.com>
References: <4E425CC4.2070808@gmail.com>
	<CALNZuwnG8M+v+e32inU+A7se39SoB5k10=irCn-6ouVQmggSkw@mail.gmail.com>
	<4E425F75.7040503@gmail.com>
	<CAHKqMb6RoE+U1w-n7XHTZG14ODnkzynyqA_JKwAJcjab4MOE+w@mail.gmail.com>
	<4E42626A.5000801@gmail.com>
	<CALNZuwmc1Y=vEGGtLPv2rvdSoXdvUVHkUKUHPKcqtqjv3wSQQw@mail.gmail.com>
Message-ID: <4E42B747.9000803@gmail.com>

What password you mean?
And offcause, freeswitch is already running:

# service freeswitch status
freeswitch (pid 7700) is running...

10.08.2011 14:34, Avi Marcus ?????:
> No, you don't need to specify the host when it's localhost. Did you change the password?
> But first.. are you sure freeswitch is currently running? Try a ps aux | grep freeswitch
>
> -Avi
>
>
> On Wed, Aug 10, 2011 at 1:50 PM, Cyril Zlachevsky <cyril.zlachevsky at gmail.com
> <mailto:cyril.zlachevsky at gmail.com>> wrote:
>
>     same behaviour:
>     /opt/freeswitch/bin/fs_cli -H 127.0.0.1
>     [ERROR] fs_cli.c:1261 main() Error Connecting [Socket Connection Error]
>
>     10.08.2011 14:03, Nandy Dagondon ?????:
>      > you need to  supply the host address: fs_cli -H 127.0.0.1
>      >
>      > On Wed, Aug 10, 2011 at 6:37 PM, Cyril Zlachevsky <cyril.zlachevsky at gmail.com
>     <mailto:cyril.zlachevsky at gmail.com>
>      > <mailto:cyril.zlachevsky at gmail.com <mailto:cyril.zlachevsky at gmail.com>>> wrote:
>      >
>      >     I got error:
>      >     # /opt/freeswitch/bin/fs_cli
>      >     [ERROR] fs_cli.c:1261 main() Error Connecting [Socket Connection Error]
>      >     May be I need to enable CLI in freeswitch.xml?
>      >
>      >
>      >     10.08.2011 13:43, Avi Marcus ?????:
>      > > the fs_cli is in the same folder as freeswitch: freeswitch/bin/fs_cli
>      > >
>      > > Read more:
>      > > http://wiki.freeswitch.org/wiki/Fs_cli
>      > >
>      > >
>      > > -Avi Marcus
>      > >
>      > >
>      > > On Wed, Aug 10, 2011 at 1:26 PM, Cyril Zlachevsky <cyril.zlachevsky at gmail.com
>     <mailto:cyril.zlachevsky at gmail.com>
>      > <mailto:cyril.zlachevsky at gmail.com <mailto:cyril.zlachevsky at gmail.com>>
>      > > <mailto:cyril.zlachevsky at gmail.com <mailto:cyril.zlachevsky at gmail.com>
>     <mailto:cyril.zlachevsky at gmail.com <mailto:cyril.zlachevsky at gmail.com>>>> wrote:
>      > >
>      > >     Hi!
>      > >     I have RHEL 5 server and FreeSWITCH installed from RPM.
>      > >     I'm start FreeSWITCH daemon by command
>      > >     /sbin/service/freeswitch start
>      > >     Process owner is freeswitch:daemon
>      > >     I have to use CLI, but don't know how get FreeSWITCH CLI prompt if daeomn is already
>     started.
>      > >
>
>     _______________________________________________
>     Join us at ClueCon 2011, Aug 9-11, Chicago
>     http://www.cluecon.com 877-7-4ACLUE
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From hmkias at gmail.com  Thu Aug 11 04:09:25 2011
From: hmkias at gmail.com (HM Kias)
Date: Thu, 11 Aug 2011 05:39:25 +0530
Subject: [Freeswitch-users] freeswitch.org is down
In-Reply-To: <33095823FD21DF429B481B5163264B7950FF060640@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B7950FF060640@VMBX102.ihostexchange.net>
Message-ID: <CAMXUpRzAASEU8hG1GkF9e9PKrNFrN9gNHUVg5QfSBA3XKSinAw@mail.gmail.com>

Looks like a DNS issue.

On Thu, Aug 11, 2011 at 12:14 AM, Yungwei Chen <yungwei at resolvity.com>wrote:

> Hi,
>
> I noticed that freeswitch.org is down. Any idea when it will be back up?
> Thanks.
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
HM Kias
91-9443467600
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110811/00bafa5b/attachment.html 

From anthony.minessale at gmail.com  Wed Aug 10 23:55:00 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 10 Aug 2011 14:55:00 -0500
Subject: [Freeswitch-users] commit broke one of my trunks
In-Reply-To: <17510.1312983492@ccs.covici.com>
References: <8940.1312964767@ccs.covici.com>
	<CAKbxfG-4eRhPFYJ1qXr1mzeb0XJPZMw5QtOurgHikUZp5GwkXA@mail.gmail.com>
	<17510.1312983492@ccs.covici.com>
Message-ID: <CAKbxfG8NRQAa0rzsKYMTm9XY_T6woV0PKBUO4k1AdMPSxq0e2A@mail.gmail.com>

the problem is when you mix codecs with different ptimes the only real
way to do it is how we now do it with multiple m= lines per the RFC,
asterisk just needs to learn how to look through multiple rtp streams
or simply set the var when you know you are calling asterisk and you
should be ok.



On Wed, Aug 10, 2011 at 8:38 AM,  <covici at ccs.covici.com> wrote:
> Yep, but it also said the behavior is undefined, so I could not figure
> out why this would happen -- no strange codecs or anything that I know
> of -- any asterisk setting I can change instead?
>
> Anthony Minessale <anthony.minessale at gmail.com> wrote:
>
>> Like it says set the variable sdp_m_per_ptime=false before calling or set it
>> it in vars.xml to make it permanent.
>> On Aug 10, 2011 3:27 AM, <covici at ccs.covici.com> wrote:
>> > Hi. I am having a problem where I am using an asterisk box as a
>> > gateway (asterisk 1.4) and the following commit breaks this trunk -- I
>> > get a 491 response for any revision after that commit and it works
>> > correctly before this commit. Here is what git says:
>> > commit 56d67eadf66c1b22652ff1f77002a8d024a93fca
>> > Author: Anthony Minessale <anthm at freeswitch.org>
>> > Date: Mon Aug 1 10:22:55 2011 -0500
>> >
>> > sdp_m_per_ptime is now implied to be true, if you don't like this
>> > set it to false but its going to be undefined behaviour. This
>> > basically means if you call in with ptime 30 then you have a bunch
>> > of ptime 20 codecs in your outbound list that there will be one m=
>> > line with 30 and the original inbound codec and more
>> > m= lines for each discinct ptime in your list. This is, of course, will
>> > depend on disable_trancoding or absolute_codec_string as well
>> >
>> > So, I am not sure what is the matter, I am just using g711 ulaw, but in
>> > any case, is this a bug, or is there a way to configure for this gateway
>> > onlyto get the old behavior back?
>> >
>> > Thanks in advance for any suggestions.
>> >
>> > --
>> > Your life is like a penny. You're going to lose it. The question is:
>> > How do
>> > you spend it?
>> >
>> > John Covici
>> > covici at ccs.covici.com
>> >
>> > _______________________________________________
>> > Join us at ClueCon 2011, Aug 9-11, Chicago
>> > http://www.cluecon.com 877-7-4ACLUE
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> ----------------------------------------------------
>> Alternatives:
>>
>> ----------------------------------------------------
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> --
> Your life is like a penny. ?You're going to lose it. ?The question is:
> How do
> you spend it?
>
> ? ? ? ? John Covici
> ? ? ? ? covici at ccs.covici.com
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From mi.ke at null.net  Thu Aug 11 01:31:02 2011
From: mi.ke at null.net (Mi Ke)
Date: Wed, 10 Aug 2011 21:31:02 +0000
Subject: [Freeswitch-users] Fw: Re: playing value with mod_say & getting
 DTMF input at the same time
Message-ID: <20110810213103.167940@gmx.com>

it works ;)

 some things in freeswitch are hard to belieive until you try them and they just work ... thanks !



----- Original Message -----
From: Mi Ke
Sent: 08/09/11 10:54 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] playing value with mod_say & getting DTMF input at the same time

can I then use phrase:macro_name as a part of the played prompt in play_and_get_digits i.e. "prompt1.wav&phrase:macro_name&prompt2.wav ... etc" ? will such construction really work ?



----- Original Message -----
From: Avi Marcus
Sent: 08/09/11 09:59 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] playing value with mod_say & getting DTMF input at the same time

Create a phrase macro for that and call it for your play_and_get_digits 

-Avi

On Tue, Aug 9, 2011 at 9:55 PM, Mi Ke < mi.ke at null.net > wrote:
Hi All,

 Using mod_say and read function at once - is that possible with Lua ? I need to play a numeric value (like credit amount or time) to Leg A and catch user's input at the same time. As a temporary solution, I wrote a function translating numeric values into a string "prompt1.wav&prompt2.wav....etc" and then suppliy is to the read function. Is there anyway I could use say for that ?

 Thanks / Mike

 _______________________________________________
 Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com  877-7-4ACLUE

 FreeSWITCH-users mailing list
 FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110810/436d23ea/attachment.html 
-------------- next part --------------
_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

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

From ovvenkatesan at gmail.com  Wed Aug 10 21:26:43 2011
From: ovvenkatesan at gmail.com (ovvenkat)
Date: Wed, 10 Aug 2011 22:56:43 +0530
Subject: [Freeswitch-users] error loading mod_xml_cdr
In-Reply-To: <CAFiqYu=o_6nyN+0XR0DLR25mWqDeRL4pm1NpJm2FFQ90cyMp5Q@mail.gmail.com>
References: <CAABykDoopsM7FeEH9s1AUptSD9dJNNYfDJG-Ea66sjmxGkBnag@mail.gmail.com>
	<CAFiqYu=o_6nyN+0XR0DLR25mWqDeRL4pm1NpJm2FFQ90cyMp5Q@mail.gmail.com>
Message-ID: <CAABykDpt45SLFcUCoc+6YiEE-=KsQJ108-2iNCbmEVYOJfYmyQ@mail.gmail.com>

Hi Steve,

I have removed freeswitch source and installation directory and did fresh
git checkout and installation , now works fine.

thanks.


Regards,
Venkat.


On Wed, Aug 10, 2011 at 3:24 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Was it a fresh git checkout or did you git pull? Perhaps you've got a
> mixture of 2 different versions...
>
> -Steve
>
>
> On 10 August 2011 07:33, ovvenkat <ovvenkatesan at gmail.com> wrote:
>
>>
>> Hi to all,
>>
>>
>> I am using FC13 and latest freeswitch
>>
>> freeswitch at 192.168.1.110@internal> version
>> FreeSWITCH Version 1.0.head (git-6d1d4a9 2011-08-09 16-48-58 -0500)
>>
>> I have installed mod_xml_cdr and verified following files
>>
>> /usr/local/freeswitch/mod/mod_xml_cdr.la
>> /usr/local/freeswitch/mod/mod_xml_cdr.so
>>
>> When I am trying to load xml cdr module, I am getting below error
>> message.
>> I tried with googling to find out the issue but, no luck.
>>
>> 2011-08-10 11:50:40.088429 [CRIT] switch_loadable_module.c:929 Error
>> Loading module /usr/local/freeswitch/mod/mod_xml_cdr.so
>> **/usr/lib/libnssutil3.so: undefined symbol: PR_GetDirectorySeparator**
>>
>> Any one plz help me to find out what is going wrong here?
>> --
>>
>> Regards
>> Venkatesan OV.
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 

If you have come to help me, you are wasting your time.
If you have come to because your liberation is bound up in mine, we can work
together.


Regards
Venkatesan OV.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110810/7c3379ac/attachment-0001.html 

From rzhang at gosilverplus.com  Wed Aug 10 23:03:23 2011
From: rzhang at gosilverplus.com (ran zhang)
Date: Wed, 10 Aug 2011 12:03:23 -0700
Subject: [Freeswitch-users] FS terminated maybe due to memory leak?
Message-ID: <4E42D5FB.2020609@gosilverplus.com>

hi:  i'm experiencing problem that FS has memory leak when is an active 
phone call over few hours, and eventually it is terminated, only modules 
we are using are 'mod_console', 'mod_event_socket', 'mod_sofia', 
'mod_freetdm', 'mod_commands', 'mod_conference', 'mod_dptools', 
'mod_dialplan_xml', 'mod_tone_stream'.


From jaybinks at gmail.com  Thu Aug 11 04:48:44 2011
From: jaybinks at gmail.com (jay binks)
Date: Thu, 11 Aug 2011 10:48:44 +1000
Subject: [Freeswitch-users] freeswitch.org is down
In-Reply-To: <33095823FD21DF429B481B5163264B7950FF060640@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B7950FF060640@VMBX102.ihostexchange.net>
Message-ID: <CAO47Hpf4J1Az0WCY8TtSRW-wiGHzcifBH+XoyzkdESbaO8EoSQ@mail.gmail.com>

freeswitch.org is hosted by ICALL I believe...

-------------

EVENT ID:           ICALL08102011-OUTAGE
DATE:               08-10-2011
EVENT START TIME:   12:02 PM CST
EVENT END TIME:     7:24 PM CST
SERVICES/EQUIPMENT: Network, power, voice services
TYPE OF EVENT:      Unplanned outage
IMPACT OF EVENT:    Network service interruption, suboptimal routing

Event Description:

As you may be aware, at 12:02PM CST our primary datacenter, Colo4 in Dallas,
Texas, suffered a major power failure.  Fluctuations in a 2000A 480V primary
electrical feed into the building caused the transfer switch to fail,
resulting in the loss of both generator and utility power.

We immediately routed around all service issues we possibly could.
 Unfortunately, many of our voice services remained down for the duration of
the event.  No service to our new datacenter at 1505 Federal St. datacenter
was affected.

At this time, all services are restored.  Please open a new trouble ticket
if you are experiencing any further issues.  A more detailed explanation
will be sent first thing tomorrow.

We apologize for this unfortunate event.


You are receiving this email because you have signed up as a customer of
iCall Carrier Services.  To prevent some future notices, please login to
your iCall Carrier Services account at http://carriers.icall.com. iCall,
Inc.  1505 Federal St., Suite 200, Dallas, TX, 75201, USA.

On Thu, Aug 11, 2011 at 4:44 AM, Yungwei Chen <yungwei at resolvity.com> wrote:

> Hi,
>
> I noticed that freeswitch.org is down. Any idea when it will be back up?
> Thanks.
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

Jay
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110811/be0482f5/attachment.html 

From michel.daggelinckx at gmail.com  Wed Aug 10 21:54:17 2011
From: michel.daggelinckx at gmail.com (Michel Daggelinckx)
Date: Wed, 10 Aug 2011 19:54:17 +0200
Subject: [Freeswitch-users] Sofia not taking new ipaddress from local ip
	'127.0.0.1'
In-Reply-To: <4E41E05B.5050209@gosilverplus.com>
References: <4E41E05B.5050209@gosilverplus.com>
Message-ID: <CAFYZzbsWkvfYqN02TvDV2J2LK36O0QyPst3WBPqXvhZS6Z3b_w@mail.gmail.com>

127.0.0.1 is a loopback adress that never changes.
it is also known as  'localhost'


On Wed, Aug 10, 2011 at 3:35 AM, ran zhang <rzhang at gosilverplus.com> wrote:

> when I do 'sofia status' after network ipaddress is changed from the
> local ipaddress of '127.0.0.1', its not updated,
> has anyone have any idea why???  the new ipaddress is 192.168.0.2
>
>
>
> Name                    Type
> Data                                 State
>
> ================================================================================================
> internal                 profile
> sip:mod_sofia at 127.0.0.1:5062      RUNNING (0)
> 192.168.0.2         alias
> internal                               ALIASED
>
> =================================================================================================
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110810/4339977f/attachment.html 

From curriegrad2004 at gmail.com  Thu Aug 11 05:24:32 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Wed, 10 Aug 2011 18:24:32 -0700
Subject: [Freeswitch-users] freeswitch.org is down
In-Reply-To: <CAO47Hpf4J1Az0WCY8TtSRW-wiGHzcifBH+XoyzkdESbaO8EoSQ@mail.gmail.com>
References: <33095823FD21DF429B481B5163264B7950FF060640@VMBX102.ihostexchange.net>
	<CAO47Hpf4J1Az0WCY8TtSRW-wiGHzcifBH+XoyzkdESbaO8EoSQ@mail.gmail.com>
Message-ID: <CAMWiMy_8Zi0ZD=CaTteuzzQ3qWzQnOX+i1Um35uHXnL1pSww1A@mail.gmail.com>

It's up now. JIRA and Fisheye are still down as we speak

On Wed, Aug 10, 2011 at 5:48 PM, jay binks <jaybinks at gmail.com> wrote:
> freeswitch.org is hosted by ICALL I believe...
> -------------
> EVENT ID: ? ? ? ? ? ICALL08102011-OUTAGE
> DATE: ? ? ? ? ? ? ? 08-10-2011
> EVENT START TIME: ? 12:02 PM CST
> EVENT END TIME: ? ? 7:24 PM CST
> SERVICES/EQUIPMENT: Network, power, voice services
> TYPE OF EVENT: ? ? ?Unplanned outage
> IMPACT OF EVENT: ? ?Network service interruption, suboptimal routing
>
> Event Description:
>
> As you may be aware, at 12:02PM CST our primary datacenter, Colo4 in Dallas,
> Texas, suffered a major power failure. ?Fluctuations in a 2000A 480V primary
> electrical feed into the building caused the transfer switch to fail,
> resulting in the loss of both generator and utility power.
>
> We immediately routed around all service issues we possibly could.
> ?Unfortunately, many of our voice services remained down for the duration of
> the event. ?No service to our new datacenter at 1505 Federal St. datacenter
> was affected.
>
> At this time, all services are restored. ?Please open a new trouble ticket
> if you are experiencing any further issues. ?A more detailed explanation
> will be sent first thing tomorrow.
>
> We apologize for this unfortunate event.
>
>
> You are receiving this email because you have signed up as a customer of
> iCall Carrier Services. ?To prevent some future notices, please login to
> your iCall Carrier Services account at?http://carriers.icall.com. iCall,
> Inc. ?1505 Federal St., Suite 200, Dallas, TX, 75201, USA.
>
> On Thu, Aug 11, 2011 at 4:44 AM, Yungwei Chen <yungwei at resolvity.com> wrote:
>>
>> Hi,
>>
>> I noticed that freeswitch.org is down. Any idea when it will be back up?
>> Thanks.
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> Jay
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From lakersman2006 at yahoo.com  Wed Aug 10 20:12:56 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Wed, 10 Aug 2011 09:12:56 -0700 (PDT)
Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
In-Reply-To: <757CA8EF-25EA-4372-AD39-0853551F4399@gmail.com>
References: <4E4164C0.8030507@tiendalinux.com>
	<1312937649.7702.YahooMailNeo@web161011.mail.bf1.yahoo.com>
	<757CA8EF-25EA-4372-AD39-0853551F4399@gmail.com>
Message-ID: <1312992776.42043.YahooMailNeo@web161011.mail.bf1.yahoo.com>

So have you had to retrieve the dial status from bridging a call in freeswitch? For the life of me I cannot properly get the answered_time? when looking up the channel variables after the bridge call finishes an answered call.



________________________________
From: Moe Navid <manavid at gmail.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, August 10, 2011 1:44 AM
Subject: Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide


There is no way by any means to compare Asterisk's AGI with the different facilities FreeSWITCH offers you in terms of controlling your call flow.

For almost 3 years I managed a cluster of Asterisk + AGI + AMI with tones of channel locks and core dumps? Asterisk's dial status might seem compelling when you want to do simple things like calling cards etc? but when it comes to complex accounting and routing sky is limitless with the power of FreeSWITCH.

I found FreeSWITCH's learning curve to be like vim, initially it may seem a bit difficult but in long run it pays of very well. ?

If you know the difference between Dial command in Asterisk and Bridge in FreeSWITCH you would never go back to Asterisk. I give you just 3 simple examples:
1) Bridge command (via the channel variables) gives you the ability to control PDD on calls. Asterisk does not have such facility nonetheless it does not even bother to give you any useful information about your "Dial Status"! To control the PDD I had to tweak my kamailio.

2) If you want to implement a simple rate engine + fail over routing with asterisk + agi for failover you have to have a loop and watch for CONGESTIONs to select your next route/carrier where as in FreeSWITCH you can just simply define your fail overs in your bridge args.

3) If you are in a cluster, have multiple gateways acting as proxy and you want to define outbound proxy for your carriers/endpoints you either have to define bunch of sip peers with outbound proxies or do it in dirty way which I did, I used to add a header in my outgoing calls X-Carrier-IP and had my kamailio to take care of the rest. In FreeSWITCH you just simply add ;fspath= to your bridge args.

List can go on and on and on?

Asterisk's dial status was the most annoying part of asterisk in my opinion?:)


On Aug 9, 2011, at 5:54 PM, Sam wrote:

I find that Asterisk's AGI is much easier to use, they allow you to retrieve the dial status much easier than freeswitch's api's. Come on freeswitch, if you want to be better than asterisk, you should make it easier to get the dialstatus, etc. At this point asterisk is still defacto.
>
>
>
>
>________________________________
>From: Nestor A Diaz <nestor at tiendalinux.com>
>To: freeswitch-users at lists.freeswitch.org
>Sent: Tuesday, August 9, 2011 9:48 AM
>Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
>
>Hi Guys.
>
>I am starting to use FreeSWITCH, i am an asterisk user since the
1.0.7
release appears as a package on the debian distribution, at the
beginning i was amazed by the fact i can build a PBX for my own
business
and i did, later i began to install this system for my customers and
sooner i meet the problems, however being the software open source i
always find a way to fix things using patchs from others, sometimes i
felt how my life was at risk when the system stops working and that
usually happens when i have to use queues and dealing with digium
hardware.
>
>Fixing those problems either by applying patches or by changing
the
hardware where the digium cards were supposed to be installed helps me,
but that was to much stress for me and seeking for a balance that will
let me invest more time on services, configuration and hoping to have
better hardware
options brings me to freeswitch.
>
>I agree with freeswitch philosophy that instead of having
thousands of
modules
that don't work fine i prefer just a few that works the way it should
be, a rock solid system for a corporate pbx and a call center is what i
want.
>
>So here i am trying to begin the conversion, and i hope the
information
we can transcript in this list will help others that want to try
another
alternative to asterisk.
>
>First of all i think the saner for a migration is to have the two
systems
running either on the same machine or different and use the stable
features of each one.
>
>So could you please freeswitch users help me with this rosetta
stone migration guide in order to post it to voip-info.org or
freeswitch wiki (i list only the ones i currently use ):
>
>
> 
>Technology Asterisk Freeswitch 
>PSTN Connectivity (Digium /
Sangoma) dahdi freetdm 
>IAX2 mod_iax ?? none stable yet.
>Use Asterisk to forward traffic via SIP.
>Enable Hardware HPET for IAX2 trunk if card not available for Asterisk  
>Bluetooth Channel chan_mobile ??
>Use asterisk via SIP
> 
>Skype Skypeforasterisk (no longer for sale) mod_skypeopen 
>CDR Stadistics Arternic cdr-stats  ?? 
>Queue Statistics  Asteriskguru queue-stats ?? 
>Web Management Freepbx ?? 
>IVR AGI / AMI Event Socket 
>Codec G.729 Transcodind Cards
>G.729 licenses
>Free G.729 (Intel IPP) Transcodind Cards
>G.729 licenses
>fsg729 Intel IPP(any experience with it ? ) 
>Fax Handling Iaxmodem with Hylafax ??
>Iaxmodem via asterisk to FS via SIP ?
> 
>SIP chan_sip sofia 
>ACD app_queue mod_callcenter 
>
>Thank you all
>
>
>-- 
>Nestor A. Diaz
>Ingeniero de Sistemas
>Tel. +57 1-485-3020 x 211
>Cel. +57 316-227-3593
>Tel. SIP: sip:211 at tiendalinux.com
>Email/MSN: nestor at tiendalinux.com
>http://www.tiendalinux.com/
>Bogota, Colombia 
> 
>
>_______________________________________________
>Join us at ClueCon 2011, Aug 9-11, Chicago
>http://www.cluecon.com 877-7-4ACLUE
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>
>_______________________________________________
>Join us at ClueCon 2011, Aug 9-11, Chicago
>http://www.cluecon.com 877-7-4ACLUE
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>

_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110810/4c23e6cf/attachment-0001.html 

From lakersman2006 at yahoo.com  Wed Aug 10 20:28:39 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Wed, 10 Aug 2011 09:28:39 -0700 (PDT)
Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
In-Reply-To: <CAKbxfG_802p+G1a9HtLEh_i-K1C06fpZgyR4_NP=jZJR68YK8g@mail.gmail.com>
References: <4E4164C0.8030507@tiendalinux.com>
	<1312937649.7702.YahooMailNeo@web161011.mail.bf1.yahoo.com>
	<CAKbxfG_802p+G1a9HtLEh_i-K1C06fpZgyR4_NP=jZJR68YK8g@mail.gmail.com>
Message-ID: <1312993719.14274.YahooMailNeo@web161018.mail.bf1.yahoo.com>

Thanks for being so accommodating. I was a bit frustrated in trying to port over an asterisk agi script to freeswitch. I have spent many hours trying to learn how to configure freeswitch, I was about to give up, but I will play with the new changes you made and see if that works for me. 


One other question, when the bridged call hangs up I do not see any value for the hangup time when using getVariable("hangup_time"), so how can I get it?



________________________________
From: Anthony Minessale <anthony.minessale at gmail.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, August 10, 2011 8:52 AM
Subject: Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide


=D?

ok, sure. ?If that's your only complaint.... see commit?9d98d49f0556fb79656c8403f285ae0a615439d3

Some caveats

1) There is actually less?specific, more generalized data in this DIALSTATUS variable than what we already report, when you're ready to move on see the originate_disposition variable: ?It's kind of like going from reporting the precise geo-location of a cafe in Paris to generalizing it to "EUROPE"?

We follow the Q.850 standard for call cause codes and follow the SIP RFC to map sip response codes to/from the Q.850?equivalent. ?Also each module has its own version "sip_hangup_disposition" for sip so you have both the real sip response code AND the official Q.850 equiv variables set on each call.


2) We don't have a torture feature so we never return that code.


3) Since our originate can return before a call is answered I added "EARLY" which means the originate succeeded but its still not answered.

4) For any others that do not map directly to FreeSWITCH, I just installed a copy of originate_disposition for good measure.

P.S?

This email took longer to compose than the patch did while sitting in the middle of a crowded room so you probably could have simply parsed the originate originate_disposition yourself but if it helps people get over being stuck in a?paradigm, it's worth it for me to write........
?


On Tue, Aug 9, 2011 at 7:54 PM, Sam <lakersman2006 at yahoo.com> wrote:

I find that Asterisk's AGI is much easier to use, they allow you to retrieve the dial status much easier than freeswitch's api's. Come on freeswitch, if you want to be better than asterisk, you should make it easier to get the dialstatus, etc. At this point asterisk is still defacto.
>
>
>
>
>________________________________
> From: Nestor A Diaz <nestor at tiendalinux.com>
>To: freeswitch-users at lists.freeswitch.org
>Sent: Tuesday, August 9, 2011 9:48 AM
>Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
>
>
>Hi Guys.
>
>I am starting to use FreeSWITCH, i am an asterisk user since the
1.0.7
release appears as a package on the debian distribution, at the
beginning i was amazed by the fact i can build a PBX for my own
business
and i did, later i began to install this system for my customers and
sooner i meet the problems, however being the software open source i
always find a way to fix things using patchs from others, sometimes i
felt how my life was at risk when the system stops working and that
usually happens when i have to use queues and dealing with digium
hardware.
>
>Fixing those problems either by applying patches or by changing
the
hardware where the digium cards were supposed to be installed helps me,
but that was to much stress for me and seeking for a balance that will
let me invest more time on services, configuration and hoping to have
better hardware
options brings me to freeswitch.
>
>I agree with freeswitch philosophy that instead of having
thousands of
modules
that don't work fine i prefer just a few that works the way it should
be, a rock solid system for a corporate pbx and a call center is what i
want.
>
>So here i am trying to begin the conversion, and i hope the
information
we can transcript in this list will help others that want to try
another
alternative to asterisk.
>
>First of all i think the saner for a migration is to have the two
systems
running either on the same machine or different and use the stable
features of each one.
>
>So could you please freeswitch users help me with this rosetta
stone migration guide in order to post it to voip-info.org or
freeswitch wiki (i list only the ones i currently use ):
>
>
>
>Technology Asterisk Freeswitch 
>PSTN Connectivity (Digium /
Sangoma) dahdi freetdm 
>IAX2 mod_iax ?? none stable yet.
>Use Asterisk to forward traffic via SIP.
>Enable Hardware HPET for IAX2 trunk if card not available for Asterisk  
>Bluetooth Channel chan_mobile ??
>Use asterisk via SIP
> 
>Skype Skypeforasterisk (no longer for sale) mod_skypeopen 
>CDR Stadistics Arternic cdr-stats  ?? 
>Queue Statistics  Asteriskguru queue-stats ?? 
>Web Management Freepbx ?? 
>IVR AGI / AMI Event Socket 
>Codec G.729 Transcodind Cards
>G.729 licenses
>Free G.729 (Intel IPP) Transcodind Cards
>G.729 licenses
>fsg729 Intel IPP(any experience with it ? ) 
>Fax Handling Iaxmodem with Hylafax ??
>Iaxmodem via asterisk to FS via SIP ?
> 
>SIP chan_sip sofia 
>ACD app_queue mod_callcenter 
>
>Thank you all
>
>
>-- 
>Nestor A. Diaz
>Ingeniero de Sistemas
>Tel. +57 1-485-3020 x 211
>Cel. +57 316-227-3593
>Tel. SIP: sip:211 at tiendalinux.com
>Email/MSN: nestor at tiendalinux.com
>http://www.tiendalinux.com/
>Bogota, Colombia 
> 
>
>
>_______________________________________________
>Join us at ClueCon 2011, Aug 9-11, Chicago
>http://www.cluecon.com 877-7-4ACLUE
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>
>
>_______________________________________________
>Join us at ClueCon 2011, Aug 9-11, Chicago
>http://www.cluecon.com 877-7-4ACLUE
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900

_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110810/0683222f/attachment.html 

From cyril.zlachevsky at gmail.com  Wed Aug 10 20:55:36 2011
From: cyril.zlachevsky at gmail.com (Cyril Zlachevsky)
Date: Wed, 10 Aug 2011 19:55:36 +0300
Subject: [Freeswitch-users] FreeSWITCH CLI in Red Hat Linux
In-Reply-To: <201108100735.09072.justlikeef@gmail.com>
References: <4E425CC4.2070808@gmail.com>
	<CAHKqMb6RoE+U1w-n7XHTZG14ODnkzynyqA_JKwAJcjab4MOE+w@mail.gmail.com>
	<4E42626A.5000801@gmail.com>
	<201108100735.09072.justlikeef@gmail.com>
Message-ID: <4E42B808.80301@gmail.com>

Thank you - this is solution for my problem:

<load module="mod_event_socket"/>


10.08.2011 14:35, Rob Hutton ?????:
> fs_cli uses mod_event_socket:
>
>
> http://wiki.freeswitch.org/wiki/Mod_event_socket
>
>
> Make sure your password matches, the port that you have defined in the event socket config is not
> being blocked by a firewall, etc. It is a network service just like any other.
>
>
> On Wednesday 10 August 2011 06:50:18 Cyril Zlachevsky wrote:
>
>  > same behaviour:
>
>  > /opt/freeswitch/bin/fs_cli -H 127.0.0.1
>
>  > [ERROR] fs_cli.c:1261 main() Error Connecting [Socket Connection Error]
>
>  >
>
>  > 10.08.2011 14:03, Nandy Dagondon ?????:
>
>  > > you need to supply the host address: fs_cli -H 127.0.0.1
>
>  > >
>
>  > > On Wed, Aug 10, 2011 at 6:37 PM, Cyril Zlachevsky <cyril.zlachevsky at gmail.com
>
>  > > <mailto:cyril.zlachevsky at gmail.com>> wrote:
>
>  > >
>
>  > > I got error:
>
>  > > # /opt/freeswitch/bin/fs_cli
>
>  > > [ERROR] fs_cli.c:1261 main() Error Connecting [Socket Connection Error]
>
>  > > May be I need to enable CLI in freeswitch.xml?
>
>  > >
>
>  > >
>
>  > > 10.08.2011 13:43, Avi Marcus ?????:
>
>  > > > the fs_cli is in the same folder as freeswitch: freeswitch/bin/fs_cli
>
>  > > >
>
>  > > > Read more:
>
>  > > > http://wiki.freeswitch.org/wiki/Fs_cli
>
>  > > >
>
>  > > >
>
>  > > > -Avi Marcus
>
>  > > >
>
>  > > >
>
>  > > > On Wed, Aug 10, 2011 at 1:26 PM, Cyril Zlachevsky <cyril.zlachevsky at gmail.com
>
>  > > <mailto:cyril.zlachevsky at gmail.com>
>
>  > > > <mailto:cyril.zlachevsky at gmail.com <mailto:cyril.zlachevsky at gmail.com>>> wrote:
>
>  > > >
>
>  > > > Hi!
>
>  > > > I have RHEL 5 server and FreeSWITCH installed from RPM.
>
>  > > > I'm start FreeSWITCH daemon by command
>
>  > > > /sbin/service/freeswitch start
>
>  > > > Process owner is freeswitch:daemon
>
>  > > > I have to use CLI, but don't know how get FreeSWITCH CLI prompt if daeomn is already started.
>
>  > > >
>
>  >
>
>  > _______________________________________________
>
>  > Join us at ClueCon 2011, Aug 9-11, Chicago
>
>  > http://www.cluecon.com 877-7-4ACLUE
>
>  >
>
>  > FreeSWITCH-users mailing list
>
>  > FreeSWITCH-users at lists.freeswitch.org
>
>  > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
>  > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
>  > http://www.freeswitch.org
>
>  >
>
>



From covici at ccs.covici.com  Thu Aug 11 05:54:48 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Wed, 10 Aug 2011 21:54:48 -0400
Subject: [Freeswitch-users] commit broke one of my trunks
In-Reply-To: <CAKbxfG8NRQAa0rzsKYMTm9XY_T6woV0PKBUO4k1AdMPSxq0e2A@mail.gmail.com>
References: <8940.1312964767@ccs.covici.com>
	<CAKbxfG-4eRhPFYJ1qXr1mzeb0XJPZMw5QtOurgHikUZp5GwkXA@mail.gmail.com>
	<17510.1312983492@ccs.covici.com>
	<CAKbxfG8NRQAa0rzsKYMTm9XY_T6woV0PKBUO4k1AdMPSxq0e2A@mail.gmail.com>
Message-ID: <22874.1313027688@ccs.covici.com>

OK, I will try and see what happens.

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

> the problem is when you mix codecs with different ptimes the only real
> way to do it is how we now do it with multiple m= lines per the RFC,
> asterisk just needs to learn how to look through multiple rtp streams
> or simply set the var when you know you are calling asterisk and you
> should be ok.
> 
> 
> 
> On Wed, Aug 10, 2011 at 8:38 AM,  <covici at ccs.covici.com> wrote:
> > Yep, but it also said the behavior is undefined, so I could not figure
> > out why this would happen -- no strange codecs or anything that I know
> > of -- any asterisk setting I can change instead?
> >
> > Anthony Minessale <anthony.minessale at gmail.com> wrote:
> >
> >> Like it says set the variable sdp_m_per_ptime=false before calling or set it
> >> it in vars.xml to make it permanent.
> >> On Aug 10, 2011 3:27 AM, <covici at ccs.covici.com> wrote:
> >> > Hi. I am having a problem where I am using an asterisk box as a
> >> > gateway (asterisk 1.4) and the following commit breaks this trunk -- I
> >> > get a 491 response for any revision after that commit and it works
> >> > correctly before this commit. Here is what git says:
> >> > commit 56d67eadf66c1b22652ff1f77002a8d024a93fca
> >> > Author: Anthony Minessale <anthm at freeswitch.org>
> >> > Date: Mon Aug 1 10:22:55 2011 -0500
> >> >
> >> > sdp_m_per_ptime is now implied to be true, if you don't like this
> >> > set it to false but its going to be undefined behaviour. This
> >> > basically means if you call in with ptime 30 then you have a bunch
> >> > of ptime 20 codecs in your outbound list that there will be one m=
> >> > line with 30 and the original inbound codec and more
> >> > m= lines for each discinct ptime in your list. This is, of course, will
> >> > depend on disable_trancoding or absolute_codec_string as well
> >> >
> >> > So, I am not sure what is the matter, I am just using g711 ulaw, but in
> >> > any case, is this a bug, or is there a way to configure for this gateway
> >> > onlyto get the old behavior back?
> >> >
> >> > Thanks in advance for any suggestions.
> >> >
> >> > --
> >> > Your life is like a penny. You're going to lose it. The question is:
> >> > How do
> >> > you spend it?
> >> >
> >> > John Covici
> >> > covici at ccs.covici.com
> >> >
> >> > _______________________________________________
> >> > Join us at ClueCon 2011, Aug 9-11, Chicago
> >> > http://www.cluecon.com 877-7-4ACLUE
> >> >
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >>
> >> ----------------------------------------------------
> >> Alternatives:
> >>
> >> ----------------------------------------------------
> >> _______________________________________________
> >> Join us at ClueCon 2011, Aug 9-11, Chicago
> >> http://www.cluecon.com 877-7-4ACLUE
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > --
> > Your life is like a penny. ?You're going to lose it. ?The question is:
> > How do
> > you spend it?
> >
> > ? ? ? ? John Covici
> > ? ? ? ? covici at ccs.covici.com
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From peder at networkoblivion.com  Thu Aug 11 06:23:16 2011
From: peder at networkoblivion.com (Peder)
Date: Wed, 10 Aug 2011 21:23:16 -0500
Subject: [Freeswitch-users] freeswitch.org is down
In-Reply-To: <CAO47Hpf4J1Az0WCY8TtSRW-wiGHzcifBH+XoyzkdESbaO8EoSQ@mail.gmail.com>
References: <33095823FD21DF429B481B5163264B7950FF060640@VMBX102.ihostexchange.net>
	<CAO47Hpf4J1Az0WCY8TtSRW-wiGHzcifBH+XoyzkdESbaO8EoSQ@mail.gmail.com>
Message-ID: <003601cc57cd$a1348e30$e39daa90$@com>

Yep, we have stuff at Colo4 as well and we were down for 8+ hours.  Complete
outage of phone, email and Internet.  Weeee..

 

Peder

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of jay
binks
Sent: Wednesday, August 10, 2011 7:49 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] freeswitch.org is down

 

freeswitch.org is hosted by ICALL I believe...

 

-------------

 

EVENT ID:           ICALL08102011-OUTAGE
DATE:               08-10-2011
EVENT START TIME:   12:02 PM CST
EVENT END TIME:     7:24 PM CST
SERVICES/EQUIPMENT: Network, power, voice services
TYPE OF EVENT:      Unplanned outage
IMPACT OF EVENT:    Network service interruption, suboptimal routing

Event Description:

As you may be aware, at 12:02PM CST our primary datacenter, Colo4 in Dallas,
Texas, suffered a major power failure.  Fluctuations in a 2000A 480V primary
electrical feed into the building caused the transfer switch to fail,
resulting in the loss of both generator and utility power.

We immediately routed around all service issues we possibly could.
Unfortunately, many of our voice services remained down for the duration of
the event.  No service to our new datacenter at 1505 Federal St. datacenter
was affected.

At this time, all services are restored.  Please open a new trouble ticket
if you are experiencing any further issues.  A more detailed explanation
will be sent first thing tomorrow.

We apologize for this unfortunate event.


You are receiving this email because you have signed up as a customer of
iCall Carrier Services.  To prevent some future notices, please login to
your iCall Carrier Services account at  <http://carriers.icall.com/>
http://carriers.icall.com. iCall, Inc.  1505 Federal St., Suite 200, Dallas,
TX, 75201, USA.

On Thu, Aug 11, 2011 at 4:44 AM, Yungwei Chen <yungwei at resolvity.com> wrote:

Hi,

I noticed that freeswitch.org is down. Any idea when it will be back up?
Thanks.

_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

FreeSWITCH-users mailing list
FreeSWITCH-users at 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

Jay

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110810/47929f33/attachment.html 

From gcd at i.ph  Thu Aug 11 06:40:31 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Thu, 11 Aug 2011 10:40:31 +0800
Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
In-Reply-To: <1312993719.14274.YahooMailNeo@web161018.mail.bf1.yahoo.com>
References: <4E4164C0.8030507@tiendalinux.com>
	<1312937649.7702.YahooMailNeo@web161011.mail.bf1.yahoo.com>
	<CAKbxfG_802p+G1a9HtLEh_i-K1C06fpZgyR4_NP=jZJR68YK8g@mail.gmail.com>
	<1312993719.14274.YahooMailNeo@web161018.mail.bf1.yahoo.com>
Message-ID: <CAHKqMb5v891O92bLj5t81Hm2H2XwZJ0J7QyWcYw3ho_XTL2jOw@mail.gmail.com>

hi sam,

i found this post
http://comments.gmane.org/gmane.comp.telephony.freeswitch.user/8477

modify the script to suit your need.

hope it helps.  just dig on w/ FS :-)
-nandy

On Thu, Aug 11, 2011 at 12:28 AM, Sam <lakersman2006 at yahoo.com> wrote:

> Thanks for being so accommodating. I was a bit frustrated in trying to port
> over an asterisk agi script to freeswitch. I have spent many hours trying to
> learn how to configure freeswitch, I was about to give up, but I will play
> with the new changes you made and see if that works for me.
>
> One other question, when the bridged call hangs up I do not see any value
> for the hangup time when using getVariable("hangup_time"), so how can I get
> it?
>
> ------------------------------
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Wednesday, August 10, 2011 8:52 AM
> *Subject:* Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
> =D
>
> ok, sure.  If that's your only complaint.... see
> commit 9d98d49f0556fb79656c8403f285ae0a615439d3
>
> Some caveats
>
> 1) There is actually less specific, more generalized data in this
> DIALSTATUS variable than what we already report, when you're ready to move
> on see the originate_disposition variable:  It's kind of like going from
> reporting the precise geo-location of a cafe in Paris to generalizing it to
> "EUROPE"
>
> We follow the Q.850 standard for call cause codes and follow the SIP RFC to
> map sip response codes to/from the Q.850 equivalent.  Also each module has
> its own version "sip_hangup_disposition" for sip so you have both the real
> sip response code AND the official Q.850 equiv variables set on each call.
>
>
> 2) We don't have a torture feature so we never return that code.
>
>
> 3) Since our originate can return before a call is answered I added "EARLY"
> which means the originate succeeded but its still not answered.
>
> 4) For any others that do not map directly to FreeSWITCH, I just installed
> a copy of originate_disposition for good measure.
>
> P.S
>
> This email took longer to compose than the patch did while sitting in the
> middle of a crowded room so you probably could have simply parsed the
> originate originate_disposition yourself but if it helps people get over
> being stuck in a paradigm, it's worth it for me to write........
>
>
> On Tue, Aug 9, 2011 at 7:54 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> I find that Asterisk's AGI is much easier to use, they allow you to
> retrieve the dial status much easier than freeswitch's api's. Come on
> freeswitch, if you want to be better than asterisk, you should make it
> easier to get the dialstatus, etc. At this point asterisk is still defacto.
>
> ------------------------------
> *From:* Nestor A Diaz <nestor at tiendalinux.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Tuesday, August 9, 2011 9:48 AM
> *Subject:* [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
> Hi Guys.
>
> I am starting to use FreeSWITCH, i am an asterisk user since the 1.0.7
> release appears as a package on the debian distribution, at the beginning i
> was amazed by the fact i can build a PBX for my own business and i did,
> later i began to install this system for my customers and sooner i meet the
> problems, however being the software open source i always find a way to fix
> things using patchs from others, sometimes i felt how my life was at risk
> when the system stops working and that usually happens when i have to use
> queues and dealing with digium hardware.
>
> Fixing those problems either by applying patches or by changing the
> hardware where the digium cards were supposed to be installed helps me, but
> that was to much stress for me and seeking for a balance that will let me
> invest more time on services, configuration and hoping to have better
> hardware options brings me to freeswitch.
>
> I agree with freeswitch philosophy that instead of having thousands of
> modules that don't work fine i prefer just a few that works the way it
> should be, a rock solid system for a corporate pbx and a call center is what
> i want.
>
> So here i am trying to begin the conversion, and i hope the information we
> can transcript in this list will help others that want to try another
> alternative to asterisk.
>
> First of all i think the saner for a migration is to have the two systems
> running either on the same machine or different and use the stable features
> of each one.
>
> So could you please freeswitch users help me with this rosetta stone
> migration guide in order to post it to voip-info.org or freeswitch wiki (i
> list only the ones i currently use ):
>
>
>   *Technology* *Asterisk* *Freeswitch*  PSTN Connectivity (Digium /
> Sangoma) dahdi freetdm  IAX2 mod_iax ?? none stable yet.
> Use Asterisk to forward traffic via SIP.
> Enable Hardware HPET for IAX2 trunk if card not available for Asterisk  Bluetooth
> Channel chan_mobile ??
> Use asterisk via SIP
>   Skype Skypeforasterisk (no longer for sale) mod_skypeopen  CDR
> Stadistics Arternic cdr-stats ??  Queue Statistics  Asteriskguru
> queue-stats ??  Web Management Freepbx ??  IVR AGI / AMI Event Socket  Codec
> G.729 Transcodind Cards
> G.729 licenses
> Free G.729 (Intel IPP) Transcodind Cards
> G.729 licenses
> fsg729 Intel IPP(any experience with it ? )  Fax Handling Iaxmodem with
> Hylafax ??
> Iaxmodem via asterisk to FS via SIP ?
>   SIP chan_sip sofia  ACD app_queue mod_callcenter
>
> Thank you all
>
>
> --
> Nestor A. Diaz
> Ingeniero de Sistemas
> Tel. +57 1-485-3020 x 211
> Cel. +57 316-227-3593
> Tel. SIP: sip:211 at tiendalinux.com
> Email/MSN: nestor at tiendalinux.com
> http://www.tiendalinux.com/
> Bogota, Colombia
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110811/e6395121/attachment-0001.html 

From nestor at tiendalinux.com  Wed Aug 10 04:19:53 2011
From: nestor at tiendalinux.com (Nestor A Diaz)
Date: Tue, 09 Aug 2011 19:19:53 -0500
Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
In-Reply-To: <CALNZuwkWenmeQFh6U93fh5V1bF+b1Sn7VNkRQhX_pQydHQ2Axw@mail.gmail.com>
References: <4E4164C0.8030507@tiendalinux.com>
	<CALNZuwkWenmeQFh6U93fh5V1bF+b1Sn7VNkRQhX_pQydHQ2Axw@mail.gmail.com>
Message-ID: <4E41CEA9.7040004@tiendalinux.com>

Talking about GUIs, which one do you recommend ?

    * blue.box
    * fusionpbx
    * wikipbx

I haven't found any screenshos of blue.box, does anybody know where i 
can take a look at them ? wikipbx seems dead, fusionpbx seems to be 
enought for my requirements, but what about blue.box ?

Thanks.

On 08/09/2011 06:11 PM, Avi Marcus wrote:
> A lot of this information is on the freeswitch wiki here: 
> http://wiki.freeswitch.org/wiki/Rosetta_stone
> CDR management: http://wiki.freeswitch.org/wiki/Cdr
> Web GUIs: 
> http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ#Q:_Is_there_a_GUI_for_configuring_FreeSWITCH.3F
> Queues: mod_fifo or mod_callcenter
>

[...]

-- 
Nestor A. Diaz
Ingeniero de Sistemas
Tel. +57 1-485-3020 x 211
Cel. +57 316-227-3593
Tel. SIP: sip:211 at tiendalinux.com
Email/MSN: nestor at tiendalinux.com
http://www.tiendalinux.com/
Bogota, Colombia

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110809/3ed77308/attachment.html 

From nestor at tiendalinux.com  Wed Aug 10 19:00:55 2011
From: nestor at tiendalinux.com (Nestor A Diaz)
Date: Wed, 10 Aug 2011 10:00:55 -0500
Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
In-Reply-To: <CAFiqYu=092K=O5xExqfmW2ZvBeWrxsRC_YAoNjkXSbMhe_zMEQ@mail.gmail.com>
References: <4E4164C0.8030507@tiendalinux.com>
	<CAFiqYu=092K=O5xExqfmW2ZvBeWrxsRC_YAoNjkXSbMhe_zMEQ@mail.gmail.com>
Message-ID: <4E429D27.7040605@tiendalinux.com>

On 08/10/2011 04:51 AM, Steven Ayre wrote:
> As with Asterisk there's no official GUI, but there are several open 
> source projects providing one. Or you can build your own. A few 
> include blue.box (2800hz project) and FusionPBX.
> http://wiki.freeswitch.org/wiki/Freeswitch_Gui

Hi, i know there is always the possibility to build my own, but that's 
not the point, i prefer something ready to use, i neither use those kind 
of GUIs, those are for my customers, as i always prefer working on a 
command line and i don't have plans to change my mind regarding this.

> As for G729 which you mentioned... do NOT use the Intel IPP codec. It 
> is ILLEGAL unless you have purchased a valid licence for it, which is 
> extremely unlikely. You can support it using a hardware transcoding 
> card (Sangoma), mod_com_g729 (version licensed by FreeSWITCH) and 
> mod_g729 (which is passthrough only, no transcoding but fine for 
> bridging calls).

I know the use of the free g.729 is illegal, but if you want to test it 
without going to production i don't see the reason why i can't use it on 
my batcave.

> chan_mobile's closest match is probably mod_gsmopen. I believe it uses 
> a cable rather than bluetooth though, and is faily new so probably 
> 'experimental'.

Comming from the asterisk world i understand experimental is the same as 
not working, beta = maybe, and first stable means some release behind 
for production environments.  Anyway chan_mobile is a hack module but 
works fine for one or two cell phones.

Anybody have used mod_gsmopen ? does this thing really works ? do you 
have to transfer the dial number tones for making a call ? it that's 
true i still prefer a sip gateway and a cell plant.

I would love to have something like asterisk's chan_sebi running under 
freeswitch (it never works for me on asterisk but the idea is nice, very 
nice)

> IAX2 is supported by mod_opal. As you noted though, it isn't as stable 
> as say mod_sofia.

discarded, i prefer to use asterisk in another machine for that and 
brigde the calls to asterisk via SIP. (see chan_mobiles comments for 
explanation :) )

> For fax handling check the T38 functionality provided by mod_spandsp.
>

T.38 seems good, anybody have been able to make it work with hylafax ?? 
is possible ?

Slds.

-- 
Nestor A. Diaz
Ingeniero de Sistemas
Tel. +57 1-485-3020 x 211
Cel. +57 316-227-3593
Tel. SIP: sip:211 at tiendalinux.com
Email/MSN: nestor at tiendalinux.com
http://www.tiendalinux.com/
Bogota, Colombia



From adrottenberg at gmail.com  Thu Aug 11 01:43:59 2011
From: adrottenberg at gmail.com (Duvid Rottenberg)
Date: Wed, 10 Aug 2011 21:43:59 +0000 (UTC)
Subject: [Freeswitch-users] Play wmv/wma streams into calls
Message-ID: <loom.20110810T232430-465@post.gmane.org>

Is there any way I can play windows media streams (.wma & .wmv) into a call? I 
am only looking for the audio stream in the .wmv files.
The playback application does not appear to support this format even with 
mod_shout enabled, are there any modules that do support this format?



From adrottenberg at gmail.com  Thu Aug 11 05:05:16 2011
From: adrottenberg at gmail.com (Duvid Rottenberg)
Date: Wed, 10 Aug 2011 21:05:16 -0400
Subject: [Freeswitch-users] Playback .wma/.wmv
In-Reply-To: <CAC4JBEBPhsU0h119nMh1ghq3GYEBKMUrKBNyo=bf8PDwQcZ7ZQ@mail.gmail.com>
References: <CAC4JBEBPhsU0h119nMh1ghq3GYEBKMUrKBNyo=bf8PDwQcZ7ZQ@mail.gmail.com>
Message-ID: <CAC4JBEDuBqK1KfsWG4rY1dd8ASk9zGzAgP76YqxbpJX9Dg-YmA@mail.gmail.com>

I would like to playback into a channel a windows media stream (mms
streaming .wma or the audio track of a .wmv), however it appears that the
playback command does not support this. I have loaded the mod_shout module
but it appear this also doesn't support this format. Is there any module
available that supports this format?

Thanks,
Duvid Rottenberg
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110810/4009bf26/attachment.html 

From msc at freeswitch.org  Thu Aug 11 07:08:25 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 10 Aug 2011 22:08:25 -0500
Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
In-Reply-To: <4E429D27.7040605@tiendalinux.com>
References: <4E4164C0.8030507@tiendalinux.com>
	<CAFiqYu=092K=O5xExqfmW2ZvBeWrxsRC_YAoNjkXSbMhe_zMEQ@mail.gmail.com>
	<4E429D27.7040605@tiendalinux.com>
Message-ID: <CAKzWOxXOXD-xPjoYSCBVg9gGgtZF33L+t3_+5xzMqt8XF1sDpA@mail.gmail.com>

>
>
> > As for G729 which you mentioned... do NOT use the Intel IPP codec. It
> > is ILLEGAL unless you have purchased a valid licence for it, which is
> > extremely unlikely. You can support it using a hardware transcoding
> > card (Sangoma), mod_com_g729 (version licensed by FreeSWITCH) and
> > mod_g729 (which is passthrough only, no transcoding but fine for
> > bridging calls).
>
> I know the use of the free g.729 is illegal, but if you want to test it
> without going to production i don't see the reason why i can't use it on
> my batcave.
>

Honestly, there is no reason to test this, even in your batcave. The
free/experimental codec is not nearly as stable and streamlined as the
actual mod_com_g729. You are much better off spending $10 on a license for
your test server and using the actual module that would be used in a
production environment.

-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110810/f413d6cc/attachment.html 

From xing2kin at yahoo.com  Thu Aug 11 10:59:37 2011
From: xing2kin at yahoo.com (king2kin)
Date: Wed, 10 Aug 2011 23:59:37 -0700 (PDT)
Subject: [Freeswitch-users] May FS module Application NOT return data like
	FS module API?
In-Reply-To: <CAFiqYu=092K=O5xExqfmW2ZvBeWrxsRC_YAoNjkXSbMhe_zMEQ@mail.gmail.com>
Message-ID: <1313045977.40216.YahooMailClassic@web39707.mail.mud.yahoo.com>

Hi folks,
?
I looked into a few modules (e.g. mod_db, mod_hash, mod_unimrcp, mod_skel?and mod_say_xx), and find out that unlike module api,? a module app seems not able to return any data. If so, how do we know operation result after we submit app request to FS? for example, 
?<action application="appx" data="run/my/data"/> how do I know the above app task is executed successfully or not?xk
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110810/6a9bc025/attachment.html 

From nazim.aghabayov at gmail.com  Thu Aug 11 11:00:04 2011
From: nazim.aghabayov at gmail.com (Nazim Aghabayov)
Date: Thu, 11 Aug 2011 12:00:04 +0500
Subject: [Freeswitch-users] Custom SIP Profile
In-Reply-To: <CAMXUpRy=ft+puH8T9YZabiffPMC=cZSJGV-adKEoa64f0=oBzQ@mail.gmail.com>
References: <CAMXUpRy=ft+puH8T9YZabiffPMC=cZSJGV-adKEoa64f0=oBzQ@mail.gmail.com>
Message-ID: <4E437DF4.2050108@gmail.com>

Hi,
There is a nice wiki entry on sofia sip profiles at 
http://wiki.freeswitch.org/wiki/Sofia . It should answer most of your 
questions regarding sip profiles.

Best Regards,
Nazim

On 08/11/2011 04:37 AM, HM Kias wrote:
> Hi All,
>
> I m trying to create a SIP profile equivalent to the below asterisk config,
> please advise. Domain&  outboundproxy are very important in the
> registration.
>
> register =>
> XXXXXXXX at sia-nas01ca146.srg.com.bs:yyyyy:XXXXXXXX at nas-sbc-01.srg.com.bs:5060/XXXXXXXX
> [XXXXXXXX]
> host=sia-nas01ca146.srg.com.bs
> outboundproxy=nas-sbc-01.srg.com.bs
> type=friend
> username=XXXXXXXX
> fromuser=XXXXXXXX
> fromdomain=sia-nas01ca146.srg.com.bs
> secret=yyyyy
> dtmfmode=rfc2833
> disallow=all
> allow=ulaw&gsm
> context=users
> usereqphone=yes
> canreinvite=no
> insecure=port,invite
>
> Thanks in advance.
>
>
> Regards,
>
>


From steveayre at gmail.com  Thu Aug 11 12:31:31 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 11 Aug 2011 09:31:31 +0100
Subject: [Freeswitch-users] May FS module Application NOT return data
 like FS module API?
In-Reply-To: <1313045977.40216.YahooMailClassic@web39707.mail.mud.yahoo.com>
References: <CAFiqYu=092K=O5xExqfmW2ZvBeWrxsRC_YAoNjkXSbMhe_zMEQ@mail.gmail.com>
	<1313045977.40216.YahooMailClassic@web39707.mail.mud.yahoo.com>
Message-ID: <CAFiqYukVG3S89OJ6VPRw+9wM3mMPukGS0ea8twDmp8DEipU=mg@mail.gmail.com>

That really depends on the app. If the module providing the app hasn't been
loaded you'll get an error in your log and it'll skip to the next app. For
apps that are loaded, they'll generally either do something or set channel
variables that indicate the result.

Check the wiki documentation on http://wiki.freeswitch.org/

-Steve


On 11 August 2011 07:59, king2kin <xing2kin at yahoo.com> wrote:

> Hi folks,
>
> I looked into a few modules (e.g. mod_db, mod_hash, mod_unimrcp,
> mod_skel and mod_say_xx), and find out that unlike module api,  a module app
> seems not able to return any data. If so, how do we know operation result
> after we submit app request to FS? for example,
>
>
> <action application="appx" data="run/my/data"/>
>
> how do I know the above app task is executed successfully or not?
>
> xk
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110811/136bf04a/attachment-0001.html 

From covici at ccs.covici.com  Thu Aug 11 14:20:52 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 11 Aug 2011 06:20:52 -0400
Subject: [Freeswitch-users] rewind and fast forward while playing a file
Message-ID: <16715.1313058052@ccs.covici.com>

Hi.  I would like very much to be able to rewind or fast forward by some
amount (10 or 20 seconds or whatever) while playing a file to a
channel.  The nearest I have come is while streaming a file I can get
dtmf events in the background, but what can I do once I have the dtmf?

Thanks in advance for any suggestions.

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From tculjaga at gmail.com  Thu Aug 11 14:23:27 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 11 Aug 2011 12:23:27 +0200
Subject: [Freeswitch-users] FS performance using ESL
Message-ID: <CAJoKKdU9VPzOS-C3jHCjG6Y0V-Jo_j7hqYwt6-Zh7ohb1pq3xQ@mail.gmail.com>

hello,

im wondering how much performance do we loose when using ESL instead of
running it via dialplan?


without ESL with a fine tuned FS and a short dialplan ( answer, playback
like 20 seconds file, hangup ) im able to service 75 CPS. On the same FS,
when i use ESL to answer the call, playback the same file and hangup, im not
able to run more than 2 CPS... this is a huge impact and i really can't
believe it.

I'm using event-socket outbound e.g.:

<action application="socket" data="127.0.0.1:8088 async full"/>

my extension looks like:

<extension name="ESL_C">
  <condition field="destination_number" expression="^(6666)$">
    <action application="socket" data="127.0.0.1:8088 async full"/>
    <action application="sleep" data="1000"/>
    <action application="hangup"/>
  </condition>
</extension>


im using testserver from lib/esl/ and i just removed the conference command
and added the playback one.... also i moved the esl_debug lvl to 0


anyhow, FS cannot run more than 2 CPS compared to 75 CPS when the playback
is done from the dialplan.


Please, can someone give me a clue on what is going on?
Maybe im doing something wrong?
how to get maximum FS performance using ESL ?



Regards,
Tihomir.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110811/cff9d658/attachment.html 

From avi at avimarcus.net  Thu Aug 11 14:29:30 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 11 Aug 2011 13:29:30 +0300
Subject: [Freeswitch-users] rewind and fast forward while playing a file
In-Reply-To: <16715.1313058052@ccs.covici.com>
References: <16715.1313058052@ccs.covici.com>
Message-ID: <CALNZuwmP_dNyNeqxrGqrbnnM9ET+DCti3-vpvP35bv23e-Be1A@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Mod_commands#uuid_fileman
has a seek option.

It's an API however, so you may need to use these instructions:
http://wiki.freeswitch.org/wiki/Mod_commands#From_the_Dialplan

There's a path from a week or two ago that lets you do that right from
bind-digit-action, but it might be simpler to put the action into an
extension in the feature context and bind-digi-action to execute_extension
to run it.

-Avi Marcus


On Thu, Aug 11, 2011 at 1:20 PM, <covici at ccs.covici.com> wrote:

> Hi.  I would like very much to be able to rewind or fast forward by some
> amount (10 or 20 seconds or whatever) while playing a file to a
> channel.  The nearest I have come is while streaming a file I can get
> dtmf events in the background, but what can I do once I have the dtmf?
>
> Thanks in advance for any suggestions.
>
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>         John Covici
>         covici at ccs.covici.com
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110811/3d70e5d7/attachment.html 

From tculjaga at gmail.com  Thu Aug 11 14:35:18 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 11 Aug 2011 12:35:18 +0200
Subject: [Freeswitch-users] Mod_rad_auth issue for FS working with
 FreeRadius server
In-Reply-To: <CAOTmbMx8_aDoB2Q9ocuVtuXoQf0hbTJ0=VL+RtaWiZQgAw=YTg@mail.gmail.com>
References: <CAOTmbMzvd5OHL8z=_8vu6Orp2_prGxaw1gmXYCM2=2+pgzs_WA@mail.gmail.com>
	<CAJoKKdX01Qc8ibs25PUELi8GrzMDNOrHb_bQvYdq3ZqA4XLy4w@mail.gmail.com>
	<CAOTmbMxL3pLOuw655fyT5VzAON_N0PrDN+-i-++=oZoYU6+Fmg@mail.gmail.com>
	<CAOTmbMyvyDA-CRpaMC=WRGVkiyTG=Rs+n4hyeQWdyTZpqysSzw@mail.gmail.com>
	<CAJoKKdXSwF02S9tBkorGwOjj57AtkRpU0c177qXASEpodKVk=w@mail.gmail.com>
	<CAOTmbMzu2qDYh2T24JSpCYn1xH2J4iGiK7UkUB4Kq=-KdcDWGQ@mail.gmail.com>
	<CAJoKKdU66jK3f+mOEV90Y3zrmzpTSdWnwGsWimZxir42khiJ-w@mail.gmail.com>
	<CAOTmbMxW=EDr3-Ayo2R_MsCCTUZ4PnDkkRUuR2FisH9tK5xkWA@mail.gmail.com>
	<CAJoKKdVSkQOV3Kou_ro5AQZTyaqH6r0WRE-hZdrA=ye8ARE_vQ@mail.gmail.com>
	<CAOTmbMyg5_q-pFFtpzOooUcSZ8Gkb38y4sk1pYeJLYqguA=Jbg@mail.gmail.com>
	<CAJoKKdWqjt46Gvbj=4Tz-apQb+EV2UHJTsRBTE_bJZ95ogo2DA@mail.gmail.com>
	<CAOTmbMx8_aDoB2Q9ocuVtuXoQf0hbTJ0=VL+RtaWiZQgAw=YTg@mail.gmail.com>
Message-ID: <CAJoKKdXW+vrYYMMM163XFewWyzcHW14MLkyTr53oaVnKP4-9jQ@mail.gmail.com>

hello,

the example down below is just an example. In the real application you will
be using channel variables instead of direct input.
anyhow everything depends of what is your application intended for and how
you would like to behave.

Right now, you cannot authorize registrations as there is no event handler
built into the module. Its on the roadmap but not gonna happen in next few
weeks.

what you can do is to authorize calls (INVITEs) by triggering the
application within the dialplan. Also, FS extensions have their own ANI and
you can authorize by ANI. If this is not enough, you can try to fetch the
calling user password from the database and populate a session variable....
than use this variable to trigger radius authorization.

Anyhow, i think this is quite easy to do ... if you don't manage to do it on
your own, drop me an e-mail and i can help ya.


Cheers,
Thiomir.



On Tue, Aug 9, 2011 at 2:58 PM, fieldpeak <fieldpeak at gmail.com> wrote:

> Hi Tihomir,
>
> As my understanding, when using mod_rad_auth, we have to send both username
> and password to FreeRadius, like the example in wiki below (marked in
> yellow), the example is for a fixed password, however in real world, we have
> to dynamically inject the password as per user on-the-fly, e.g. user 1001 's
> password is 1234, user 1002's password is 2345 etc. in other word, we have
> to dynamically get the specific user's password and inject to the dial plan.
> Can you please advise how we should write the dial plan for the real case?
> Thanks in avdvance.
>
> P.S. What I'm concerning are both REGISTERATON and INVITE...how can we do
> the auth by Freeradius...
>
> <extension name="unitest_rad-ANI-balance">
>     <condition field="destination_number" expression="^602$">
>       <action application="log" data="INFO  PRIJE RAD_AUTH "/>
>
>       <action inline="true" application="set" data="CALLID=h323-conf-id=${uuid}"/>
>       <action inline="true" application="set" data="SERVICENUM=h323-prompt-id=${destination_number}"/>
>       <action inline="true" application="set" data="TRANSACTIONID=h323-ivr-out=transactionID:1234"/>
>  <!--      <action inline="true" application="set" data="CALLINGNUMBER=${caller_id_number}"/> -->
>       <action inline="true" application="set" data="CALLINGNUMBER=38516060333"/>
>       <action inline="true" application="set" data="USERNAME=38516060333"/>
>  <!--      <action inline="true" application="set" data="USERNAME=209354"/> -->
>       <action inline="true" application="set" data="PASSWD=003282"/>
>       <action inline="true" application="set" data="DIALED_NUMBER=16094191500"/>
>
>
>
> Regards,
> Charles
>
>
> 2011/8/9 Tihomir Culjaga <tculjaga at gmail.com>
>
>> im glad it works :=)
>>
>> T.
>>
>>
>> On Mon, Aug 8, 2011 at 8:18 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>>
>>> Hi Tihomir,
>>>
>>> The issue has been resolved by correcting the client secrect, appreciated
>>> very much for your kindly help!
>>>
>>> Regards,
>>> Charles
>>>
>>> 2011/8/7 Tihomir Culjaga <tculjaga at gmail.com>
>>>
>>>> are u sure you are using the correct secret on both client and server ?
>>>>
>>>>
>>>> On Fri, Aug 5, 2011 at 10:12 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>>>>
>>>>> Hi Tihomir,
>>>>>
>>>>> Thanks for your advise, i've added below to rad_auth.conf.xml (vsas
>>>>> section), as well as tried auth-type to 0(local) and 1(system), however, the
>>>>> issue still exist.
>>>>>
>>>>>
>>>>> <param name="NAS-Port-Type" id="61" value="0" pec="0" expr="0"
>>>>> direction="in"/>
>>>>> <param name="Login-User" id="6" value="1" pec="0" expr="0"
>>>>> direction="in"/>
>>>>> <param name="Auth-Type" id="1000" value="0" pec="0" expr="0"
>>>>> direction="in"/>
>>>>>
>>>>> FreeRadius output:
>>>>>
>>>>> Found Auth-Type = PAP
>>>>> # Executing group from file /usr/local/etc/raddb/sites-enabled/default
>>>>> +- entering group PAP {...}
>>>>> [pap] login attempt with password "Q?????? ??????p???F?+??a"
>>>>> [pap] Using clear text password "1111"
>>>>> [pap] Passwords don't match
>>>>> ++[pap] returns reject
>>>>> Failed to authenticate the user.
>>>>>   WARNING: Unprintable characters in the password.        Double-check the shared secret on the server and the NAS!
>>>>>
>>>>> Using Post-Auth-Type Reject
>>>>> # Executing group from file /usr/local/etc/raddb/sites-enabled/default
>>>>> +- entering group REJECT {...}
>>>>> [attr_filter.access_reject]     expand: %{User-Name} -> 1001
>>>>>  attr_filter: Matched entry DEFAULT at line 11
>>>>> ++[attr_filter.access_reject] returns updated
>>>>> Delaying reject of request 38 for 1 seconds
>>>>>
>>>>> Regards,
>>>>> Charles
>>>>>
>>>>>
>>>>> 2011/8/5 Tihomir Culjaga <tculjaga at gmail.com>
>>>>>
>>>>>> add to rad_auth.conf.xml
>>>>>>
>>>>>> <param name="NAS-Port-Type" id="61" value="0" pec="0" expr="0"
>>>>>> direction="in"/>
>>>>>> <param name="Login-User" id="6" value="1" pec="0" expr="0"
>>>>>> direction="in"/>
>>>>>>
>>>>>>
>>>>>>
>>>>>> as for Auth Type im not sure if you need it ... this is up to your
>>>>>> server.
>>>>>> According to dictionary file you need to set it as follows:
>>>>>>
>>>>>> <param name="Auth-Type" id="1000" value="?" pec="0" expr="0"
>>>>>> direction="in"/>
>>>>>>
>>>>>> the value (set as ?) is one of the folowing. Again, not sure what is
>>>>>> required by your server.
>>>>>>
>>>>>> VALUE           Auth-Type               Local                   0
>>>>>> VALUE           Auth-Type               System                  1
>>>>>> VALUE           Auth-Type               SecurID                 2
>>>>>> VALUE           Auth-Type               Crypt-Local             3
>>>>>> VALUE           Auth-Type               Reject                  4
>>>>>>
>>>>>> #
>>>>>> #       Cistron extensions
>>>>>> #
>>>>>> VALUE           Auth-Type               Pam                     253
>>>>>> VALUE           Auth-Type               Accept                  254
>>>>>>
>>>>>>
>>>>>>
>>>>>> regards,
>>>>>> Tihomir.
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Wed, Aug 3, 2011 at 6:32 AM, fieldpeak <fieldpeak at gmail.com>wrote:
>>>>>>
>>>>>>> Hi Tihomir,
>>>>>>>
>>>>>>> Sorry, i missed your mail in gmail before, just now saw it, and after
>>>>>>> using your dictionary.all, the dictionary issue was resolved, very
>>>>>>> appreciated for your kindly help! however, it did not fully functional yet,
>>>>>>>
>>>>>>> Attached are configuration files that i used, when i dial 601 to
>>>>>>> trigger to auth, the freeradius server shows log below, the supecious log is
>>>>>>> the value User-Password, it should be '1111' that i've set in the mysql db
>>>>>>> of freeradisu server for the user 1001 .
>>>>>>>
>>>>>>> i searched in google, for "known good" password issue, i suggest
>>>>>>> change user-password to cleartext-password, however, i did not find where it
>>>>>>> is.
>>>>>>> and also the Auth-Type, where to configure it...
>>>>>>>
>>>>>>> Freeradius server log:
>>>>>>>
>>>>>>> rad_recv: Access-Request packet from host 127.0.0.1 port 52684,
>>>>>>> id=49, length=111
>>>>>>>         User-Name = "1001"
>>>>>>>         User-Password = "?\210\365@\263\t\306\343\243iT?\311C\t\002"
>>>>>>>         Called-Station-Id = "888"
>>>>>>>         h323-conf-id = "749d2b5a-16ad-48e4-af58-24011949d1b5"
>>>>>>>         Calling-Station-Id = "1001"
>>>>>>>         NAS-Port = 0
>>>>>>>         NAS-IP-Address = 127.0.0.1
>>>>>>> # Executing section authorize from file
>>>>>>> /usr/local/etc/raddb/sites-enabled/default
>>>>>>> +- entering group authorize {...}
>>>>>>> ++[preprocess] returns ok
>>>>>>> [auth_log]      expand:
>>>>>>> /usr/local/var/log/radius/radacct/%{Client-IP-Address}/auth-detail-%Y%m%d ->
>>>>>>> /usr/local/var/log/radius/radacct/127.0.0.1/auth-detail-20110803
>>>>>>> [auth_log]
>>>>>>> /usr/local/var/log/radius/radacct/%{Client-IP-Address}/auth-detail-%Y%m%d
>>>>>>> expands to /usr/local/var/log/radius/radacct/
>>>>>>> 127.0.0.1/auth-detail-20110803
>>>>>>> [auth_log]      expand: %t -> Wed Aug  3 12:06:33 2011
>>>>>>> ++[auth_log] returns ok
>>>>>>> ++[chap] returns noop
>>>>>>> ++[mschap] returns noop
>>>>>>> ++[digest] returns noop
>>>>>>> [suffix] No '@' in User-Name = "1001", looking up realm NULL
>>>>>>> [suffix] No such realm "NULL"
>>>>>>> ++[suffix] returns noop
>>>>>>> [eap] No EAP-Message, not doing EAP
>>>>>>> ++[eap] returns noop
>>>>>>> ++[unix] returns notfound
>>>>>>> ++[files] returns noop
>>>>>>> [sql]   expand: %{User-Name} -> 1001
>>>>>>> [sql] sql_set_user escaped user --> '1001'
>>>>>>> rlm_sql (sql): Reserving sql socket id: 4
>>>>>>> [sql]   expand: SELECT id, username, attribute, value, op
>>>>>>> FROM radcheck           WHERE username = '%{SQL-User-Name}'           ORDER
>>>>>>> BY id -> SELECT id, username, attribute, value, op           FROM
>>>>>>> radcheck           WHERE username = '1001'           ORDER BY id
>>>>>>> [sql]   expand: SELECT groupname           FROM
>>>>>>> radusergroup           WHERE username = '%{SQL-User-Name}'           ORDER
>>>>>>> BY priority -> SELECT groupname           FROM radusergroup           WHERE
>>>>>>> username = '1001'           ORDER BY priority
>>>>>>> rlm_sql (sql): Released sql socket id: 4
>>>>>>> [sql] User 1001 not found
>>>>>>> ++[sql] returns notfound
>>>>>>> ++[expiration] returns noop
>>>>>>> ++[logintime] returns noop
>>>>>>> [pap] WARNING! No "known good" password found for the user.
>>>>>>> Authentication may fail because of this.
>>>>>>> ++[pap] returns noop
>>>>>>> ERROR: No authenticate method (Auth-Type) found for the request:
>>>>>>> Rejecting the user
>>>>>>> Failed to authenticate the user.
>>>>>>>   WARNING: Unprintable characters in the password.
>>>>>>> Double-check the shared secret on the server and the NAS!
>>>>>>> Using Post-Auth-Type Reject
>>>>>>> # Executing group from file
>>>>>>> /usr/local/etc/raddb/sites-enabled/default
>>>>>>> +- entering group REJECT {...}
>>>>>>> [attr_filter.access_reject]     expand: %{User-Name} -> 1001
>>>>>>>  attr_filter: Matched entry DEFAULT at line 11
>>>>>>> ++[attr_filter.access_reject] returns updated
>>>>>>> Delaying reject of request 8 for 1 seconds
>>>>>>> Going to the next request
>>>>>>> Waking up in 0.9 seconds.
>>>>>>> Sending delayed reject for request 8
>>>>>>> Sending Access-Reject of id 49 to 127.0.0.1 port 52684
>>>>>>> Waking up in 4.9 seconds.
>>>>>>> Cleaning up request 8 ID 49 with timestamp +7674
>>>>>>> Ready to process requests.
>>>>>>> WARNING! No "known good" password found for the user
>>>>>>>
>>>>>>> Regards,
>>>>>>> Charles
>>>>>>>
>>>>>>>
>>>>>>> 2011/8/3 Tihomir Culjaga <tculjaga at gmail.com>
>>>>>>>
>>>>>>>> did u use the dictionary i have attached ?
>>>>>>>>
>>>>>>>>
>>>>>>>> On Tue, Aug 2, 2011 at 10:08 AM, fieldpeak <fieldpeak at gmail.com>wrote:
>>>>>>>>
>>>>>>>>> i tried change to 'h323-conf-id' to 'h323-call-origin' in
>>>>>>>>> 02_unitest_rad-ANI-auth.xml, rad_auth.conf.xml, however, it still prompt
>>>>>>>>> '[ERR] mod_rad_auth.c:428 Unknown attribute: key:h323-conf-id, not
>>>>>>>>> found in dictionary', so where the mod_rad_auth read out the 'h323-conf-id'?
>>>>>>>>> very very strange, which dictionary it was using...
>>>>>>>>>
>>>>>>>>> Regards,
>>>>>>>>> Charles
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> 2011/8/2 fieldpeak <fieldpeak at gmail.com>
>>>>>>>>>
>>>>>>>>>> Hi Tihomir,
>>>>>>>>>>
>>>>>>>>>> Finally the answer coming, i see the hope, thanks for your reply,
>>>>>>>>>> :)
>>>>>>>>>>
>>>>>>>>>> As your advise, i only use one attribute(h323-conf-id) in my
>>>>>>>>>> dialplan, and only one attribute(h323-conf-id) in rad_auth.conf.xml, and
>>>>>>>>>> using the attached dictionary (from ciso) which contains this attribute,
>>>>>>>>>> however, it still prompt 'unknown attribute', so i suspected if it was
>>>>>>>>>> reading /usr/local/etc/radiusclient/dictionary, so i copy the same
>>>>>>>>>> dictionary to /usr/local/freeswitch/radius/, it did not any help at all...
>>>>>>>>>> very strange...
>>>>>>>>>>
>>>>>>>>>> Log:
>>>>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>>> default_realm := .
>>>>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>>> radius_timeout := 3.
>>>>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>>> radius_retries := 2.
>>>>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>>> radius_deadtime := 0.
>>>>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set bindaddr
>>>>>>>>>> := *.
>>>>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:371 ... radius:
>>>>>>>>>> User-Name: 38516060333
>>>>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:380 ... radius:
>>>>>>>>>> User-Password: 003282
>>>>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:396 ... radius:
>>>>>>>>>> Called-station-Id: 16094191500
>>>>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:413 Handle
>>>>>>>>>> attribute: h323-conf-id
>>>>>>>>>> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:428 Unknown
>>>>>>>>>> attribute: key:h323-conf-id, not found in dictionary
>>>>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:538 abort
>>>>>>>>>> sending radius packet.
>>>>>>>>>> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:546 An error
>>>>>>>>>> occured during RADIUS Authentication(RC=-1)
>>>>>>>>>> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:702 An error
>>>>>>>>>> occured during radius authorization.
>>>>>>>>>>
>>>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO
>>>>>>>>>> AUTH_RESULT=)
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>   <extension name="unitest_rad-ANI-auth">
>>>>>>>>>>     <condition field="destination_number" expression="^601$">
>>>>>>>>>>       <!-- <action application="log" data="INFO  Before Auth "/>
>>>>>>>>>> -->
>>>>>>>>>>
>>>>>>>>>>       <action inline="true" application="set" data="CALLID=
>>>>>>>>>> h323-conf-id=${uuid}"/>
>>>>>>>>>>
>>>>>>>>>>       <action inline="true" application="set"
>>>>>>>>>> data="USERNAME=1001"/>
>>>>>>>>>>       <action inline="true" application="set" data="PASSWD=1111"/>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>       <action application="sleep" data="2000"/>
>>>>>>>>>>       <action application="auth_function" data="in
>>>>>>>>>> ${DIALED_NUMBER}, in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>
>>>>>>>>>>
>>>>>>>>>>     </condition>
>>>>>>>>>>   </extension>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> <configuration name="rad_auth.conf" description="radius
>>>>>>>>>> authentification module">
>>>>>>>>>>   <settings>
>>>>>>>>>>
>>>>>>>>>>   </settings>
>>>>>>>>>>
>>>>>>>>>>   <client>
>>>>>>>>>>     <param name="authserver" value="127.0.0.1:1812:gateway"/>
>>>>>>>>>>     <param name="dictionary"
>>>>>>>>>> value="/usr/local/etc/radiusclient/dictionary"/>
>>>>>>>>>>     <param name="seqfile" value="/var/run/radius.seq"/>
>>>>>>>>>>     <param name="mapfile"
>>>>>>>>>> value="/usr/local/etc/radiusclient/port-id-map"/>
>>>>>>>>>>     <param name="default_realm" value=""/>
>>>>>>>>>>     <param name="radius_timeout" value="3"/>
>>>>>>>>>>     <param name="radius_retries" value="2"/>
>>>>>>>>>>     <param name="radius_deadtime" value="0"/>
>>>>>>>>>>     <param name="bindaddr" value="*"/>
>>>>>>>>>>   </client>
>>>>>>>>>>
>>>>>>>>>>   <vsas>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>     <param name="h323-conf-id" id="24" value="CALLID" pec="9"
>>>>>>>>>> expr="1" direction="in"/>
>>>>>>>>>>
>>>>>>>>>>   </vsas>
>>>>>>>>>>  </configuration>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> 2011/8/2 Tihomir Culjaga <tculjaga at gmail.com>
>>>>>>>>>>
>>>>>>>>>>> hi,
>>>>>>>>>>>
>>>>>>>>>>> dictionary.all is just the name of a file containing all
>>>>>>>>>>> attributes i needed at that time.
>>>>>>>>>>>
>>>>>>>>>>> you can include other dictionaries by putting #INCLUDE <pathname>
>>>>>>>>>>> at the end of the dictionary file you reference in rad_auth.conf.xml.
>>>>>>>>>>> if the INCLUDE doesn't work, just append dictionary.cisco to your
>>>>>>>>>>> dictionary file... and make your own file.
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> check inline comments down below...
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> T.
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> On Sun, Jul 31, 2011 at 10:46 AM, fieldpeak <fieldpeak at gmail.com
>>>>>>>>>>> > wrote:
>>>>>>>>>>>
>>>>>>>>>>>> Hello Gurus,
>>>>>>>>>>>>
>>>>>>>>>>>> i met a issue when using
>>>>>>>>>>>> mod_rad_auth(http://wiki.freeswitch.org/wiki/Mod_rad_auth) to
>>>>>>>>>>>> works
>>>>>>>>>>>> with freeradius server+mysql for AAA, the details is below,
>>>>>>>>>>>> Could
>>>>>>>>>>>> anyone give any hints, Thanks in advance.
>>>>>>>>>>>>
>>>>>>>>>>>> i setup a dial plan "unitest_rad-ANI-auth" as wiki above,
>>>>>>>>>>>> however,
>>>>>>>>>>>> when i dialed 601 to trigger the dial plan, the console show
>>>>>>>>>>>> errors,
>>>>>>>>>>>> it looks "h323-conf-id" is not in the directory, then i tried to
>>>>>>>>>>>> add
>>>>>>>>>>>> this attribute to the dictionary, however, it does not help, in
>>>>>>>>>>>> the
>>>>>>>>>>>> wiki, it mentioned the rad_auth.conf.xml contains <param
>>>>>>>>>>>> name="dictionary"
>>>>>>>>>>>> value="/usr/local/etc/radiusclient/dictionary.all"/>, however i
>>>>>>>>>>>> did
>>>>>>>>>>>> not find the file "dictionary.all" at that directory, so i use
>>>>>>>>>>>> dictionary. BTW, the freeradius server + mysql works well.
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> i just appended the information needed into dictionary.all
>>>>>>>>>>> file... (vendor and attribute definition).
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> console errors:
>>>>>>>>>>>>
>>>>>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 auth_function(in ,
>>>>>>>>>>>> in
>>>>>>>>>>>> 38516060333, in 003282, out AUTH_RESULT)
>>>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:301 allocate
>>>>>>>>>>>> initial
>>>>>>>>>>>> structure.
>>>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:313 initialzed
>>>>>>>>>>>> configuration.
>>>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>>>>> authserver
>>>>>>>>>>>> := 127.0.0.1:1812:gateway.
>>>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>>>>> dictionary
>>>>>>>>>>>> := /usr/local/etc/radiusclient/dictionary.
>>>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>>>>> seqfile :=
>>>>>>>>>>>> /var/run/radius.seq.
>>>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>>>>> mapfile :=
>>>>>>>>>>>> /usr/local/etc/radiusclient/port-id-map.
>>>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>>>>> default_realm := .
>>>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>>>>> radius_timeout := 3.
>>>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>>>>> radius_retries := 2.
>>>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>>>>> radius_deadtime := 0.
>>>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>>>>> bindaddr := *.
>>>>>>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:371 ...
>>>>>>>>>>>> radius:
>>>>>>>>>>>> User-Name: 38516060333
>>>>>>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:380 ...
>>>>>>>>>>>> radius:
>>>>>>>>>>>> User-Password: 003282
>>>>>>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:391 ...
>>>>>>>>>>>> radius:
>>>>>>>>>>>> Called-station-Id is empty, ignoring...
>>>>>>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:413 Handle
>>>>>>>>>>>> attribute: h323-conf-id
>>>>>>>>>>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:428 Unknown
>>>>>>>>>>>> attribute:
>>>>>>>>>>>> key:h323-conf-id, not found in dictionary
>>>>>>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:538 abort
>>>>>>>>>>>> sending
>>>>>>>>>>>> radius packet.
>>>>>>>>>>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:546 An error
>>>>>>>>>>>> occured
>>>>>>>>>>>> during RADIUS Authentication(RC=-1)
>>>>>>>>>>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:702 An error
>>>>>>>>>>>> occured
>>>>>>>>>>>> during radius authorization.
>>>>>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO
>>>>>>>>>>>>  AUTH_RESULT=)
>>>>>>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202
>>>>>>>>>>>>  AUTH_RESULT=
>>>>>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO
>>>>>>>>>>>>  billing_model=)
>>>>>>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202
>>>>>>>>>>>>  billing_model=
>>>>>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO
>>>>>>>>>>>>  credit_amount=)
>>>>>>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202
>>>>>>>>>>>>  credit_amount=
>>>>>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO
>>>>>>>>>>>>  currency=)
>>>>>>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  currency=
>>>>>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO
>>>>>>>>>>>>  preffered_lang=)
>>>>>>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202
>>>>>>>>>>>>  preffered_lang=
>>>>>>>>>>>>
>>>>>>>>>>>> added below in the
>>>>>>>>>>>> dictionary(/usr/local/etc/radiusclient/dictionary):
>>>>>>>>>>>>
>>>>>>>>>>>> ATTRIBUTE       h323-conf-id            1008    string
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> you need the vendor definition as well
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> dial plan:
>>>>>>>>>>>> <extension name="unitest_rad-ANI-auth">
>>>>>>>>>>>>    <condition field="destination_number" expression="^601$">
>>>>>>>>>>>>      <action application="log" data="INFO  Before Auth "/>
>>>>>>>>>>>>
>>>>>>>>>>>>      <action inline="true" application="set"
>>>>>>>>>>>> data="CALLID=h323-conf-id=${uuid}"/>
>>>>>>>>>>>>      <action inline="true" application="set"
>>>>>>>>>>>> data="SERVICENUM=h323-prompt-id=${destination_number}"/>
>>>>>>>>>>>>      <action inline="true" application="set"
>>>>>>>>>>>> data="TRANSACTIONID=h323-ivr-out=transactionID:1234"/>
>>>>>>>>>>>>  <!--      <action inline="true" application="set"
>>>>>>>>>>>> data="CALLINGNUMBER=${caller_id_number}"/> -->
>>>>>>>>>>>>      <action inline="true" application="set"
>>>>>>>>>>>> data="CALLINGNUMBER=38516060333"/>
>>>>>>>>>>>>      <action inline="true" application="set"
>>>>>>>>>>>> data="USERNAME=38516060333"/>
>>>>>>>>>>>>  <!--      <action inline="true" application="set"
>>>>>>>>>>>> data="USERNAME=209354"/> -->
>>>>>>>>>>>>      <action inline="true" application="set"
>>>>>>>>>>>> data="PASSWD=003282"/>
>>>>>>>>>>>>  <!--      <action inline="true" application="set"
>>>>>>>>>>>> data="DIALED_NUMBER=16094191500"/>  -->
>>>>>>>>>>>>
>>>>>>>>>>>>      <action application="sleep" data="2000"/>
>>>>>>>>>>>>      <action application="auth_function" data="in
>>>>>>>>>>>> ${DIALED_NUMBER},
>>>>>>>>>>>> in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>>>>>  AUTH_RESULT=${AUTH_RESULT}"/>
>>>>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>>>>>  billing_model=${billing_model}"/>
>>>>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>>>>>  credit_amount=${credit_amount}"/>
>>>>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>>>>>  currency=${currency}"/>
>>>>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>>>>>  preffered_lang=${preffered_lang}"/>
>>>>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>>>>>  credit_time=${credit_time}"/>
>>>>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>>>>> h323_ivr_duration=${h323_ivr_duration}"/>
>>>>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>>>>>  return_code=${return_code}"/>
>>>>>>>>>>>>      <!-- <action application="execute_extension" data="AUTH XML
>>>>>>>>>>>> default"/> -->
>>>>>>>>>>>>    </condition>
>>>>>>>>>>>>  </extension>
>>>>>>>>>>>>
>>>>>>>>>>>>  radius_cdr.conf.xml:
>>>>>>>>>>>>  <configuration name="radius_cdr.conf" description="RADIUS CDR
>>>>>>>>>>>> Configuration">
>>>>>>>>>>>>
>>>>>>>>>>>>        <settings>
>>>>>>>>>>>>
>>>>>>>>>>>>                <!-- location of the radius dictionary files -->
>>>>>>>>>>>>
>>>>>>>>>>>>                <param name="dictionary"
>>>>>>>>>>>> value="/usr/local/freeswitch/conf/radius/dictionary"/>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>> your dictionary file need to contain all the attributes you are
>>>>>>>>>>> trying to use or to include other dictionaries (In this case
>>>>>>>>>>> dictionary.cisco) from the dictionary file you are referencing here.
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>>                <!-- number of retries for each server -->
>>>>>>>>>>>>
>>>>>>>>>>>>                <param name="radius_retries" value="3"/>
>>>>>>>>>>>>
>>>>>>>>>>>>                <!-- number of seconds to wait between retries
>>>>>>>>>>>> -->
>>>>>>>>>>>>
>>>>>>>>>>>>                <param name="radius_timeout" value="5"/>
>>>>>>>>>>>>
>>>>>>>>>>>>                <!-- accounting servers, up to 8 allowed -->
>>>>>>>>>>>>
>>>>>>>>>>>>                <!-- value is "host:port:secret", port is
>>>>>>>>>>>> optional -->
>>>>>>>>>>>>
>>>>>>>>>>>>                <!-- use IP ADDRESSES, not hostnames -->
>>>>>>>>>>>>
>>>>>>>>>>>>                <param name="acct_server" value="127.0.0.1:1813
>>>>>>>>>>>> :testing123"/>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>        </settings>
>>>>>>>>>>>>
>>>>>>>>>>>> </configuration>
>>>>>>>>>>>>
>>>>>>>>>>>>  the FS version:
>>>>>>>>>>>>  FreeSWITCH Version 1.0.head (git-492bc6b 2011-07-23 12-53-04
>>>>>>>>>>>> -0400)
>>>>>>>>>>>>
>>>>>>>>>>>>  Regards,
>>>>>>>>>>>>  Charles
>>>>>>>>>>>>
>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>>>
>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>
>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>>
>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110811/f93f4b04/attachment-0001.html 

From avi at avimarcus.net  Thu Aug 11 15:28:27 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 11 Aug 2011 14:28:27 +0300
Subject: [Freeswitch-users] odbc basic_calls
Message-ID: <CALNZuwk20r8=rmu9rOOd-aChOS6XOGVm7mxKNhd1-DnC+id7Hg@mail.gmail.com>

I just upgraded FS since.. 7 weeks ago I think. Now: FreeSWITCH Version
1.0.head (git-9d98d49 2011-08-10 08-38-55 -0500)
While testing the new build clean, when I hit F4 for show calls, I now get:

freeswitch at internal> 2011-08-11 14:24:04.574797 [ERR]
switch_core_sqldb.c:825 ERR: [select * from basic_calls where
hostname='sip2' order by call_created_epoch]
[STATE: 42S02 CODE 1146 ERROR: [unixODBC][MySQL][ODBC 3.51
Driver][mysqld-5.1.41-3ubuntu12.10-log]Table 'freeswitch.basic_calls'
doesn't exist

Aren't the core odbc tables supposed to be auto-created?

-Avi
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110811/aa69a08f/attachment.html 

From covici at ccs.covici.com  Thu Aug 11 17:32:55 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 11 Aug 2011 09:32:55 -0400
Subject: [Freeswitch-users] rewind and fast forward while playing a file
In-Reply-To: <CALNZuwmP_dNyNeqxrGqrbnnM9ET+DCti3-vpvP35bv23e-Be1A@mail.gmail.com>
References: <16715.1313058052@ccs.covici.com>
	<CALNZuwmP_dNyNeqxrGqrbnnM9ET+DCti3-vpvP35bv23e-Be1A@mail.gmail.com>
Message-ID: <30657.1313069575@ccs.covici.com>

Thanks much -- I want to put this in a perl program, so we shall see how
that goes, or maybe  just execute an extension.

Avi Marcus <avi at avimarcus.net> wrote:

> http://wiki.freeswitch.org/wiki/Mod_commands#uuid_fileman
> has a seek option.
> 
> It's an API however, so you may need to use these instructions:
> http://wiki.freeswitch.org/wiki/Mod_commands#From_the_Dialplan
> 
> There's a path from a week or two ago that lets you do that right from
> bind-digit-action, but it might be simpler to put the action into an
> extension in the feature context and bind-digi-action to execute_extension
> to run it.
> 
> -Avi Marcus
> 
> 
> On Thu, Aug 11, 2011 at 1:20 PM, <covici at ccs.covici.com> wrote:
> 
> > Hi.  I would like very much to be able to rewind or fast forward by some
> > amount (10 or 20 seconds or whatever) while playing a file to a
> > channel.  The nearest I have come is while streaming a file I can get
> > dtmf events in the background, but what can I do once I have the dtmf?
> >
> > Thanks in advance for any suggestions.
> >
> > --
> > Your life is like a penny.  You're going to lose it.  The question is:
> > How do
> > you spend it?
> >
> >         John Covici
> >         covici at ccs.covici.com
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From anthony.minessale at gmail.com  Thu Aug 11 17:33:08 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 11 Aug 2011 08:33:08 -0500
Subject: [Freeswitch-users] odbc basic_calls
In-Reply-To: <CALNZuwk20r8=rmu9rOOd-aChOS6XOGVm7mxKNhd1-DnC+id7Hg@mail.gmail.com>
References: <CALNZuwk20r8=rmu9rOOd-aChOS6XOGVm7mxKNhd1-DnC+id7Hg@mail.gmail.com>
Message-ID: <CAKbxfG-8E=5Do_mLf62tokgL5eN8=LfH954tUNRawcNCtmMTLA@mail.gmail.com>

its a view that should be created? perhaps there is an error on
startup creating the view in mysuckwell?


On Thu, Aug 11, 2011 at 6:28 AM, Avi Marcus <avi at avimarcus.net> wrote:
> I just upgraded FS since.. 7 weeks ago I think. Now:?FreeSWITCH Version
> 1.0.head (git-9d98d49 2011-08-10 08-38-55 -0500)
> While testing the new build clean, when I hit F4 for show calls, I now get:
> freeswitch at internal> 2011-08-11 14:24:04.574797 [ERR]
> switch_core_sqldb.c:825 ERR: [select * from basic_calls where
> hostname='sip2' order by call_created_epoch]
> [STATE: 42S02 CODE 1146 ERROR: [unixODBC][MySQL][ODBC 3.51
> Driver][mysqld-5.1.41-3ubuntu12.10-log]Table 'freeswitch.basic_calls'
> doesn't exist
> Aren't the core odbc tables supposed to be auto-created?
> -Avi
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From avi at avimarcus.net  Thu Aug 11 17:48:03 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 11 Aug 2011 16:48:03 +0300
Subject: [Freeswitch-users] odbc basic_calls
In-Reply-To: <CAKbxfG-8E=5Do_mLf62tokgL5eN8=LfH954tUNRawcNCtmMTLA@mail.gmail.com>
References: <CALNZuwk20r8=rmu9rOOd-aChOS6XOGVm7mxKNhd1-DnC+id7Hg@mail.gmail.com>
	<CAKbxfG-8E=5Do_mLf62tokgL5eN8=LfH954tUNRawcNCtmMTLA@mail.gmail.com>
Message-ID: <CALNZuwkwgFmdmWxYMzkzGeW8u-TiRg2SbeJ1zFpOfxEPiyS2xg@mail.gmail.com>

I don't see anything in the startup log about it checking the tables:
http://pastebin.freeswitch.org/17013
This happens when I do "show calls". I know you changed something for show
calls to even show 1 legged IVRs since my last update, but not having looked
at the code, I don't see how that would be related.
I haven't heard of this basic_calls table before today.

-Avi Marcus

On Thu, Aug 11, 2011 at 4:33 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> its a view that should be created? perhaps there is an error on
> startup creating the view in mysuckwell?
>
>
> On Thu, Aug 11, 2011 at 6:28 AM, Avi Marcus <avi at avimarcus.net> wrote:
> > I just upgraded FS since.. 7 weeks ago I think. Now: FreeSWITCH Version
> > 1.0.head (git-9d98d49 2011-08-10 08-38-55 -0500)
> > While testing the new build clean, when I hit F4 for show calls, I now
> get:
> > freeswitch at internal> 2011-08-11 14:24:04.574797 [ERR]
> > switch_core_sqldb.c:825 ERR: [select * from basic_calls where
> > hostname='sip2' order by call_created_epoch]
> > [STATE: 42S02 CODE 1146 ERROR: [unixODBC][MySQL][ODBC 3.51
> > Driver][mysqld-5.1.41-3ubuntu12.10-log]Table 'freeswitch.basic_calls'
> > doesn't exist
> > Aren't the core odbc tables supposed to be auto-created?
> > -Avi
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110811/e44d44f0/attachment.html 

From jeff at jefflenk.com  Thu Aug 11 18:05:06 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 11 Aug 2011 07:05:06 -0700 (PDT)
Subject: [Freeswitch-users] freeswitch.org is down
In-Reply-To: <003601cc57cd$a1348e30$e39daa90$@com>
References: <33095823FD21DF429B481B5163264B7950FF060640@VMBX102.ihostexchange.net>
	<CAO47Hpf4J1Az0WCY8TtSRW-wiGHzcifBH+XoyzkdESbaO8EoSQ@mail.gmail.com>
	<003601cc57cd$a1348e30$e39daa90$@com>
Message-ID: <1313071506800-6676501.post@n2.nabble.com>

Jira and Fisheye are now up.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/freeswitch-org-is-down-tp6674580p6676501.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From jeff at jefflenk.com  Thu Aug 11 18:09:07 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 11 Aug 2011 07:09:07 -0700 (PDT)
Subject: [Freeswitch-users] FS terminated maybe due to memory leak?
In-Reply-To: <4E42D5FB.2020609@gosilverplus.com>
References: <4E42D5FB.2020609@gosilverplus.com>
Message-ID: <1313071747872-6676511.post@n2.nabble.com>

you will need to provide some more details here:
debug log of failed call
sofia global siptrace on - log

provide a link to pastebin.freeswitch.org

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-terminated-maybe-due-to-memory-leak-tp6674632p6676511.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From yungwei at resolvity.com  Thu Aug 11 18:10:02 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Thu, 11 Aug 2011 10:10:02 -0400
Subject: [Freeswitch-users] Making changes to dialplans and users using API
	commands
Message-ID: <33095823FD21DF429B481B5163264B7950FF06077F@VMBX102.ihostexchange.net>

Hi,

I am wondering if Freeswitch API commands allow us to make changes to dialplans and users. 
If not, is there a way to do that?
Thanks.


From avi at avimarcus.net  Thu Aug 11 18:16:53 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 11 Aug 2011 17:16:53 +0300
Subject: [Freeswitch-users] Making changes to dialplans and users using
 API commands
In-Reply-To: <33095823FD21DF429B481B5163264B7950FF06077F@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B7950FF06077F@VMBX102.ihostexchange.net>
Message-ID: <CALNZuw=awUHxOT=V4WrBxwJ79ci45s4=a3P-9OfACdWEkxaTGw@mail.gmail.com>

Each reloadxml reloads them from the actual files.. or each request is
pulled from mod_xml_curl. So even if such functionality could be made, I
don't see practically how it would work.

One way, however, is to make calls to SQL or to db or hash or global
variable to see what to do. You can then set those from the dialplan in some
manner (or with a lua script). Just beware, hash and variables are non
persistent.
-Avi Marcus

718-989-9485 (USA)
054-844-3271 (Israel Kosher)
077-228-5055 (Israel Landline)
020-3519-3606 (UK)



On Thu, Aug 11, 2011 at 5:10 PM, Yungwei Chen <yungwei at resolvity.com> wrote:

> Hi,
>
> I am wondering if Freeswitch API commands allow us to make changes to
> dialplans and users.
> If not, is there a way to do that?
> Thanks.
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110811/2297e3af/attachment-0001.html 

From Nabble at slickdeals.endjunk.com  Thu Aug 11 18:30:15 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 11 Aug 2011 07:30:15 -0700 (PDT)
Subject: [Freeswitch-users] odbc basic_calls
In-Reply-To: <CALNZuwk20r8=rmu9rOOd-aChOS6XOGVm7mxKNhd1-DnC+id7Hg@mail.gmail.com>
References: <CALNZuwk20r8=rmu9rOOd-aChOS6XOGVm7mxKNhd1-DnC+id7Hg@mail.gmail.com>
Message-ID: <1313073015901-6676584.post@n2.nabble.com>

Avi,

I also just upgraded FS since .. 03/26/2011 ago. Now, my Seagate DockStar is
hosting the same FS version you have. Notice that I don't know much about
using ODBC, but my current FS version has been compiled with the
/--enable-core-odbc-support/ option. So, I haven't done anything to create
any SQL database to use with my FS. However, then I pressed F4 key while I
was on a call to ext. 3000 (conference call), it showed me with the call in
progress and didn't show me the error message as you posted above. I don't
know if having no SQL database file has anything to do with it.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/odbc-basic-calls-tp6676013p6676584.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From covici at ccs.covici.com  Thu Aug 11 18:44:39 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 11 Aug 2011 10:44:39 -0400
Subject: [Freeswitch-users] freeswitch.org is down
In-Reply-To: <1313071506800-6676501.post@n2.nabble.com>
References: <33095823FD21DF429B481B5163264B7950FF060640@VMBX102.ihostexchange.net>
	<CAO47Hpf4J1Az0WCY8TtSRW-wiGHzcifBH+XoyzkdESbaO8EoSQ@mail.gmail.com>
	<003601cc57cd$a1348e30$e39daa90$@com>
	<1313071506800-6676501.post@n2.nabble.com>
Message-ID: <8041.1313073879@ccs.covici.com>

But the wiki seems to be down.

Jeff Lenk <jeff at jefflenk.com> wrote:

> Jira and Fisheye are now up.
> 
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/freeswitch-org-is-down-tp6674580p6676501.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From covici at ccs.covici.com  Thu Aug 11 18:53:09 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 11 Aug 2011 10:53:09 -0400
Subject: [Freeswitch-users] freeswitch.org is down
In-Reply-To: <8041.1313073879@ccs.covici.com>
References: <33095823FD21DF429B481B5163264B7950FF060640@VMBX102.ihostexchange.net>
	<CAO47Hpf4J1Az0WCY8TtSRW-wiGHzcifBH+XoyzkdESbaO8EoSQ@mail.gmail.com>
	<003601cc57cd$a1348e30$e39daa90$@com>
	<1313071506800-6676501.post@n2.nabble.com>
	<8041.1313073879@ccs.covici.com>
Message-ID: <9272.1313074389@ccs.covici.com>

Its back up now -- sorry for the noise.

covici at ccs.covici.com wrote:

> But the wiki seems to be down.
> 
> Jeff Lenk <jeff at jefflenk.com> wrote:
> 
> > Jira and Fisheye are now up.
> > 
> > --
> > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/freeswitch-org-is-down-tp6674580p6676501.html
> > Sent from the freeswitch-users mailing list archive at Nabble.com.
> > 
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> -- 
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
> 
>          John Covici
>          covici at ccs.covici.com
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From tculjaga at gmail.com  Thu Aug 11 19:10:05 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 11 Aug 2011 17:10:05 +0200
Subject: [Freeswitch-users] FS terminated maybe due to memory leak?
In-Reply-To: <1313071747872-6676511.post@n2.nabble.com>
References: <4E42D5FB.2020609@gosilverplus.com>
	<1313071747872-6676511.post@n2.nabble.com>
Message-ID: <CAJoKKdUjQXVUPV_CRiMxuzJKUQKyfbO-KSr7UUEXghWquj7oRA@mail.gmail.com>

also, use last git as a known memory leak was fixed recently.

T.

On Thu, Aug 11, 2011 at 4:09 PM, Jeff Lenk <jeff at jefflenk.com> wrote:

> you will need to provide some more details here:
> debug log of failed call
> sofia global siptrace on - log
>
> provide a link to pastebin.freeswitch.org
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/FS-terminated-maybe-due-to-memory-leak-tp6674632p6676511.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110811/b8485e1c/attachment.html 

From vipkilla at gmail.com  Thu Aug 11 19:26:15 2011
From: vipkilla at gmail.com (vip killa)
Date: Thu, 11 Aug 2011 11:26:15 -0400
Subject: [Freeswitch-users] collect dtmf and store in channel variable
Message-ID: <CAC-LwPOmcTxMZYUZKny2mfryY1MVvqJ9mWrtqS+VGV+G_cis2w@mail.gmail.com>

Hi everyone,
I'm trying to collect DTMF digits and store them in a channel variable so
when the channel hangs up it uses those digits to "mark" (or rename) the
recording of the call. The DTMF will be entered by the called party (i think
that would be the B-leg?). I've been experimenting with "bind_meta_app" and
"bind_digit_action", it seems like "bind_digit_action" may be the one i need
to use but im not sure. I'll explain the scenario to make things more
clear...we are trying to install this in a call center type environment
where all calls are being recorded. A caller gets an agent, the caller gives
the agent the account number of their case, the agent uses DTMF to mark the
recording of the call with the account number of the caller's case. Does
that make sense? Please let me know if this is possible, thanks.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110811/88b7afd1/attachment.html 

From x.liu at hw.ac.uk  Thu Aug 11 18:59:02 2011
From: x.liu at hw.ac.uk (xl127)
Date: Thu, 11 Aug 2011 15:59:02 +0100
Subject: [Freeswitch-users] any IVR example in C/C++?
Message-ID: <4E43EE36.60209@hw.ac.uk>

Hi,

I've seen some examples for ASR/TTS in JavaScripts/Lua/Python etc.
I can set my extension in the dialplan and point to my application e.g. by
<action application="python" data="mySpeechApp.py"/>.

I am wondering how I could do this for a C/C++ application?
And in the scripts languages I can set a callback method, e.g.
   session.setInputCallback(myInputCallback)
but I didn't find how to do this in C/C++.

Googled around, but didn't find much helpful info on it.

Any suggestions please?

Thanks!
Xing


-- 
Heriot-Watt University is a Scottish charity
registered under charity number SC000278.



From alec at efuse.co.uk  Thu Aug 11 19:06:06 2011
From: alec at efuse.co.uk (Alec Glassford)
Date: Thu, 11 Aug 2011 16:06:06 +0100
Subject: [Freeswitch-users] Running Post-Bridge scripts
Message-ID: <1ABE503ACB3B1D49A68430AE5DA62A552067B70B21@ESBS2K8.AMG.local>

Hi,

I need to run a LUA script after a Bridge to write call information to a MySQL DB.

The problem is the script is only running intermittently, maybe 1 in 10 calls.

I have tried the exec_after_bridge_app command, I've tried uuid_bridge_continue_on_cancel, but it neither works.  I can provide output if necessary.

I am bridging FreeTDM Sangoma A101 E1 ISDN30 calls to a SIP Provider, I do a DB lookup to find the correct Presentation number then bridge the call, below is from my Dialplan:

<condition field="destination_number" expression="^(\d{11})$">
     <action application="lua" data="lookup02.lua"/>
     <action application="set" data="effective_caller_id_number=${Presentation}"/>
     <action application="set" data="effective_caller_id_name=${Presentation}"/>
     <action application="log" data="INFO Dialling $1 presenting ${Presentation} from dblookup"/>
     <action application="set" data="uuid_bridge_continue_on_cancel=true"/>
     <action application="set" data="exec_after_bridge_app=lua"/>
     <action application="set" data="exec_after_bridge_arg=endcall01.lua"/>
     <action application="bridge" data="sofia/gateway/ProviderZZZ/$1" />
     <action application="lua" data="endcall01.lua"/>
     <action application="hangup" />
</condition>

Any help much appreciated


Alec Glassford
efuse
Tel:  0844 847 9707
Mob: 07540 417395
Fax: 0844 847 9708
www.efuse.co.uk<http://www.efuse.co.uk/>

This is an email from efuse, IT Solutions providers: www.efuse.co.uk<http://www.efuse.co.uk/>

P Save a tree...please don't print this e-mail unless you really need to
Its contents are confidential and legally privileged and it is intended only for the use of the addressees named above.  If you are not an addressee you must not read it and must not use any information contained in it nor copy it nor inform any person other than Efuse Solutions or the addressees of its existence or contents.
If you have received this email and are not a named addressee please delete it and notify support at efuse.co.uk<mailto:helpdesk at efuse.co.uk>
Please note that Efuse Solutions nor the sender accepts any responsibility for viruses and that it is your responsibility to scan any attachments.  No contractual obligations may be established on behalf of Efuse Solutions by means of email communication.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110811/9df6703e/attachment-0001.html 

From covici at ccs.covici.com  Thu Aug 11 20:22:16 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 11 Aug 2011 12:22:16 -0400
Subject: [Freeswitch-users] collect dtmf and store in channel variable
In-Reply-To: <CAC-LwPOmcTxMZYUZKny2mfryY1MVvqJ9mWrtqS+VGV+G_cis2w@mail.gmail.com>
References: <CAC-LwPOmcTxMZYUZKny2mfryY1MVvqJ9mWrtqS+VGV+G_cis2w@mail.gmail.com>
Message-ID: <23723.1313079736@ccs.covici.com>

Could not the agent just type it on a screen -- it would seem to be much
easier.

vip killa <vipkilla at gmail.com> wrote:

> Hi everyone,
> I'm trying to collect DTMF digits and store them in a channel variable so
> when the channel hangs up it uses those digits to "mark" (or rename) the
> recording of the call. The DTMF will be entered by the called party (i think
> that would be the B-leg?). I've been experimenting with "bind_meta_app" and
> "bind_digit_action", it seems like "bind_digit_action" may be the one i need
> to use but im not sure. I'll explain the scenario to make things more
> clear...we are trying to install this in a call center type environment
> where all calls are being recorded. A caller gets an agent, the caller gives
> the agent the account number of their case, the agent uses DTMF to mark the
> recording of the call with the account number of the caller's case. Does
> that make sense? Please let me know if this is possible, thanks.
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From vipkilla at gmail.com  Thu Aug 11 20:30:00 2011
From: vipkilla at gmail.com (vip killa)
Date: Thu, 11 Aug 2011 12:30:00 -0400
Subject: [Freeswitch-users] collect dtmf and store in channel variable
In-Reply-To: <23723.1313079736@ccs.covici.com>
References: <CAC-LwPOmcTxMZYUZKny2mfryY1MVvqJ9mWrtqS+VGV+G_cis2w@mail.gmail.com>
	<23723.1313079736@ccs.covici.com>
Message-ID: <CAC-LwPOSH8FxVoJB1PejS=+j==JY515u=_FDFo7M_0LeAG32nA@mail.gmail.com>

indeed it may be easier but this is what the client is asking for.... i know
it has to be possible... i just need some direction.

On Thu, Aug 11, 2011 at 12:22 PM, <covici at ccs.covici.com> wrote:

> Could not the agent just type it on a screen -- it would seem to be much
> easier.
>
> vip killa <vipkilla at gmail.com> wrote:
>
> > Hi everyone,
> > I'm trying to collect DTMF digits and store them in a channel variable so
> > when the channel hangs up it uses those digits to "mark" (or rename) the
> > recording of the call. The DTMF will be entered by the called party (i
> think
> > that would be the B-leg?). I've been experimenting with "bind_meta_app"
> and
> > "bind_digit_action", it seems like "bind_digit_action" may be the one i
> need
> > to use but im not sure. I'll explain the scenario to make things more
> > clear...we are trying to install this in a call center type environment
> > where all calls are being recorded. A caller gets an agent, the caller
> gives
> > the agent the account number of their case, the agent uses DTMF to mark
> the
> > recording of the call with the account number of the caller's case. Does
> > that make sense? Please let me know if this is possible, thanks.
> >
> > ----------------------------------------------------
> > Alternatives:
> >
> > ----------------------------------------------------
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>         John Covici
>         covici at ccs.covici.com
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110811/8492ffda/attachment.html 

From avi at avimarcus.net  Thu Aug 11 20:39:19 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 11 Aug 2011 19:39:19 +0300
Subject: [Freeswitch-users] Running Post-Bridge scripts
In-Reply-To: <1ABE503ACB3B1D49A68430AE5DA62A552067B70B21@ESBS2K8.AMG.local>
References: <1ABE503ACB3B1D49A68430AE5DA62A552067B70B21@ESBS2K8.AMG.local>
Message-ID: <CALNZuwkxMvkybAVPDJRVy=2RfxGXLDG38oA3HmgOpO+whsR9UQ@mail.gmail.com>

uuid_bridge_continue_on_cancel sounds like an API, not a dialplan tool..

exec_after_bridge_app according to the wiki means once the bridge ENDS. So
how about just using a hangup hook?

http://wiki.freeswitch.org/wiki/Variable_api_hangup_hook

-Avi Marcus

718-989-9485 (USA)
054-844-3271 (Israel Kosher)
077-228-5055 (Israel Landline)
020-3519-3606 (UK)



On Thu, Aug 11, 2011 at 6:06 PM, Alec Glassford <alec at efuse.co.uk> wrote:

> Hi,****
>
> ** **
>
> I need to run a LUA script after a Bridge to write call information to a
> MySQL DB.****
>
> ** **
>
> The problem is the script is only running intermittently, maybe 1 in 10
> calls.****
>
> ** **
>
> I have tried the exec_after_bridge_app command, I?ve tried
> uuid_bridge_continue_on_cancel, but it neither works.  I can provide output
> if necessary.****
>
> ** **
>
> I am bridging FreeTDM Sangoma A101 E1 ISDN30 calls to a SIP Provider, I do
> a DB lookup to find the correct Presentation number then bridge the call,
> below is from my Dialplan:****
>
> ** **
>
> <condition field="destination_number" expression="^(\d{11})$"> ****
>
>      <action application="lua" data="lookup02.lua"/> ****
>
>      <action application="set"
> data="effective_caller_id_number=${Presentation}"/>****
>
>      <action application="set"
> data="effective_caller_id_name=${Presentation}"/>****
>
>      <action application="log" data="INFO Dialling $1 presenting
> ${Presentation} from dblookup"/>****
>
>      <action application="set" data="uuid_bridge_continue_on_cancel=true"/>
> ****
>
>      <action application="set" data="exec_after_bridge_app=lua"/>****
>
>      <action application="set" data="exec_after_bridge_arg=endcall01.lua"/>
> ****
>
>      <action application="bridge" data="sofia/gateway/ProviderZZZ/$1" />**
> **
>
>      <action application="lua" data="endcall01.lua"/>****
>
>      <action application="hangup" />****
>
> </condition>****
>
> ** **
>
> Any help much appreciated****
>
> ** **
>
> ** **
>
> *Alec Glassford*****
>
> *efuse*****
>
> Tel:  0844 847 9707****
>
> Mob: 07540 417395****
>
> Fax: 0844 847 9708****
>
> www.efuse.co.uk****
>
> ** **
>
> This is an email from efuse, IT Solutions providers: www.efuse.co.uk****
>
>  ****
>
> P Save a tree...please don't print this e-mail* unless you really need to*
> ****
>
> Its contents are confidential and legally privileged and it is intended
> only for the use of the addressees named above.  If you are not an addressee
> you must not read it and must not use any information contained in it nor
> copy it nor inform any person other than Efuse Solutions or the addressees
> of its existence or contents.****
>
> If you have received this email and are not a named addressee please delete
> it and notify support at efuse.co.uk <helpdesk at efuse.co.uk>****
>
> Please note that Efuse Solutions nor the sender accepts any responsibility
> for viruses and that it is your responsibility to scan any attachments.  No
> contractual obligations may be established on behalf of Efuse Solutions by
> means of email communication.****
>
> ** **
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110811/3c3a32be/attachment.html 

From avi at avimarcus.net  Thu Aug 11 20:41:26 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 11 Aug 2011 19:41:26 +0300
Subject: [Freeswitch-users] collect dtmf and store in channel variable
In-Reply-To: <CAC-LwPOSH8FxVoJB1PejS=+j==JY515u=_FDFo7M_0LeAG32nA@mail.gmail.com>
References: <CAC-LwPOmcTxMZYUZKny2mfryY1MVvqJ9mWrtqS+VGV+G_cis2w@mail.gmail.com>
	<23723.1313079736@ccs.covici.com>
	<CAC-LwPOSH8FxVoJB1PejS=+j==JY515u=_FDFo7M_0LeAG32nA@mail.gmail.com>
Message-ID: <CALNZuw=kwNEcAaN+-oxrkOM4OxWiAc2Koryct1YbuhkMA28_MA@mail.gmail.com>

You can set any channel variable from the dialplan or from a lua script.
I'd reccommend a standalone lua script before the agent gets on the phone,
which can verify their account number and then present it to the agent even.
But yes, you can do anything with the dtmf including setting a variable
which can be saved from the xml_cdrs or mod_cdr_csv or whatever other cdr
system you use.

-Avi


On Thu, Aug 11, 2011 at 7:30 PM, vip killa <vipkilla at gmail.com> wrote:

> indeed it may be easier but this is what the client is asking for.... i
> know it has to be possible... i just need some direction.
>
>
> On Thu, Aug 11, 2011 at 12:22 PM, <covici at ccs.covici.com> wrote:
>
>> Could not the agent just type it on a screen -- it would seem to be much
>> easier.
>>
>> vip killa <vipkilla at gmail.com> wrote:
>>
>> > Hi everyone,
>> > I'm trying to collect DTMF digits and store them in a channel variable
>> so
>> > when the channel hangs up it uses those digits to "mark" (or rename) the
>> > recording of the call. The DTMF will be entered by the called party (i
>> think
>> > that would be the B-leg?). I've been experimenting with "bind_meta_app"
>> and
>> > "bind_digit_action", it seems like "bind_digit_action" may be the one i
>> need
>> > to use but im not sure. I'll explain the scenario to make things more
>> > clear...we are trying to install this in a call center type environment
>> > where all calls are being recorded. A caller gets an agent, the caller
>> gives
>> > the agent the account number of their case, the agent uses DTMF to mark
>> the
>> > recording of the call with the account number of the caller's case. Does
>> > that make sense? Please let me know if this is possible, thanks.
>> >
>> > ----------------------------------------------------
>> > Alternatives:
>> >
>> > ----------------------------------------------------
>> > _______________________________________________
>> > Join us at ClueCon 2011, Aug 9-11, Chicago
>> > http://www.cluecon.com 877-7-4ACLUE
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> --
>> Your life is like a penny.  You're going to lose it.  The question is:
>> How do
>> you spend it?
>>
>>         John Covici
>>         covici at ccs.covici.com
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110811/a24ba9f5/attachment-0001.html 

From jeff at jefflenk.com  Thu Aug 11 20:51:13 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 11 Aug 2011 09:51:13 -0700 (PDT)
Subject: [Freeswitch-users] Playback .wma/.wmv
In-Reply-To: <CAC4JBEDuBqK1KfsWG4rY1dd8ASk9zGzAgP76YqxbpJX9Dg-YmA@mail.gmail.com>
References: <CAC4JBEDuBqK1KfsWG4rY1dd8ASk9zGzAgP76YqxbpJX9Dg-YmA@mail.gmail.com>
Message-ID: <1313081473061-6677105.post@n2.nabble.com>

Not at this time. AFAIK

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Playback-wma-wmv-tp6674880p6677105.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From adrottenberg at gmail.com  Thu Aug 11 21:15:52 2011
From: adrottenberg at gmail.com (Duvid Rottenberg)
Date: Thu, 11 Aug 2011 13:15:52 -0400
Subject: [Freeswitch-users] Playback .wma/.wmv
In-Reply-To: <1313081473061-6677105.post@n2.nabble.com>
References: <CAC4JBEDuBqK1KfsWG4rY1dd8ASk9zGzAgP76YqxbpJX9Dg-YmA@mail.gmail.com>
	<1313081473061-6677105.post@n2.nabble.com>
Message-ID: <CAC4JBECvNjZqC66HC9MqE9HVghHkTOn5xFSt_RLZ6gBk+TCOuw@mail.gmail.com>

Thanks for the response. I guess I will have to write that myself (need to
brush up my c programming skills), I am thinking of doing a mod using
gstreamer. Does anyone have any recommendations?

On Thu, Aug 11, 2011 at 12:51 PM, Jeff Lenk <jeff at jefflenk.com> wrote:

> Not at this time. AFAIK
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Playback-wma-wmv-tp6674880p6677105.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110811/652ea96f/attachment.html 

From kris at livecall.com  Thu Aug 11 21:34:25 2011
From: kris at livecall.com (Kris)
Date: Thu, 11 Aug 2011 10:34:25 -0700
Subject: [Freeswitch-users] freeswitch.org is down
References: <33095823FD21DF429B481B5163264B7950FF060640@VMBX102.ihostexchange.net><CAO47Hpf4J1Az0WCY8TtSRW-wiGHzcifBH+XoyzkdESbaO8EoSQ@mail.gmail.com><003601cc57cd$a1348e30$e39daa90$@com><1313071506800-6676501.post@n2.nabble.com><8041.1313073879@ccs.covici.com>
	<9272.1313074389@ccs.covici.com>
Message-ID: <B6C8BA94F614432F9779DD00CFCDBBA8@stor1>

With the recent down time and not being able to lookup things on the wiki, I 
am wondering if there is or there should be a way to download backups of the 
web site, wiki, jira..If the site is destroyed by power problems, 
maliciously wiped out,etc, at least someone, somewhere, will have a tar ball 
to restore from.

Kris
----- Original Message ----- 
From: <covici at ccs.covici.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Thursday, August 11, 2011 7:53 AM
Subject: Re: [Freeswitch-users] freeswitch.org is down


> Its back up now -- sorry for the noise.
>
> covici at ccs.covici.com wrote:
>
>> But the wiki seems to be down.
>>
>> Jeff Lenk <jeff at jefflenk.com> wrote:
>>
>> > Jira and Fisheye are now up.
>> >
>> > --
>> > View this message in context: 
>> > http://freeswitch-users.2379917.n2.nabble.com/freeswitch-org-is-down-tp6674580p6676501.html
>> > Sent from the freeswitch-users mailing list archive at Nabble.com.
>> >
>> > _______________________________________________
>> > Join us at ClueCon 2011, Aug 9-11, Chicago
>> > http://www.cluecon.com 877-7-4ACLUE
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> -- 
>> Your life is like a penny.  You're going to lose it.  The question is:
>> How do
>> you spend it?
>>
>>          John Covici
>>          covici at ccs.covici.com
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> -- 
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>         John Covici
>         covici at ccs.covici.com
>
>
> 



From lakersman2006 at yahoo.com  Thu Aug 11 21:52:07 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Thu, 11 Aug 2011 10:52:07 -0700 (PDT)
Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
In-Reply-To: <CAHKqMb5v891O92bLj5t81Hm2H2XwZJ0J7QyWcYw3ho_XTL2jOw@mail.gmail.com>
References: <4E4164C0.8030507@tiendalinux.com>
	<1312937649.7702.YahooMailNeo@web161011.mail.bf1.yahoo.com>
	<CAKbxfG_802p+G1a9HtLEh_i-K1C06fpZgyR4_NP=jZJR68YK8g@mail.gmail.com>
	<1312993719.14274.YahooMailNeo@web161018.mail.bf1.yahoo.com>
	<CAHKqMb5v891O92bLj5t81Hm2H2XwZJ0J7QyWcYw3ho_XTL2jOw@mail.gmail.com>
Message-ID: <1313085127.89976.YahooMailNeo@web161019.mail.bf1.yahoo.com>

Michael,

Were you referring to the link http://www.nabble.com/Re:-Conference-javascript-and-hanuphooks-giving-me-headaches-p21614840.html? because the page no longer exists.



________________________________
From: Nandy Dagondon <gcd at i.ph>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, August 10, 2011 7:40 PM
Subject: Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide


hi sam,

i found this posthttp://comments.gmane.org/gmane.comp.telephony.freeswitch.user/8477

modify the script to suit your need.

hope it helps. ?just dig on w/ FS :-)
-nandy


On Thu, Aug 11, 2011 at 12:28 AM, Sam <lakersman2006 at yahoo.com> wrote:

Thanks for being so accommodating. I was a bit frustrated in trying to port over an asterisk agi script to freeswitch. I have spent many hours trying to learn how to configure freeswitch, I was about to give up, but I will play with the new changes you made and see if that works for me. 
>
>
>
>One other question, when the bridged call hangs up I do not see any value for the hangup time when using getVariable("hangup_time"), so how can I get it?
>
>
>
>
>________________________________
>From: Anthony Minessale <anthony.minessale at gmail.com>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>Sent: Wednesday, August 10, 2011 8:52 AM
>Subject: Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
>
>
>=D?
>
>
>ok, sure. ?If that's your only complaint.... see commit?9d98d49f0556fb79656c8403f285ae0a615439d3
>
>
>Some caveats
>
>
>1) There is actually less?specific, more generalized data in this DIALSTATUS variable than what we already report, when you're ready to move on see the originate_disposition variable: ?It's kind of like going from reporting the precise geo-location of a cafe in Paris to generalizing it to "EUROPE"?
>
>
>We follow the Q.850 standard for call cause codes and follow the SIP RFC to map sip response codes to/from the Q.850?equivalent. ?Also each module has its own version "sip_hangup_disposition" for sip so you have both the real sip response code AND the official Q.850 equiv variables set on each call.
>
>
>
>
>2) We don't have a torture feature so we never return that code.
>
>
>
>
>3) Since our originate can return before a call is answered I added "EARLY" which means the originate succeeded but its still not answered.
>
>
>4) For any others that do not map directly to FreeSWITCH, I just installed a copy of originate_disposition for good measure.
>
>P.S?
>
>
>This email took longer to compose than the patch did while sitting in the middle of a crowded room so you probably could have simply parsed the originate originate_disposition yourself but if it helps people get over being stuck in a?paradigm, it's worth it for me to write........
>?
>
>
>On Tue, Aug 9, 2011 at 7:54 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
>I find that Asterisk's AGI is much easier to use, they allow you to retrieve the dial status much easier than freeswitch's api's. Come on freeswitch, if you want to be better than asterisk, you should make it easier to get the dialstatus, etc. At this point asterisk is still defacto.
>>
>>
>>
>>
>>________________________________
>> From: Nestor A Diaz <nestor at tiendalinux.com>
>>To: freeswitch-users at lists.freeswitch.org
>>Sent: Tuesday, August 9, 2011 9:48 AM
>>Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>>
>>
>>
>>Hi Guys.
>>
>>I am starting to use FreeSWITCH, i am an asterisk user since the
1.0.7
release appears as a package on the debian distribution, at the
beginning i was amazed by the fact i can build a PBX for my own
business
and i did, later i began to install this system for my customers and
sooner i meet the problems, however being the software open source i
always find a way to fix things using patchs from others, sometimes i
felt how my life was at risk when the system stops working and that
usually happens when i have to use queues and dealing with digium
hardware.
>>
>>Fixing those problems either by applying patches or by changing
the
hardware where the digium cards were supposed to be installed helps me,
but that was to much stress for me and seeking for a balance that will
let me invest more time on services, configuration and hoping to have
better hardware
options brings me to freeswitch.
>>
>>I agree with freeswitch philosophy that instead of having
thousands of
modules
that don't work fine i prefer just a few that works the way it should
be, a rock solid system for a corporate pbx and a call center is what i
want.
>>
>>So here i am trying to begin the conversion, and i hope the
information
we can transcript in this list will help others that want to try
another
alternative to asterisk.
>>
>>First of all i think the saner for a migration is to have the two
systems
running either on the same machine or different and use the stable
features of each one.
>>
>>So could you please freeswitch users help me with this rosetta
stone migration guide in order to post it to voip-info.org or
freeswitch wiki (i list only the ones i currently use ):
>>
>>
>>
>>Technology Asterisk Freeswitch 
>>PSTN Connectivity (Digium /
Sangoma) dahdi freetdm 
>>IAX2 mod_iax ?? none stable yet.
>>Use Asterisk to forward traffic via SIP.
>>Enable Hardware HPET for IAX2 trunk if card not available for Asterisk  
>>Bluetooth Channel chan_mobile ??
>>Use asterisk via SIP
>> 
>>Skype Skypeforasterisk (no longer for sale) mod_skypeopen 
>>CDR Stadistics Arternic cdr-stats  ?? 
>>Queue Statistics  Asteriskguru queue-stats ?? 
>>Web Management Freepbx ?? 
>>IVR AGI / AMI Event Socket 
>>Codec G.729 Transcodind Cards
>>G.729 licenses
>>Free G.729 (Intel IPP) Transcodind Cards
>>G.729 licenses
>>fsg729 Intel IPP(any experience with it ? ) 
>>Fax Handling Iaxmodem with Hylafax ??
>>Iaxmodem via asterisk to FS via SIP ?
>> 
>>SIP chan_sip sofia 
>>ACD app_queue mod_callcenter 
>>
>>Thank you all
>>
>>
>>-- 
>>Nestor A. Diaz
>>Ingeniero de Sistemas
>>Tel. +57 1-485-3020 x 211
>>Cel. +57 316-227-3593
>>Tel. SIP: sip:211 at tiendalinux.com
>>Email/MSN: nestor at tiendalinux.com
>>http://www.tiendalinux.com/
>>Bogota, Colombia 
>> 
>>
>>
>>_______________________________________________
>>Join us at ClueCon 2011, Aug 9-11, Chicago
>>http://www.cluecon.com 877-7-4ACLUE
>>
>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>>
>>
>>_______________________________________________
>>Join us at ClueCon 2011, Aug 9-11, Chicago
>>http://www.cluecon.com 877-7-4ACLUE
>>
>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>>
>
>
>-- 
>Anthony Minessale II
>
>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
>googletalk:conf+888 at conference.freeswitch.org
>pstn:+19193869900
>
>_______________________________________________
>Join us at ClueCon 2011, Aug 9-11, Chicago
>http://www.cluecon.com 877-7-4ACLUE
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>
>
>_______________________________________________
>Join us at ClueCon 2011, Aug 9-11, Chicago
>http://www.cluecon.com 877-7-4ACLUE
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>

_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110811/511a5331/attachment-0001.html 

From vipkilla at gmail.com  Thu Aug 11 21:52:14 2011
From: vipkilla at gmail.com (vip killa)
Date: Thu, 11 Aug 2011 13:52:14 -0400
Subject: [Freeswitch-users] collect dtmf and store in channel variable
In-Reply-To: <CALNZuw=kwNEcAaN+-oxrkOM4OxWiAc2Koryct1YbuhkMA28_MA@mail.gmail.com>
References: <CAC-LwPOmcTxMZYUZKny2mfryY1MVvqJ9mWrtqS+VGV+G_cis2w@mail.gmail.com>
	<23723.1313079736@ccs.covici.com>
	<CAC-LwPOSH8FxVoJB1PejS=+j==JY515u=_FDFo7M_0LeAG32nA@mail.gmail.com>
	<CALNZuw=kwNEcAaN+-oxrkOM4OxWiAc2Koryct1YbuhkMA28_MA@mail.gmail.com>
Message-ID: <CAC-LwPOpgRcwhAm1Q3Y9VQ+X5qn+Wx1tQGxzesOsWm7EPJCKvQ@mail.gmail.com>

I apologize but the caller would not know the account number, it would be
for internal use only. how could xml_cdrs and mod_cdr_csv be used to stamp a
recording? how do you store the DTMF in a channel variable?

On Thu, Aug 11, 2011 at 12:41 PM, Avi Marcus <avi at avimarcus.net> wrote:

> You can set any channel variable from the dialplan or from a lua script.
> I'd reccommend a standalone lua script before the agent gets on the phone,
> which can verify their account number and then present it to the agent even.
> But yes, you can do anything with the dtmf including setting a variable
> which can be saved from the xml_cdrs or mod_cdr_csv or whatever other cdr
> system you use.
>
> -Avi
>
>
> On Thu, Aug 11, 2011 at 7:30 PM, vip killa <vipkilla at gmail.com> wrote:
>
>> indeed it may be easier but this is what the client is asking for.... i
>> know it has to be possible... i just need some direction.
>>
>>
>> On Thu, Aug 11, 2011 at 12:22 PM, <covici at ccs.covici.com> wrote:
>>
>>> Could not the agent just type it on a screen -- it would seem to be much
>>> easier.
>>>
>>> vip killa <vipkilla at gmail.com> wrote:
>>>
>>> > Hi everyone,
>>> > I'm trying to collect DTMF digits and store them in a channel variable
>>> so
>>> > when the channel hangs up it uses those digits to "mark" (or rename)
>>> the
>>> > recording of the call. The DTMF will be entered by the called party (i
>>> think
>>> > that would be the B-leg?). I've been experimenting with "bind_meta_app"
>>> and
>>> > "bind_digit_action", it seems like "bind_digit_action" may be the one i
>>> need
>>> > to use but im not sure. I'll explain the scenario to make things more
>>> > clear...we are trying to install this in a call center type environment
>>> > where all calls are being recorded. A caller gets an agent, the caller
>>> gives
>>> > the agent the account number of their case, the agent uses DTMF to mark
>>> the
>>> > recording of the call with the account number of the caller's case.
>>> Does
>>> > that make sense? Please let me know if this is possible, thanks.
>>> >
>>> > ----------------------------------------------------
>>> > Alternatives:
>>> >
>>> > ----------------------------------------------------
>>> > _______________________________________________
>>> > Join us at ClueCon 2011, Aug 9-11, Chicago
>>> > http://www.cluecon.com 877-7-4ACLUE
>>> >
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>>
>>> --
>>> Your life is like a penny.  You're going to lose it.  The question is:
>>> How do
>>> you spend it?
>>>
>>>         John Covici
>>>         covici at ccs.covici.com
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110811/5d506e1a/attachment.html 

From rzhang at gosilverplus.com  Thu Aug 11 22:05:11 2011
From: rzhang at gosilverplus.com (ran zhang)
Date: Thu, 11 Aug 2011 11:05:11 -0700
Subject: [Freeswitch-users] please help!!! playing tone only when 3rd party
 joins in bridging conference
Message-ID: <4E4419D7.4030509@gosilverplus.com>

I have a bridging conference established with 2 people in it, and i want 
to play a tone when 3rd party joins in. I can't set the conference's 
'enter-sound' to play the tone since it will play the tone when first 2 
people establish the bridging conference.





From avi at avimarcus.net  Thu Aug 11 22:07:42 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 11 Aug 2011 21:07:42 +0300
Subject: [Freeswitch-users] collect dtmf and store in channel variable
In-Reply-To: <CAC-LwPOpgRcwhAm1Q3Y9VQ+X5qn+Wx1tQGxzesOsWm7EPJCKvQ@mail.gmail.com>
References: <CAC-LwPOmcTxMZYUZKny2mfryY1MVvqJ9mWrtqS+VGV+G_cis2w@mail.gmail.com>
	<23723.1313079736@ccs.covici.com>
	<CAC-LwPOSH8FxVoJB1PejS=+j==JY515u=_FDFo7M_0LeAG32nA@mail.gmail.com>
	<CALNZuw=kwNEcAaN+-oxrkOM4OxWiAc2Koryct1YbuhkMA28_MA@mail.gmail.com>
	<CAC-LwPOpgRcwhAm1Q3Y9VQ+X5qn+Wx1tQGxzesOsWm7EPJCKvQ@mail.gmail.com>
Message-ID: <CALNZuw=ncGwazLjZ-gRdG1=L1oa_0p_26sE_j7mzj0u5DkT0DQ@mail.gmail.com>

Oh, you don't want the info in the CDR, you want the info in the recording.
Does it have to be in the name, or is an audio tag enough? I'm not sure if
tags can work once the recording started...
OK, so you can set a hangup hook to rename the file and add the variable
with the account code to the end.

Whatever you use to capture the DTMF - bind_digit_action or playandgetdigits
or anything can set a variable.
-Avi Marcus

p.s. If you want me to... actually do it for you, I'm available for
consulting.. email me offlist.


On Thu, Aug 11, 2011 at 8:52 PM, vip killa <vipkilla at gmail.com> wrote:

> I apologize but the caller would not know the account number, it would be
> for internal use only. how could xml_cdrs and mod_cdr_csv be used to stamp a
> recording? how do you store the DTMF in a channel variable?
>
>
> On Thu, Aug 11, 2011 at 12:41 PM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> You can set any channel variable from the dialplan or from a lua script.
>> I'd reccommend a standalone lua script before the agent gets on the phone,
>> which can verify their account number and then present it to the agent even.
>> But yes, you can do anything with the dtmf including setting a variable
>> which can be saved from the xml_cdrs or mod_cdr_csv or whatever other cdr
>> system you use.
>>
>> -Avi
>>
>>
>> On Thu, Aug 11, 2011 at 7:30 PM, vip killa <vipkilla at gmail.com> wrote:
>>
>>> indeed it may be easier but this is what the client is asking for.... i
>>> know it has to be possible... i just need some direction.
>>>
>>>
>>> On Thu, Aug 11, 2011 at 12:22 PM, <covici at ccs.covici.com> wrote:
>>>
>>>> Could not the agent just type it on a screen -- it would seem to be much
>>>> easier.
>>>>
>>>> vip killa <vipkilla at gmail.com> wrote:
>>>>
>>>> > Hi everyone,
>>>> > I'm trying to collect DTMF digits and store them in a channel variable
>>>> so
>>>> > when the channel hangs up it uses those digits to "mark" (or rename)
>>>> the
>>>> > recording of the call. The DTMF will be entered by the called party (i
>>>> think
>>>> > that would be the B-leg?). I've been experimenting with
>>>> "bind_meta_app" and
>>>> > "bind_digit_action", it seems like "bind_digit_action" may be the one
>>>> i need
>>>> > to use but im not sure. I'll explain the scenario to make things more
>>>> > clear...we are trying to install this in a call center type
>>>> environment
>>>> > where all calls are being recorded. A caller gets an agent, the caller
>>>> gives
>>>> > the agent the account number of their case, the agent uses DTMF to
>>>> mark the
>>>> > recording of the call with the account number of the caller's case.
>>>> Does
>>>> > that make sense? Please let me know if this is possible, thanks.
>>>> >
>>>> > ----------------------------------------------------
>>>> > Alternatives:
>>>> >
>>>> > ----------------------------------------------------
>>>> > _______________________________________________
>>>> > Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> > http://www.cluecon.com 877-7-4ACLUE
>>>> >
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>>
>>>> --
>>>> Your life is like a penny.  You're going to lose it.  The question is:
>>>> How do
>>>> you spend it?
>>>>
>>>>         John Covici
>>>>         covici at ccs.covici.com
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110811/5e43b691/attachment-0001.html 

From vipkilla at gmail.com  Thu Aug 11 22:13:37 2011
From: vipkilla at gmail.com (vip killa)
Date: Thu, 11 Aug 2011 14:13:37 -0400
Subject: [Freeswitch-users] collect dtmf and store in channel variable
In-Reply-To: <CALNZuw=ncGwazLjZ-gRdG1=L1oa_0p_26sE_j7mzj0u5DkT0DQ@mail.gmail.com>
References: <CAC-LwPOmcTxMZYUZKny2mfryY1MVvqJ9mWrtqS+VGV+G_cis2w@mail.gmail.com>
	<23723.1313079736@ccs.covici.com>
	<CAC-LwPOSH8FxVoJB1PejS=+j==JY515u=_FDFo7M_0LeAG32nA@mail.gmail.com>
	<CALNZuw=kwNEcAaN+-oxrkOM4OxWiAc2Koryct1YbuhkMA28_MA@mail.gmail.com>
	<CAC-LwPOpgRcwhAm1Q3Y9VQ+X5qn+Wx1tQGxzesOsWm7EPJCKvQ@mail.gmail.com>
	<CALNZuw=ncGwazLjZ-gRdG1=L1oa_0p_26sE_j7mzj0u5DkT0DQ@mail.gmail.com>
Message-ID: <CAC-LwPP9aWzJs1C9C-t5f+oqSZcf3rF0WSaFN4cSWZ+J29+5ug@mail.gmail.com>

thanks but i'm not interested in paying someone to do this. but yes i simply
want to collect the digits during the call from the B leg then store them in
a variable which will rename the file accordingly from the hangup hook

On Thu, Aug 11, 2011 at 2:07 PM, Avi Marcus <avi at avimarcus.net> wrote:

> Oh, you don't want the info in the CDR, you want the info in the recording.
> Does it have to be in the name, or is an audio tag enough? I'm not sure if
> tags can work once the recording started...
> OK, so you can set a hangup hook to rename the file and add the variable
> with the account code to the end.
>
> Whatever you use to capture the DTMF - bind_digit_action or
> playandgetdigits or anything can set a variable.
> -Avi Marcus
>
> p.s. If you want me to... actually do it for you, I'm available for
> consulting.. email me offlist.
>
>
> On Thu, Aug 11, 2011 at 8:52 PM, vip killa <vipkilla at gmail.com> wrote:
>
>> I apologize but the caller would not know the account number, it would be
>> for internal use only. how could xml_cdrs and mod_cdr_csv be used to stamp a
>> recording? how do you store the DTMF in a channel variable?
>>
>>
>> On Thu, Aug 11, 2011 at 12:41 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>
>>> You can set any channel variable from the dialplan or from a lua script.
>>> I'd reccommend a standalone lua script before the agent gets on the
>>> phone, which can verify their account number and then present it to the
>>> agent even. But yes, you can do anything with the dtmf including setting a
>>> variable which can be saved from the xml_cdrs or mod_cdr_csv or whatever
>>> other cdr system you use.
>>>
>>> -Avi
>>>
>>>
>>> On Thu, Aug 11, 2011 at 7:30 PM, vip killa <vipkilla at gmail.com> wrote:
>>>
>>>> indeed it may be easier but this is what the client is asking for.... i
>>>> know it has to be possible... i just need some direction.
>>>>
>>>>
>>>> On Thu, Aug 11, 2011 at 12:22 PM, <covici at ccs.covici.com> wrote:
>>>>
>>>>> Could not the agent just type it on a screen -- it would seem to be
>>>>> much
>>>>> easier.
>>>>>
>>>>> vip killa <vipkilla at gmail.com> wrote:
>>>>>
>>>>> > Hi everyone,
>>>>> > I'm trying to collect DTMF digits and store them in a channel
>>>>> variable so
>>>>> > when the channel hangs up it uses those digits to "mark" (or rename)
>>>>> the
>>>>> > recording of the call. The DTMF will be entered by the called party
>>>>> (i think
>>>>> > that would be the B-leg?). I've been experimenting with
>>>>> "bind_meta_app" and
>>>>> > "bind_digit_action", it seems like "bind_digit_action" may be the one
>>>>> i need
>>>>> > to use but im not sure. I'll explain the scenario to make things more
>>>>> > clear...we are trying to install this in a call center type
>>>>> environment
>>>>> > where all calls are being recorded. A caller gets an agent, the
>>>>> caller gives
>>>>> > the agent the account number of their case, the agent uses DTMF to
>>>>> mark the
>>>>> > recording of the call with the account number of the caller's case.
>>>>> Does
>>>>> > that make sense? Please let me know if this is possible, thanks.
>>>>> >
>>>>> > ----------------------------------------------------
>>>>> > Alternatives:
>>>>> >
>>>>> > ----------------------------------------------------
>>>>> > _______________________________________________
>>>>> > Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> > http://www.cluecon.com 877-7-4ACLUE
>>>>> >
>>>>> > FreeSWITCH-users mailing list
>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> > UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> > http://www.freeswitch.org
>>>>>
>>>>> --
>>>>> Your life is like a penny.  You're going to lose it.  The question is:
>>>>> How do
>>>>> you spend it?
>>>>>
>>>>>         John Covici
>>>>>         covici at ccs.covici.com
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110811/49a0bedb/attachment.html 

From curriegrad2004 at gmail.com  Thu Aug 11 22:30:16 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Thu, 11 Aug 2011 11:30:16 -0700
Subject: [Freeswitch-users] freeswitch.org is down
In-Reply-To: <B6C8BA94F614432F9779DD00CFCDBBA8@stor1>
References: <33095823FD21DF429B481B5163264B7950FF060640@VMBX102.ihostexchange.net>
	<CAO47Hpf4J1Az0WCY8TtSRW-wiGHzcifBH+XoyzkdESbaO8EoSQ@mail.gmail.com>
	<003601cc57cd$a1348e30$e39daa90$@com>
	<1313071506800-6676501.post@n2.nabble.com>
	<8041.1313073879@ccs.covici.com> <9272.1313074389@ccs.covici.com>
	<B6C8BA94F614432F9779DD00CFCDBBA8@stor1>
Message-ID: <CAMWiMy947m69MUf+a40hbET-q=ie-JQG=j7++ErLyvw26KbgOA@mail.gmail.com>

There is a feature that MediaWiki has, a backup of the entire site,
but I'm not too sure if it's enabled
by the admins of the wiki

On Thu, Aug 11, 2011 at 10:34 AM, Kris <kris at livecall.com> wrote:
> With the recent down time and not being able to lookup things on the wiki, I
> am wondering if there is or there should be a way to download backups of the
> web site, wiki, jira..If the site is destroyed by power problems,
> maliciously wiped out,etc, at least someone, somewhere, will have a tar ball
> to restore from.
>
> Kris
> ----- Original Message -----
> From: <covici at ccs.covici.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Thursday, August 11, 2011 7:53 AM
> Subject: Re: [Freeswitch-users] freeswitch.org is down
>
>
>> Its back up now -- sorry for the noise.
>>
>> covici at ccs.covici.com wrote:
>>
>>> But the wiki seems to be down.
>>>
>>> Jeff Lenk <jeff at jefflenk.com> wrote:
>>>
>>> > Jira and Fisheye are now up.
>>> >
>>> > --
>>> > View this message in context:
>>> > http://freeswitch-users.2379917.n2.nabble.com/freeswitch-org-is-down-tp6674580p6676501.html
>>> > Sent from the freeswitch-users mailing list archive at Nabble.com.
>>> >
>>> > _______________________________________________
>>> > Join us at ClueCon 2011, Aug 9-11, Chicago
>>> > http://www.cluecon.com 877-7-4ACLUE
>>> >
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>>
>>> --
>>> Your life is like a penny. ?You're going to lose it. ?The question is:
>>> How do
>>> you spend it?
>>>
>>> ? ? ? ? ?John Covici
>>> ? ? ? ? ?covici at ccs.covici.com
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> --
>> Your life is like a penny. ?You're going to lose it. ?The question is:
>> How do
>> you spend it?
>>
>> ? ? ? ? John Covici
>> ? ? ? ? covici at ccs.covici.com
>>
>>
>>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From curriegrad2004 at gmail.com  Thu Aug 11 22:35:21 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Thu, 11 Aug 2011 11:35:21 -0700
Subject: [Freeswitch-users] freeswitch.org is down
In-Reply-To: <CAMWiMy947m69MUf+a40hbET-q=ie-JQG=j7++ErLyvw26KbgOA@mail.gmail.com>
References: <33095823FD21DF429B481B5163264B7950FF060640@VMBX102.ihostexchange.net>
	<CAO47Hpf4J1Az0WCY8TtSRW-wiGHzcifBH+XoyzkdESbaO8EoSQ@mail.gmail.com>
	<003601cc57cd$a1348e30$e39daa90$@com>
	<1313071506800-6676501.post@n2.nabble.com>
	<8041.1313073879@ccs.covici.com> <9272.1313074389@ccs.covici.com>
	<B6C8BA94F614432F9779DD00CFCDBBA8@stor1>
	<CAMWiMy947m69MUf+a40hbET-q=ie-JQG=j7++ErLyvw26KbgOA@mail.gmail.com>
Message-ID: <CAMWiMy9Bus6y5RWLvxZ1bTpfem2BeZuDRgq_LA5kK+YE9rBxHQ@mail.gmail.com>

actually it's enabled:

http://wiki.freeswitch.org/wiki/Special:Export

On Thu, Aug 11, 2011 at 11:30 AM, curriegrad2004
<curriegrad2004 at gmail.com> wrote:
> There is a feature that MediaWiki has, a backup of the entire site,
> but I'm not too sure if it's enabled
> by the admins of the wiki
>
> On Thu, Aug 11, 2011 at 10:34 AM, Kris <kris at livecall.com> wrote:
>> With the recent down time and not being able to lookup things on the wiki, I
>> am wondering if there is or there should be a way to download backups of the
>> web site, wiki, jira..If the site is destroyed by power problems,
>> maliciously wiped out,etc, at least someone, somewhere, will have a tar ball
>> to restore from.
>>
>> Kris
>> ----- Original Message -----
>> From: <covici at ccs.covici.com>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Thursday, August 11, 2011 7:53 AM
>> Subject: Re: [Freeswitch-users] freeswitch.org is down
>>
>>
>>> Its back up now -- sorry for the noise.
>>>
>>> covici at ccs.covici.com wrote:
>>>
>>>> But the wiki seems to be down.
>>>>
>>>> Jeff Lenk <jeff at jefflenk.com> wrote:
>>>>
>>>> > Jira and Fisheye are now up.
>>>> >
>>>> > --
>>>> > View this message in context:
>>>> > http://freeswitch-users.2379917.n2.nabble.com/freeswitch-org-is-down-tp6674580p6676501.html
>>>> > Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>> >
>>>> > _______________________________________________
>>>> > Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> > http://www.cluecon.com 877-7-4ACLUE
>>>> >
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>>
>>>> --
>>>> Your life is like a penny. ?You're going to lose it. ?The question is:
>>>> How do
>>>> you spend it?
>>>>
>>>> ? ? ? ? ?John Covici
>>>> ? ? ? ? ?covici at ccs.covici.com
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>> --
>>> Your life is like a penny. ?You're going to lose it. ?The question is:
>>> How do
>>> you spend it?
>>>
>>> ? ? ? ? John Covici
>>> ? ? ? ? covici at ccs.covici.com
>>>
>>>
>>>
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>


From avi at avimarcus.net  Thu Aug 11 23:07:16 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 11 Aug 2011 22:07:16 +0300
Subject: [Freeswitch-users] collect dtmf and store in channel variable
In-Reply-To: <CAC-LwPP9aWzJs1C9C-t5f+oqSZcf3rF0WSaFN4cSWZ+J29+5ug@mail.gmail.com>
References: <CAC-LwPOmcTxMZYUZKny2mfryY1MVvqJ9mWrtqS+VGV+G_cis2w@mail.gmail.com>
	<23723.1313079736@ccs.covici.com>
	<CAC-LwPOSH8FxVoJB1PejS=+j==JY515u=_FDFo7M_0LeAG32nA@mail.gmail.com>
	<CALNZuw=kwNEcAaN+-oxrkOM4OxWiAc2Koryct1YbuhkMA28_MA@mail.gmail.com>
	<CAC-LwPOpgRcwhAm1Q3Y9VQ+X5qn+Wx1tQGxzesOsWm7EPJCKvQ@mail.gmail.com>
	<CALNZuw=ncGwazLjZ-gRdG1=L1oa_0p_26sE_j7mzj0u5DkT0DQ@mail.gmail.com>
	<CAC-LwPP9aWzJs1C9C-t5f+oqSZcf3rF0WSaFN4cSWZ+J29+5ug@mail.gmail.com>
Message-ID: <CALNZuwnF7hoysisyo2TQ40TFc17sfxk_0AH7PgBHwLzfnrqT+w@mail.gmail.com>

So some sort of bind_digit_action to collect and use that to set a hangup
hook. I then execute_extension from the action will be the easiest way to do
it, because you have several actions to accomplish.
However, doing this while the other person is on the hook.. sounds like a
bad idea.
-Avi


On Thu, Aug 11, 2011 at 9:13 PM, vip killa <vipkilla at gmail.com> wrote:

> thanks but i'm not interested in paying someone to do this. but yes i
> simply want to collect the digits during the call from the B leg then store
> them in a variable which will rename the file accordingly from the hangup
> hook
>
>
> On Thu, Aug 11, 2011 at 2:07 PM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> Oh, you don't want the info in the CDR, you want the info in the
>> recording. Does it have to be in the name, or is an audio tag enough? I'm
>> not sure if tags can work once the recording started...
>> OK, so you can set a hangup hook to rename the file and add the variable
>> with the account code to the end.
>>
>> Whatever you use to capture the DTMF - bind_digit_action or
>> playandgetdigits or anything can set a variable.
>> -Avi Marcus
>>
>> p.s. If you want me to... actually do it for you, I'm available for
>> consulting.. email me offlist.
>>
>>
>> On Thu, Aug 11, 2011 at 8:52 PM, vip killa <vipkilla at gmail.com> wrote:
>>
>>> I apologize but the caller would not know the account number, it would be
>>> for internal use only. how could xml_cdrs and mod_cdr_csv be used to stamp a
>>> recording? how do you store the DTMF in a channel variable?
>>>
>>>
>>> On Thu, Aug 11, 2011 at 12:41 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>>
>>>> You can set any channel variable from the dialplan or from a lua script.
>>>> I'd reccommend a standalone lua script before the agent gets on the
>>>> phone, which can verify their account number and then present it to the
>>>> agent even. But yes, you can do anything with the dtmf including setting a
>>>> variable which can be saved from the xml_cdrs or mod_cdr_csv or whatever
>>>> other cdr system you use.
>>>>
>>>> -Avi
>>>>
>>>>
>>>> On Thu, Aug 11, 2011 at 7:30 PM, vip killa <vipkilla at gmail.com> wrote:
>>>>
>>>>> indeed it may be easier but this is what the client is asking for.... i
>>>>> know it has to be possible... i just need some direction.
>>>>>
>>>>>
>>>>> On Thu, Aug 11, 2011 at 12:22 PM, <covici at ccs.covici.com> wrote:
>>>>>
>>>>>> Could not the agent just type it on a screen -- it would seem to be
>>>>>> much
>>>>>> easier.
>>>>>>
>>>>>> vip killa <vipkilla at gmail.com> wrote:
>>>>>>
>>>>>> > Hi everyone,
>>>>>> > I'm trying to collect DTMF digits and store them in a channel
>>>>>> variable so
>>>>>> > when the channel hangs up it uses those digits to "mark" (or rename)
>>>>>> the
>>>>>> > recording of the call. The DTMF will be entered by the called party
>>>>>> (i think
>>>>>> > that would be the B-leg?). I've been experimenting with
>>>>>> "bind_meta_app" and
>>>>>> > "bind_digit_action", it seems like "bind_digit_action" may be the
>>>>>> one i need
>>>>>> > to use but im not sure. I'll explain the scenario to make things
>>>>>> more
>>>>>> > clear...we are trying to install this in a call center type
>>>>>> environment
>>>>>> > where all calls are being recorded. A caller gets an agent, the
>>>>>> caller gives
>>>>>> > the agent the account number of their case, the agent uses DTMF to
>>>>>> mark the
>>>>>> > recording of the call with the account number of the caller's case.
>>>>>> Does
>>>>>> > that make sense? Please let me know if this is possible, thanks.
>>>>>> >
>>>>>> > ----------------------------------------------------
>>>>>> > Alternatives:
>>>>>> >
>>>>>> > ----------------------------------------------------
>>>>>> > _______________________________________________
>>>>>> > Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> > http://www.cluecon.com 877-7-4ACLUE
>>>>>> >
>>>>>> > FreeSWITCH-users mailing list
>>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> > UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> > http://www.freeswitch.org
>>>>>>
>>>>>> --
>>>>>> Your life is like a penny.  You're going to lose it.  The question is:
>>>>>> How do
>>>>>> you spend it?
>>>>>>
>>>>>>         John Covici
>>>>>>         covici at ccs.covici.com
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110811/4db8c387/attachment-0001.html 

From vipkilla at gmail.com  Thu Aug 11 23:18:35 2011
From: vipkilla at gmail.com (vip killa)
Date: Thu, 11 Aug 2011 15:18:35 -0400
Subject: [Freeswitch-users] collect dtmf and store in channel variable
In-Reply-To: <CALNZuwnF7hoysisyo2TQ40TFc17sfxk_0AH7PgBHwLzfnrqT+w@mail.gmail.com>
References: <CAC-LwPOmcTxMZYUZKny2mfryY1MVvqJ9mWrtqS+VGV+G_cis2w@mail.gmail.com>
	<23723.1313079736@ccs.covici.com>
	<CAC-LwPOSH8FxVoJB1PejS=+j==JY515u=_FDFo7M_0LeAG32nA@mail.gmail.com>
	<CALNZuw=kwNEcAaN+-oxrkOM4OxWiAc2Koryct1YbuhkMA28_MA@mail.gmail.com>
	<CAC-LwPOpgRcwhAm1Q3Y9VQ+X5qn+Wx1tQGxzesOsWm7EPJCKvQ@mail.gmail.com>
	<CALNZuw=ncGwazLjZ-gRdG1=L1oa_0p_26sE_j7mzj0u5DkT0DQ@mail.gmail.com>
	<CAC-LwPP9aWzJs1C9C-t5f+oqSZcf3rF0WSaFN4cSWZ+J29+5ug@mail.gmail.com>
	<CALNZuwnF7hoysisyo2TQ40TFc17sfxk_0AH7PgBHwLzfnrqT+w@mail.gmail.com>
Message-ID: <CAC-LwPP8_SgoLwjoYKec6buMN0Jz14=F86AS0QpTJfRobXTesQ@mail.gmail.com>

I can't seem to figure out how to store what is collected
from bind_digit_action into a channel variable... any ideas?

On Thu, Aug 11, 2011 at 3:07 PM, Avi Marcus <avi at avimarcus.net> wrote:

> So some sort of bind_digit_action to collect and use that to set a hangup
> hook. I then execute_extension from the action will be the easiest way to do
> it, because you have several actions to accomplish.
> However, doing this while the other person is on the hook.. sounds like a
> bad idea.
> -Avi
>
>
> On Thu, Aug 11, 2011 at 9:13 PM, vip killa <vipkilla at gmail.com> wrote:
>
>> thanks but i'm not interested in paying someone to do this. but yes i
>> simply want to collect the digits during the call from the B leg then store
>> them in a variable which will rename the file accordingly from the hangup
>> hook
>>
>>
>> On Thu, Aug 11, 2011 at 2:07 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>
>>> Oh, you don't want the info in the CDR, you want the info in the
>>> recording. Does it have to be in the name, or is an audio tag enough? I'm
>>> not sure if tags can work once the recording started...
>>> OK, so you can set a hangup hook to rename the file and add the variable
>>> with the account code to the end.
>>>
>>> Whatever you use to capture the DTMF - bind_digit_action or
>>> playandgetdigits or anything can set a variable.
>>> -Avi Marcus
>>>
>>> p.s. If you want me to... actually do it for you, I'm available for
>>> consulting.. email me offlist.
>>>
>>>
>>> On Thu, Aug 11, 2011 at 8:52 PM, vip killa <vipkilla at gmail.com> wrote:
>>>
>>>> I apologize but the caller would not know the account number, it would
>>>> be for internal use only. how could xml_cdrs and mod_cdr_csv be used to
>>>> stamp a recording? how do you store the DTMF in a channel variable?
>>>>
>>>>
>>>> On Thu, Aug 11, 2011 at 12:41 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>>>
>>>>> You can set any channel variable from the dialplan or from a lua
>>>>> script.
>>>>> I'd reccommend a standalone lua script before the agent gets on the
>>>>> phone, which can verify their account number and then present it to the
>>>>> agent even. But yes, you can do anything with the dtmf including setting a
>>>>> variable which can be saved from the xml_cdrs or mod_cdr_csv or whatever
>>>>> other cdr system you use.
>>>>>
>>>>> -Avi
>>>>>
>>>>>
>>>>> On Thu, Aug 11, 2011 at 7:30 PM, vip killa <vipkilla at gmail.com> wrote:
>>>>>
>>>>>> indeed it may be easier but this is what the client is asking for....
>>>>>> i know it has to be possible... i just need some direction.
>>>>>>
>>>>>>
>>>>>> On Thu, Aug 11, 2011 at 12:22 PM, <covici at ccs.covici.com> wrote:
>>>>>>
>>>>>>> Could not the agent just type it on a screen -- it would seem to be
>>>>>>> much
>>>>>>> easier.
>>>>>>>
>>>>>>> vip killa <vipkilla at gmail.com> wrote:
>>>>>>>
>>>>>>> > Hi everyone,
>>>>>>> > I'm trying to collect DTMF digits and store them in a channel
>>>>>>> variable so
>>>>>>> > when the channel hangs up it uses those digits to "mark" (or
>>>>>>> rename) the
>>>>>>> > recording of the call. The DTMF will be entered by the called party
>>>>>>> (i think
>>>>>>> > that would be the B-leg?). I've been experimenting with
>>>>>>> "bind_meta_app" and
>>>>>>> > "bind_digit_action", it seems like "bind_digit_action" may be the
>>>>>>> one i need
>>>>>>> > to use but im not sure. I'll explain the scenario to make things
>>>>>>> more
>>>>>>> > clear...we are trying to install this in a call center type
>>>>>>> environment
>>>>>>> > where all calls are being recorded. A caller gets an agent, the
>>>>>>> caller gives
>>>>>>> > the agent the account number of their case, the agent uses DTMF to
>>>>>>> mark the
>>>>>>> > recording of the call with the account number of the caller's case.
>>>>>>> Does
>>>>>>> > that make sense? Please let me know if this is possible, thanks.
>>>>>>> >
>>>>>>> > ----------------------------------------------------
>>>>>>> > Alternatives:
>>>>>>> >
>>>>>>> > ----------------------------------------------------
>>>>>>> > _______________________________________________
>>>>>>> > Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>> > http://www.cluecon.com 877-7-4ACLUE
>>>>>>> >
>>>>>>> > FreeSWITCH-users mailing list
>>>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> > UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> > http://www.freeswitch.org
>>>>>>>
>>>>>>> --
>>>>>>> Your life is like a penny.  You're going to lose it.  The question
>>>>>>> is:
>>>>>>> How do
>>>>>>> you spend it?
>>>>>>>
>>>>>>>         John Covici
>>>>>>>         covici at ccs.covici.com
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110811/e18dd348/attachment.html 

From anthony.minessale at gmail.com  Thu Aug 11 23:55:34 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 11 Aug 2011 14:55:34 -0500
Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
In-Reply-To: <1312992776.42043.YahooMailNeo@web161011.mail.bf1.yahoo.com>
References: <4E4164C0.8030507@tiendalinux.com>
	<1312937649.7702.YahooMailNeo@web161011.mail.bf1.yahoo.com>
	<757CA8EF-25EA-4372-AD39-0853551F4399@gmail.com>
	<1312992776.42043.YahooMailNeo@web161011.mail.bf1.yahoo.com>
Message-ID: <CAKbxfG8uJ7c0=gaB_QMmh4HEhCYEAtrpxLR-fU5jpdgt9X7uiw@mail.gmail.com>

in your cdr records, all of the variables are present there and they are
template-able
freeswitch separates the logic from the call handling.


On Wed, Aug 10, 2011 at 11:12 AM, Sam <lakersman2006 at yahoo.com> wrote:

> So have you had to retrieve the dial status from bridging a call in
> freeswitch? For the life of me I cannot properly get the answered_time  when
> looking up the channel variables after the bridge call finishes an answered
> call.
>
> ------------------------------
> *From:* Moe Navid <manavid at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Wednesday, August 10, 2011 1:44 AM
> *Subject:* Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
> There is no way by any means to compare Asterisk's AGI with the different
> facilities FreeSWITCH offers you in terms of controlling your call flow.
>
> For almost 3 years I managed a cluster of Asterisk + AGI + AMI with tones
> of channel locks and core dumps? Asterisk's dial status might seem
> compelling when you want to do simple things like calling cards etc? but
> when it comes to complex accounting and routing sky is limitless with the
> power of FreeSWITCH.
>
> I found FreeSWITCH's learning curve to be like vim, initially it may seem a
> bit difficult but in long run it pays of very well.
>
> If you know the difference between Dial command in Asterisk and Bridge in
> FreeSWITCH you would never go back to Asterisk. I give you just 3 simple
> examples:
> 1) Bridge command (via the channel variables) gives you the ability to
> control PDD on calls. Asterisk does not have such facility nonetheless it
> does not even bother to give you any useful information about your "Dial
> Status"! To control the PDD I had to tweak my kamailio.
>
> 2) If you want to implement a simple rate engine + fail over routing with
> asterisk + agi for failover you have to have a loop and watch for
> CONGESTIONs to select your next route/carrier where as in FreeSWITCH you can
> just simply define your fail overs in your bridge args.
>
> 3) If you are in a cluster, have multiple gateways acting as proxy and you
> want to define outbound proxy for your carriers/endpoints you either have to
> define bunch of sip peers with outbound proxies or do it in dirty way which
> I did, I used to add a header in my outgoing calls X-Carrier-IP and had my
> kamailio to take care of the rest. In FreeSWITCH you just simply add
> ;fspath= to your bridge args.
>
> List can go on and on and on?
>
> Asterisk's dial status was the most annoying part of asterisk in my
> opinion :)
>
> On Aug 9, 2011, at 5:54 PM, Sam wrote:
>
> I find that Asterisk's AGI is much easier to use, they allow you to
> retrieve the dial status much easier than freeswitch's api's. Come on
> freeswitch, if you want to be better than asterisk, you should make it
> easier to get the dialstatus, etc. At this point asterisk is still defacto.
>
> ------------------------------
> *From:* Nestor A Diaz <nestor at tiendalinux.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Tuesday, August 9, 2011 9:48 AM
> *Subject:* [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
> Hi Guys.
>
> I am starting to use FreeSWITCH, i am an asterisk user since the 1.0.7
> release appears as a package on the debian distribution, at the beginning i
> was amazed by the fact i can build a PBX for my own business and i did,
> later i began to install this system for my customers and sooner i meet the
> problems, however being the software open source i always find a way to fix
> things using patchs from others, sometimes i felt how my life was at risk
> when the system stops working and that usually happens when i have to use
> queues and dealing with digium hardware.
>
> Fixing those problems either by applying patches or by changing the
> hardware where the digium cards were supposed to be installed helps me, but
> that was to much stress for me and seeking for a balance that will let me
> invest more time on services, configuration and hoping to have better
> hardware options brings me to freeswitch.
>
> I agree with freeswitch philosophy that instead of having thousands of
> modules that don't work fine i prefer just a few that works the way it
> should be, a rock solid system for a corporate pbx and a call center is what
> i want.
>
> So here i am trying to begin the conversion, and i hope the information we
> can transcript in this list will help others that want to try another
> alternative to asterisk.
>
> First of all i think the saner for a migration is to have the two systems
> running either on the same machine or different and use the stable features
> of each one.
>
> So could you please freeswitch users help me with this rosetta stone
> migration guide in order to post it to voip-info.org or freeswitch wiki (i
> list only the ones i currently use ):
>
>
>   *Technology* *Asterisk* *Freeswitch*  PSTN Connectivity (Digium /
> Sangoma) dahdi freetdm  IAX2 mod_iax ?? none stable yet.
> Use Asterisk to forward traffic via SIP.
> Enable Hardware HPET for IAX2 trunk if card not available for Asterisk  Bluetooth
> Channel chan_mobile ??
> Use asterisk via SIP
>   Skype Skypeforasterisk (no longer for sale) mod_skypeopen  CDR
> Stadistics Arternic cdr-stats ??  Queue Statistics  Asteriskguru
> queue-stats ??  Web Management Freepbx ??  IVR AGI / AMI Event Socket  Codec
> G.729 Transcodind Cards
> G.729 licenses
> Free G.729 (Intel IPP) Transcodind Cards
> G.729 licenses
> fsg729 Intel IPP(any experience with it ? )  Fax Handling Iaxmodem with
> Hylafax ??
> Iaxmodem via asterisk to FS via SIP ?
>   SIP chan_sip sofia  ACD app_queue mod_callcenter
>
> Thank you all
>
>
> --
> Nestor A. Diaz
> Ingeniero de Sistemas
> Tel. +57 1-485-3020 x 211
> Cel. +57 316-227-3593
> Tel. SIP: sip:211 at tiendalinux.com
> Email/MSN: nestor at tiendalinux.com
> http://www.tiendalinux.com/
> Bogota, Colombia
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110811/f897ee03/attachment-0001.html 

From anthony.minessale at gmail.com  Thu Aug 11 23:59:28 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 11 Aug 2011 14:59:28 -0500
Subject: [Freeswitch-users] FS performance using ESL
In-Reply-To: <CAJoKKdU9VPzOS-C3jHCjG6Y0V-Jo_j7hqYwt6-Zh7ohb1pq3xQ@mail.gmail.com>
References: <CAJoKKdU9VPzOS-C3jHCjG6Y0V-Jo_j7hqYwt6-Zh7ohb1pq3xQ@mail.gmail.com>
Message-ID: <CAKbxfG-SWBzu0dBv_5B0v7710q7vLeeduT9L5V_cnceYHjrGJQ@mail.gmail.com>

try removing the filter and event subscriptions
it's costly to consume all of the events especially at 75cps.


On Thu, Aug 11, 2011 at 5:23 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:
> hello,
>
> im wondering how much performance do we loose when using ESL instead of
> running it via dialplan?
>
>
> without ESL with a fine tuned FS and a short dialplan ( answer, playback
> like 20 seconds file, hangup ) im able to service 75 CPS. On the same FS,
> when i use ESL to answer the call, playback the same file and hangup, im not
> able to run more than 2 CPS... this is a huge impact and i really can't
> believe it.
>
> I'm using event-socket outbound e.g.:
>
> <action application="socket" data="127.0.0.1:8088 async full"/>
>
> my extension looks like:
>
> <extension name="ESL_C">
> ? <condition field="destination_number" expression="^(6666)$">
> ??? <action application="socket" data="127.0.0.1:8088 async full"/>
> ??? <action application="sleep" data="1000"/>
> ??? <action application="hangup"/>
> ? </condition>
> </extension>
>
>
> im using testserver from lib/esl/ and i just removed the conference command
> and added the playback one.... also i moved the esl_debug lvl to 0
>
>
> anyhow, FS cannot run more than 2 CPS compared to 75 CPS when the playback
> is done from the dialplan.
>
>
> Please, can someone give me a clue on what is going on?
> Maybe im doing something wrong?
> how to get maximum FS performance using ESL ?
>
>
>
> Regards,
> Tihomir.
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From anthony.minessale at gmail.com  Fri Aug 12 00:04:05 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 11 Aug 2011 15:04:05 -0500
Subject: [Freeswitch-users] odbc basic_calls
In-Reply-To: <1313073015901-6676584.post@n2.nabble.com>
References: <CALNZuwk20r8=rmu9rOOd-aChOS6XOGVm7mxKNhd1-DnC+id7Hg@mail.gmail.com>
	<1313073015901-6676584.post@n2.nabble.com>
Message-ID: <CAKbxfG_RcTt6Uw44C8QB9BE27F7mSuVqxiu=m5gbXX7Y6p0Ofw@mail.gmail.com>

i see a small way it may happen, update again and see if it works.


On Thu, Aug 11, 2011 at 9:30 AM, mazilo <Nabble at slickdeals.endjunk.com> wrote:
> Avi,
>
> I also just upgraded FS since .. 03/26/2011 ago. Now, my Seagate DockStar is
> hosting the same FS version you have. Notice that I don't know much about
> using ODBC, but my current FS version has been compiled with the
> /--enable-core-odbc-support/ option. So, I haven't done anything to create
> any SQL database to use with my FS. However, then I pressed F4 key while I
> was on a call to ext. 3000 (conference call), it showed me with the call in
> progress and didn't show me the error message as you posted above. I don't
> know if having no SQL database file has anything to do with it.
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity.
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/odbc-basic-calls-tp6676013p6676584.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From avi at avimarcus.net  Fri Aug 12 00:15:16 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 11 Aug 2011 23:15:16 +0300
Subject: [Freeswitch-users] collect dtmf and store in channel variable
In-Reply-To: <CAC-LwPP8_SgoLwjoYKec6buMN0Jz14=F86AS0QpTJfRobXTesQ@mail.gmail.com>
References: <CAC-LwPOmcTxMZYUZKny2mfryY1MVvqJ9mWrtqS+VGV+G_cis2w@mail.gmail.com>
	<23723.1313079736@ccs.covici.com>
	<CAC-LwPOSH8FxVoJB1PejS=+j==JY515u=_FDFo7M_0LeAG32nA@mail.gmail.com>
	<CALNZuw=kwNEcAaN+-oxrkOM4OxWiAc2Koryct1YbuhkMA28_MA@mail.gmail.com>
	<CAC-LwPOpgRcwhAm1Q3Y9VQ+X5qn+Wx1tQGxzesOsWm7EPJCKvQ@mail.gmail.com>
	<CALNZuw=ncGwazLjZ-gRdG1=L1oa_0p_26sE_j7mzj0u5DkT0DQ@mail.gmail.com>
	<CAC-LwPP9aWzJs1C9C-t5f+oqSZcf3rF0WSaFN4cSWZ+J29+5ug@mail.gmail.com>
	<CALNZuwnF7hoysisyo2TQ40TFc17sfxk_0AH7PgBHwLzfnrqT+w@mail.gmail.com>
	<CAC-LwPP8_SgoLwjoYKec6buMN0Jz14=F86AS0QpTJfRobXTesQ@mail.gmail.com>
Message-ID: <CALNZuwkBHptWyKCWYb=X-ApcP79BxFWeySCq7SEKvFJhMdtDNQ@mail.gmail.com>

According to the wiki, apparently, you can't capture digits from
bind_digit_action. So you'll have to use that to trigger a
Dialplan_Tools_play_and_get_digits<http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits>
which
automatically sets a variable, or getdigits within lua/js (only?) and then
save it.

-Avi


On Thu, Aug 11, 2011 at 10:18 PM, vip killa <vipkilla at gmail.com> wrote:

> I can't seem to figure out how to store what is collected
> from bind_digit_action into a channel variable... any ideas?
>
>
> On Thu, Aug 11, 2011 at 3:07 PM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> So some sort of bind_digit_action to collect and use that to set a hangup
>> hook. I then execute_extension from the action will be the easiest way to do
>> it, because you have several actions to accomplish.
>> However, doing this while the other person is on the hook.. sounds like a
>> bad idea.
>> -Avi
>>
>>
>> On Thu, Aug 11, 2011 at 9:13 PM, vip killa <vipkilla at gmail.com> wrote:
>>
>>> thanks but i'm not interested in paying someone to do this. but yes i
>>> simply want to collect the digits during the call from the B leg then store
>>> them in a variable which will rename the file accordingly from the hangup
>>> hook
>>>
>>>
>>> On Thu, Aug 11, 2011 at 2:07 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>>
>>>> Oh, you don't want the info in the CDR, you want the info in the
>>>> recording. Does it have to be in the name, or is an audio tag enough? I'm
>>>> not sure if tags can work once the recording started...
>>>> OK, so you can set a hangup hook to rename the file and add the variable
>>>> with the account code to the end.
>>>>
>>>> Whatever you use to capture the DTMF - bind_digit_action or
>>>> playandgetdigits or anything can set a variable.
>>>> -Avi Marcus
>>>>
>>>> p.s. If you want me to... actually do it for you, I'm available for
>>>> consulting.. email me offlist.
>>>>
>>>>
>>>> On Thu, Aug 11, 2011 at 8:52 PM, vip killa <vipkilla at gmail.com> wrote:
>>>>
>>>>> I apologize but the caller would not know the account number, it would
>>>>> be for internal use only. how could xml_cdrs and mod_cdr_csv be used to
>>>>> stamp a recording? how do you store the DTMF in a channel variable?
>>>>>
>>>>>
>>>>> On Thu, Aug 11, 2011 at 12:41 PM, Avi Marcus <avi at avimarcus.net>wrote:
>>>>>
>>>>>> You can set any channel variable from the dialplan or from a lua
>>>>>> script.
>>>>>> I'd reccommend a standalone lua script before the agent gets on the
>>>>>> phone, which can verify their account number and then present it to the
>>>>>> agent even. But yes, you can do anything with the dtmf including setting a
>>>>>> variable which can be saved from the xml_cdrs or mod_cdr_csv or whatever
>>>>>> other cdr system you use.
>>>>>>
>>>>>> -Avi
>>>>>>
>>>>>>
>>>>>> On Thu, Aug 11, 2011 at 7:30 PM, vip killa <vipkilla at gmail.com>wrote:
>>>>>>
>>>>>>> indeed it may be easier but this is what the client is asking for....
>>>>>>> i know it has to be possible... i just need some direction.
>>>>>>>
>>>>>>>
>>>>>>> On Thu, Aug 11, 2011 at 12:22 PM, <covici at ccs.covici.com> wrote:
>>>>>>>
>>>>>>>> Could not the agent just type it on a screen -- it would seem to be
>>>>>>>> much
>>>>>>>> easier.
>>>>>>>>
>>>>>>>> vip killa <vipkilla at gmail.com> wrote:
>>>>>>>>
>>>>>>>> > Hi everyone,
>>>>>>>> > I'm trying to collect DTMF digits and store them in a channel
>>>>>>>> variable so
>>>>>>>> > when the channel hangs up it uses those digits to "mark" (or
>>>>>>>> rename) the
>>>>>>>> > recording of the call. The DTMF will be entered by the called
>>>>>>>> party (i think
>>>>>>>> > that would be the B-leg?). I've been experimenting with
>>>>>>>> "bind_meta_app" and
>>>>>>>> > "bind_digit_action", it seems like "bind_digit_action" may be the
>>>>>>>> one i need
>>>>>>>> > to use but im not sure. I'll explain the scenario to make things
>>>>>>>> more
>>>>>>>> > clear...we are trying to install this in a call center type
>>>>>>>> environment
>>>>>>>> > where all calls are being recorded. A caller gets an agent, the
>>>>>>>> caller gives
>>>>>>>> > the agent the account number of their case, the agent uses DTMF to
>>>>>>>> mark the
>>>>>>>> > recording of the call with the account number of the caller's
>>>>>>>> case. Does
>>>>>>>> > that make sense? Please let me know if this is possible, thanks.
>>>>>>>> >
>>>>>>>> > ----------------------------------------------------
>>>>>>>> > Alternatives:
>>>>>>>> >
>>>>>>>> > ----------------------------------------------------
>>>>>>>> > _______________________________________________
>>>>>>>> > Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>> > http://www.cluecon.com 877-7-4ACLUE
>>>>>>>> >
>>>>>>>> > FreeSWITCH-users mailing list
>>>>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> > UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> > http://www.freeswitch.org
>>>>>>>>
>>>>>>>> --
>>>>>>>> Your life is like a penny.  You're going to lose it.  The question
>>>>>>>> is:
>>>>>>>> How do
>>>>>>>> you spend it?
>>>>>>>>
>>>>>>>>         John Covici
>>>>>>>>         covici at ccs.covici.com
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110811/3543c704/attachment-0001.html 

From avi at avimarcus.net  Fri Aug 12 00:22:14 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 11 Aug 2011 23:22:14 +0300
Subject: [Freeswitch-users] odbc basic_calls
In-Reply-To: <CAKbxfG_RcTt6Uw44C8QB9BE27F7mSuVqxiu=m5gbXX7Y6p0Ofw@mail.gmail.com>
References: <CALNZuwk20r8=rmu9rOOd-aChOS6XOGVm7mxKNhd1-DnC+id7Hg@mail.gmail.com>
	<1313073015901-6676584.post@n2.nabble.com>
	<CAKbxfG_RcTt6Uw44C8QB9BE27F7mSuVqxiu=m5gbXX7Y6p0Ofw@mail.gmail.com>
Message-ID: <CALNZuwm9aAsH9z=hZgAHcQCx8f=4kNAw+s1h-2mk=EmhMJxQrw@mail.gmail.com>

I did 1) git pull 2) make sync, rather than doing a make clean, 3) restarted
and 4) got:

2011-08-11 23:20:45.654766 [ERR] switch_core_sqldb.c:825 ERR: [select * from
basic_calls where hostname='sip2' order by call_created_epoch]
[STATE: 42S02 CODE 1146 ERROR: [unixODBC][MySQL][ODBC 3.51
Driver][mysqld-5.1.41-3ubuntu12.10-log]Table 'freeswitch.basic_calls'
doesn't exist

Would a make clean be any different?
-Avi


On Thu, Aug 11, 2011 at 11:04 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> i see a small way it may happen, update again and see if it works.
>
>
> On Thu, Aug 11, 2011 at 9:30 AM, mazilo <Nabble at slickdeals.endjunk.com>
> wrote:
> > Avi,
> >
> > I also just upgraded FS since .. 03/26/2011 ago. Now, my Seagate DockStar
> is
> > hosting the same FS version you have. Notice that I don't know much about
> > using ODBC, but my current FS version has been compiled with the
> > /--enable-core-odbc-support/ option. So, I haven't done anything to
> create
> > any SQL database to use with my FS. However, then I pressed F4 key while
> I
> > was on a call to ext. 3000 (conference call), it showed me with the call
> in
> > progress and didn't show me the error message as you posted above. I
> don't
> > know if having no SQL database file has anything to do with it.
> >
> > -----
> > FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3
> Watts of electricity.
> > --
> > View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/odbc-basic-calls-tp6676013p6676584.html
> > Sent from the freeswitch-users mailing list archive at Nabble.com.
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110811/56a30393/attachment.html 

From anthony.minessale at gmail.com  Fri Aug 12 02:12:58 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 11 Aug 2011 17:12:58 -0500
Subject: [Freeswitch-users] odbc basic_calls
In-Reply-To: <CALNZuwm9aAsH9z=hZgAHcQCx8f=4kNAw+s1h-2mk=EmhMJxQrw@mail.gmail.com>
References: <CALNZuwk20r8=rmu9rOOd-aChOS6XOGVm7mxKNhd1-DnC+id7Hg@mail.gmail.com>
	<1313073015901-6676584.post@n2.nabble.com>
	<CAKbxfG_RcTt6Uw44C8QB9BE27F7mSuVqxiu=m5gbXX7Y6p0Ofw@mail.gmail.com>
	<CALNZuwm9aAsH9z=hZgAHcQCx8f=4kNAw+s1h-2mk=EmhMJxQrw@mail.gmail.com>
Message-ID: <CAKbxfG8HCdPAhSFuF30dw-q1N8XXVD=GADCqJPhzw6CZ+14Tbg@mail.gmail.com>

that's good, the error should be what triggered the creation of the view,

On Thu, Aug 11, 2011 at 3:22 PM, Avi Marcus <avi at avimarcus.net> wrote:
> I did 1) git pull 2) make sync, rather than doing a make clean, 3) restarted
> and 4) got:
> 2011-08-11 23:20:45.654766 [ERR] switch_core_sqldb.c:825 ERR: [select * from
> basic_calls where hostname='sip2' order by call_created_epoch]
> [STATE: 42S02 CODE 1146 ERROR: [unixODBC][MySQL][ODBC 3.51
> Driver][mysqld-5.1.41-3ubuntu12.10-log]Table 'freeswitch.basic_calls'
> doesn't exist
> Would a make clean be any different?
> -Avi
>
> On Thu, Aug 11, 2011 at 11:04 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> i see a small way it may happen, update again and see if it works.
>>
>>
>> On Thu, Aug 11, 2011 at 9:30 AM, mazilo <Nabble at slickdeals.endjunk.com>
>> wrote:
>> > Avi,
>> >
>> > I also just upgraded FS since .. 03/26/2011 ago. Now, my Seagate
>> > DockStar is
>> > hosting the same FS version you have. Notice that I don't know much
>> > about
>> > using ODBC, but my current FS version has been compiled with the
>> > /--enable-core-odbc-support/ option. So, I haven't done anything to
>> > create
>> > any SQL database to use with my FS. However, then I pressed F4 key while
>> > I
>> > was on a call to ext. 3000 (conference call), it showed me with the call
>> > in
>> > progress and didn't show me the error message as you posted above. I
>> > don't
>> > know if having no SQL database file has anything to do with it.
>> >
>> > -----
>> > FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3
>> > Watts of electricity.
>> > --
>> > View this message in context:
>> > http://freeswitch-users.2379917.n2.nabble.com/odbc-basic-calls-tp6676013p6676584.html
>> > Sent from the freeswitch-users mailing list archive at Nabble.com.
>> >
>> > _______________________________________________
>> > Join us at ClueCon 2011, Aug 9-11, Chicago
>> > http://www.cluecon.com 877-7-4ACLUE
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> 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
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From avi at avimarcus.net  Fri Aug 12 02:18:17 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 12 Aug 2011 01:18:17 +0300
Subject: [Freeswitch-users] odbc basic_calls
In-Reply-To: <CAKbxfG8HCdPAhSFuF30dw-q1N8XXVD=GADCqJPhzw6CZ+14Tbg@mail.gmail.com>
References: <CALNZuwk20r8=rmu9rOOd-aChOS6XOGVm7mxKNhd1-DnC+id7Hg@mail.gmail.com>
	<1313073015901-6676584.post@n2.nabble.com>
	<CAKbxfG_RcTt6Uw44C8QB9BE27F7mSuVqxiu=m5gbXX7Y6p0Ofw@mail.gmail.com>
	<CALNZuwm9aAsH9z=hZgAHcQCx8f=4kNAw+s1h-2mk=EmhMJxQrw@mail.gmail.com>
	<CAKbxfG8HCdPAhSFuF30dw-q1N8XXVD=GADCqJPhzw6CZ+14Tbg@mail.gmail.com>
Message-ID: <CALNZuwkh5cQd2qP=xhj3jN7SF81E0x49JYRR1pU--7rCbXrp7Q@mail.gmail.com>

I didn't think to try it twice. But no,

freeswitch at default> show calls

-ERR SQL Error [STATE: 42S02 CODE 1146 ERROR: [unixODBC][MySQL][ODBC 3.51
Driver][mysqld-5.1.41-3ubuntu12.10-log]Table 'freeswitch.basic_calls'
doesn't exist
]

freeswitch at default> 2011-08-12 01:17:40.494779 [ERR] switch_core_sqldb.c:825
ERR: [select * from basic_calls where hostname='sip2' order by
call_created_epoch]
[STATE: 42S02 CODE 1146 ERROR: [unixODBC][MySQL][ODBC 3.51
Driver][mysqld-5.1.41-3ubuntu12.10-log]Table 'freeswitch.basic_calls'
doesn't exist
]
show calls

-ERR SQL Error [STATE: 42S02 CODE 1146 ERROR: [unixODBC][MySQL][ODBC 3.51
Driver][mysqld-5.1.41-3ubuntu12.10-log]Table 'freeswitch.basic_calls'
doesn't exist
]

freeswitch at default> 2011-08-12 01:17:41.314777 [ERR] switch_core_sqldb.c:825
ERR: [select * from basic_calls where hostname='sip2' order by
call_created_epoch]
[STATE: 42S02 CODE 1146 ERROR: [unixODBC][MySQL][ODBC 3.51
Driver][mysqld-5.1.41-3ubuntu12.10-log]Table 'freeswitch.basic_calls'
doesn't exist





-Avi


On Fri, Aug 12, 2011 at 1:12 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> that's good, the error should be what triggered the creation of the view,
>
> On Thu, Aug 11, 2011 at 3:22 PM, Avi Marcus <avi at avimarcus.net> wrote:
> > I did 1) git pull 2) make sync, rather than doing a make clean, 3)
> restarted
> > and 4) got:
> > 2011-08-11 23:20:45.654766 [ERR] switch_core_sqldb.c:825 ERR: [select *
> from
> > basic_calls where hostname='sip2' order by call_created_epoch]
> > [STATE: 42S02 CODE 1146 ERROR: [unixODBC][MySQL][ODBC 3.51
> > Driver][mysqld-5.1.41-3ubuntu12.10-log]Table 'freeswitch.basic_calls'
> > doesn't exist
> > Would a make clean be any different?
> > -Avi
> >
> > On Thu, Aug 11, 2011 at 11:04 PM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> i see a small way it may happen, update again and see if it works.
> >>
> >>
> >> On Thu, Aug 11, 2011 at 9:30 AM, mazilo <Nabble at slickdeals.endjunk.com>
> >> wrote:
> >> > Avi,
> >> >
> >> > I also just upgraded FS since .. 03/26/2011 ago. Now, my Seagate
> >> > DockStar is
> >> > hosting the same FS version you have. Notice that I don't know much
> >> > about
> >> > using ODBC, but my current FS version has been compiled with the
> >> > /--enable-core-odbc-support/ option. So, I haven't done anything to
> >> > create
> >> > any SQL database to use with my FS. However, then I pressed F4 key
> while
> >> > I
> >> > was on a call to ext. 3000 (conference call), it showed me with the
> call
> >> > in
> >> > progress and didn't show me the error message as you posted above. I
> >> > don't
> >> > know if having no SQL database file has anything to do with it.
> >> >
> >> > -----
> >> > FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes
> 3
> >> > Watts of electricity.
> >> > --
> >> > View this message in context:
> >> >
> http://freeswitch-users.2379917.n2.nabble.com/odbc-basic-calls-tp6676013p6676584.html
> >> > Sent from the freeswitch-users mailing list archive at Nabble.com.
> >> >
> >> > _______________________________________________
> >> > Join us at ClueCon 2011, Aug 9-11, Chicago
> >> > http://www.cluecon.com 877-7-4ACLUE
> >> >
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> 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
> >> googletalk:conf+888 at conference.freeswitch.org
> >> pstn:+19193869900
> >>
> >> _______________________________________________
> >> Join us at ClueCon 2011, Aug 9-11, Chicago
> >> http://www.cluecon.com 877-7-4ACLUE
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110812/0700a621/attachment-0001.html 

From anthony.minessale at gmail.com  Fri Aug 12 02:32:42 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 11 Aug 2011 17:32:42 -0500
Subject: [Freeswitch-users] collect dtmf and store in channel variable
In-Reply-To: <CALNZuwkBHptWyKCWYb=X-ApcP79BxFWeySCq7SEKvFJhMdtDNQ@mail.gmail.com>
References: <CAC-LwPOmcTxMZYUZKny2mfryY1MVvqJ9mWrtqS+VGV+G_cis2w@mail.gmail.com>
	<23723.1313079736@ccs.covici.com>
	<CAC-LwPOSH8FxVoJB1PejS=+j==JY515u=_FDFo7M_0LeAG32nA@mail.gmail.com>
	<CALNZuw=kwNEcAaN+-oxrkOM4OxWiAc2Koryct1YbuhkMA28_MA@mail.gmail.com>
	<CAC-LwPOpgRcwhAm1Q3Y9VQ+X5qn+Wx1tQGxzesOsWm7EPJCKvQ@mail.gmail.com>
	<CALNZuw=ncGwazLjZ-gRdG1=L1oa_0p_26sE_j7mzj0u5DkT0DQ@mail.gmail.com>
	<CAC-LwPP9aWzJs1C9C-t5f+oqSZcf3rF0WSaFN4cSWZ+J29+5ug@mail.gmail.com>
	<CALNZuwnF7hoysisyo2TQ40TFc17sfxk_0AH7PgBHwLzfnrqT+w@mail.gmail.com>
	<CAC-LwPP8_SgoLwjoYKec6buMN0Jz14=F86AS0QpTJfRobXTesQ@mail.gmail.com>
	<CALNZuwkBHptWyKCWYb=X-ApcP79BxFWeySCq7SEKvFJhMdtDNQ@mail.gmail.com>
Message-ID: <CAKbxfG8A3XNoEhXfQwWNPwEjCKtU=sY_xWUpAEABZbzcqyxdNg@mail.gmail.com>

digits_dialed variable available in the cdr will contain any digits
dialed the whole call duration.


On Thu, Aug 11, 2011 at 3:15 PM, Avi Marcus <avi at avimarcus.net> wrote:
> According to the wiki, apparently, you can't capture digits from
> bind_digit_action. So you'll have to use that to trigger
> a?Dialplan_Tools_play_and_get_digits?which automatically sets a variable, or
> getdigits within lua/js (only?) and then save it.
>
> -Avi
>
>
> On Thu, Aug 11, 2011 at 10:18 PM, vip killa <vipkilla at gmail.com> wrote:
>>
>> I can't seem to figure out how to store what is collected
>> from?bind_digit_action into a channel variable... any ideas?
>>
>> On Thu, Aug 11, 2011 at 3:07 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>>
>>> So some sort of bind_digit_action to collect and use that to set a hangup
>>> hook. I then execute_extension from the action will be the easiest way to do
>>> it, because you have several actions to accomplish.
>>> However, doing this while the other person is on the hook.. sounds like a
>>> bad idea.
>>> -Avi
>>>
>>> On Thu, Aug 11, 2011 at 9:13 PM, vip killa <vipkilla at gmail.com> wrote:
>>>>
>>>> thanks but i'm not interested in paying someone to do this. but yes i
>>>> simply want to collect the digits during the call from the B leg then store
>>>> them in a variable which will rename the file accordingly from the hangup
>>>> hook
>>>>
>>>> On Thu, Aug 11, 2011 at 2:07 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>>>>
>>>>> Oh, you don't want the info in the CDR, you want the info in the
>>>>> recording. Does it have to be in the name, or is an audio tag enough? I'm
>>>>> not sure if tags can work once the recording started...
>>>>> OK, so you can set a hangup hook to rename the file and add the
>>>>> variable with the account code to the end.
>>>>> Whatever you use to capture the DTMF - bind_digit_action or
>>>>> playandgetdigits or anything can set a variable.
>>>>> -Avi Marcus
>>>>>
>>>>> p.s. If you want me to... actually do it for you, I'm available for
>>>>> consulting.. email me offlist.
>>>>>
>>>>> On Thu, Aug 11, 2011 at 8:52 PM, vip killa <vipkilla at gmail.com> wrote:
>>>>>>
>>>>>> I apologize but the caller would not know the account number, it would
>>>>>> be for internal use only. how could xml_cdrs and mod_cdr_csv be used to
>>>>>> stamp a recording? how do you store the DTMF in a channel variable?
>>>>>>
>>>>>> On Thu, Aug 11, 2011 at 12:41 PM, Avi Marcus <avi at avimarcus.net>
>>>>>> wrote:
>>>>>>>
>>>>>>> You can set any channel variable from the dialplan or from a lua
>>>>>>> script.
>>>>>>> I'd reccommend a standalone lua script before the agent gets on the
>>>>>>> phone, which can verify their account number and then present it to the
>>>>>>> agent even. But yes, you can do anything with the dtmf including setting a
>>>>>>> variable which can be saved from the xml_cdrs or mod_cdr_csv or whatever
>>>>>>> other cdr system you use.
>>>>>>> -Avi
>>>>>>>
>>>>>>> On Thu, Aug 11, 2011 at 7:30 PM, vip killa <vipkilla at gmail.com>
>>>>>>> wrote:
>>>>>>>>
>>>>>>>> indeed it may be easier but this is what the client is asking
>>>>>>>> for.... i know it has to be possible... i just need some direction.
>>>>>>>>
>>>>>>>> On Thu, Aug 11, 2011 at 12:22 PM, <covici at ccs.covici.com> wrote:
>>>>>>>>>
>>>>>>>>> Could not the agent just type it on a screen -- it would seem to be
>>>>>>>>> much
>>>>>>>>> easier.
>>>>>>>>>
>>>>>>>>> vip killa <vipkilla at gmail.com> wrote:
>>>>>>>>>
>>>>>>>>> > Hi everyone,
>>>>>>>>> > I'm trying to collect DTMF digits and store them in a channel
>>>>>>>>> > variable so
>>>>>>>>> > when the channel hangs up it uses those digits to "mark" (or
>>>>>>>>> > rename) the
>>>>>>>>> > recording of the call. The DTMF will be entered by the called
>>>>>>>>> > party (i think
>>>>>>>>> > that would be the B-leg?). I've been experimenting with
>>>>>>>>> > "bind_meta_app" and
>>>>>>>>> > "bind_digit_action", it seems like "bind_digit_action" may be the
>>>>>>>>> > one i need
>>>>>>>>> > to use but im not sure. I'll explain the scenario to make things
>>>>>>>>> > more
>>>>>>>>> > clear...we are trying to install this in a call center type
>>>>>>>>> > environment
>>>>>>>>> > where all calls are being recorded. A caller gets an agent, the
>>>>>>>>> > caller gives
>>>>>>>>> > the agent the account number of their case, the agent uses DTMF
>>>>>>>>> > to mark the
>>>>>>>>> > recording of the call with the account number of the caller's
>>>>>>>>> > case. Does
>>>>>>>>> > that make sense? Please let me know if this is possible, thanks.
>>>>>>>>> >
>>>>>>>>> > ----------------------------------------------------
>>>>>>>>> > Alternatives:
>>>>>>>>> >
>>>>>>>>> > ----------------------------------------------------
>>>>>>>>> > _______________________________________________
>>>>>>>>> > Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>> > http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>> >
>>>>>>>>> > FreeSWITCH-users mailing list
>>>>>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> >
>>>>>>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> > http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>> --
>>>>>>>>> Your life is like a penny. ?You're going to lose it. ?The question
>>>>>>>>> is:
>>>>>>>>> How do
>>>>>>>>> you spend it?
>>>>>>>>>
>>>>>>>>> ? ? ? ? John Covici
>>>>>>>>> ? ? ? ? covici at ccs.covici.com
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From anthony.minessale at gmail.com  Fri Aug 12 02:33:45 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 11 Aug 2011 17:33:45 -0500
Subject: [Freeswitch-users] odbc basic_calls
In-Reply-To: <CALNZuwkh5cQd2qP=xhj3jN7SF81E0x49JYRR1pU--7rCbXrp7Q@mail.gmail.com>
References: <CALNZuwk20r8=rmu9rOOd-aChOS6XOGVm7mxKNhd1-DnC+id7Hg@mail.gmail.com>
	<1313073015901-6676584.post@n2.nabble.com>
	<CAKbxfG_RcTt6Uw44C8QB9BE27F7mSuVqxiu=m5gbXX7Y6p0Ofw@mail.gmail.com>
	<CALNZuwm9aAsH9z=hZgAHcQCx8f=4kNAw+s1h-2mk=EmhMJxQrw@mail.gmail.com>
	<CAKbxfG8HCdPAhSFuF30dw-q1N8XXVD=GADCqJPhzw6CZ+14Tbg@mail.gmail.com>
	<CALNZuwkh5cQd2qP=xhj3jN7SF81E0x49JYRR1pU--7rCbXrp7Q@mail.gmail.com>
Message-ID: <CAKbxfG-qCbZ3p9U4hpgVbpjL+Yj6eTGwiuH3Qio=4OCEpW1=mw@mail.gmail.com>

the create stmt for the view must be offending it.
Maybe you can dig up the error in your logs?


On Thu, Aug 11, 2011 at 5:18 PM, Avi Marcus <avi at avimarcus.net> wrote:
> I didn't think to try it twice. But no,
> freeswitch at default> show calls
> -ERR SQL Error [STATE: 42S02 CODE 1146 ERROR: [unixODBC][MySQL][ODBC 3.51
> Driver][mysqld-5.1.41-3ubuntu12.10-log]Table 'freeswitch.basic_calls'
> doesn't exist
> ]
> freeswitch at default> 2011-08-12 01:17:40.494779 [ERR] switch_core_sqldb.c:825
> ERR: [select * from basic_calls where hostname='sip2' order by
> call_created_epoch]
> [STATE: 42S02 CODE 1146 ERROR: [unixODBC][MySQL][ODBC 3.51
> Driver][mysqld-5.1.41-3ubuntu12.10-log]Table 'freeswitch.basic_calls'
> doesn't exist
> ]
> show calls
> -ERR SQL Error [STATE: 42S02 CODE 1146 ERROR: [unixODBC][MySQL][ODBC 3.51
> Driver][mysqld-5.1.41-3ubuntu12.10-log]Table 'freeswitch.basic_calls'
> doesn't exist
> ]
> freeswitch at default> 2011-08-12 01:17:41.314777 [ERR] switch_core_sqldb.c:825
> ERR: [select * from basic_calls where hostname='sip2' order by
> call_created_epoch]
> [STATE: 42S02 CODE 1146 ERROR: [unixODBC][MySQL][ODBC 3.51
> Driver][mysqld-5.1.41-3ubuntu12.10-log]Table 'freeswitch.basic_calls'
> doesn't exist
>
>
>
>
> -Avi
>
> On Fri, Aug 12, 2011 at 1:12 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> that's good, the error should be what triggered the creation of the view,
>>
>> On Thu, Aug 11, 2011 at 3:22 PM, Avi Marcus <avi at avimarcus.net> wrote:
>> > I did 1) git pull 2) make sync, rather than doing a make clean, 3)
>> > restarted
>> > and 4) got:
>> > 2011-08-11 23:20:45.654766 [ERR] switch_core_sqldb.c:825 ERR: [select *
>> > from
>> > basic_calls where hostname='sip2' order by call_created_epoch]
>> > [STATE: 42S02 CODE 1146 ERROR: [unixODBC][MySQL][ODBC 3.51
>> > Driver][mysqld-5.1.41-3ubuntu12.10-log]Table 'freeswitch.basic_calls'
>> > doesn't exist
>> > Would a make clean be any different?
>> > -Avi
>> >
>> > On Thu, Aug 11, 2011 at 11:04 PM, Anthony Minessale
>> > <anthony.minessale at gmail.com> wrote:
>> >>
>> >> i see a small way it may happen, update again and see if it works.
>> >>
>> >>
>> >> On Thu, Aug 11, 2011 at 9:30 AM, mazilo <Nabble at slickdeals.endjunk.com>
>> >> wrote:
>> >> > Avi,
>> >> >
>> >> > I also just upgraded FS since .. 03/26/2011 ago. Now, my Seagate
>> >> > DockStar is
>> >> > hosting the same FS version you have. Notice that I don't know much
>> >> > about
>> >> > using ODBC, but my current FS version has been compiled with the
>> >> > /--enable-core-odbc-support/ option. So, I haven't done anything to
>> >> > create
>> >> > any SQL database to use with my FS. However, then I pressed F4 key
>> >> > while
>> >> > I
>> >> > was on a call to ext. 3000 (conference call), it showed me with the
>> >> > call
>> >> > in
>> >> > progress and didn't show me the error message as you posted above. I
>> >> > don't
>> >> > know if having no SQL database file has anything to do with it.
>> >> >
>> >> > -----
>> >> > FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY
>> >> > consumes 3
>> >> > Watts of electricity.
>> >> > --
>> >> > View this message in context:
>> >> >
>> >> > http://freeswitch-users.2379917.n2.nabble.com/odbc-basic-calls-tp6676013p6676584.html
>> >> > Sent from the freeswitch-users mailing list archive at Nabble.com.
>> >> >
>> >> > _______________________________________________
>> >> > Join us at ClueCon 2011, Aug 9-11, Chicago
>> >> > http://www.cluecon.com 877-7-4ACLUE
>> >> >
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >>
>> >>
>> >>
>> >> --
>> >> Anthony Minessale II
>> >>
>> >> 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
>> >> googletalk:conf+888 at conference.freeswitch.org
>> >> pstn:+19193869900
>> >>
>> >> _______________________________________________
>> >> Join us at ClueCon 2011, Aug 9-11, Chicago
>> >> http://www.cluecon.com 877-7-4ACLUE
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > Join us at ClueCon 2011, Aug 9-11, Chicago
>> > http://www.cluecon.com 877-7-4ACLUE
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> 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
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From avi at avimarcus.net  Fri Aug 12 02:42:15 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 12 Aug 2011 01:42:15 +0300
Subject: [Freeswitch-users] odbc basic_calls
In-Reply-To: <CAKbxfG-qCbZ3p9U4hpgVbpjL+Yj6eTGwiuH3Qio=4OCEpW1=mw@mail.gmail.com>
References: <CALNZuwk20r8=rmu9rOOd-aChOS6XOGVm7mxKNhd1-DnC+id7Hg@mail.gmail.com>
	<1313073015901-6676584.post@n2.nabble.com>
	<CAKbxfG_RcTt6Uw44C8QB9BE27F7mSuVqxiu=m5gbXX7Y6p0Ofw@mail.gmail.com>
	<CALNZuwm9aAsH9z=hZgAHcQCx8f=4kNAw+s1h-2mk=EmhMJxQrw@mail.gmail.com>
	<CAKbxfG8HCdPAhSFuF30dw-q1N8XXVD=GADCqJPhzw6CZ+14Tbg@mail.gmail.com>
	<CALNZuwkh5cQd2qP=xhj3jN7SF81E0x49JYRR1pU--7rCbXrp7Q@mail.gmail.com>
	<CAKbxfG-qCbZ3p9U4hpgVbpjL+Yj6eTGwiuH3Qio=4OCEpW1=mw@mail.gmail.com>
Message-ID: <CALNZuwkhBhA3U-ykA3G_YoPic41uwpLMWZ-q+7KmMQJvxu8LCQ@mail.gmail.com>

Where would those logs be?
A "grep -ir basic_calls /var/logs/*" returned nothing and I don't see how to
get beyond /log 7, although I'm sure I've done it before.
-Avi


On Fri, Aug 12, 2011 at 1:33 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> the create stmt for the view must be offending it.
> Maybe you can dig up the error in your logs?
>
>
> On Thu, Aug 11, 2011 at 5:18 PM, Avi Marcus <avi at avimarcus.net> wrote:
> > I didn't think to try it twice. But no,
> > freeswitch at default> show calls
> > -ERR SQL Error [STATE: 42S02 CODE 1146 ERROR: [unixODBC][MySQL][ODBC 3.51
> > Driver][mysqld-5.1.41-3ubuntu12.10-log]Table 'freeswitch.basic_calls'
> > doesn't exist
> > ]
> > freeswitch at default> 2011-08-12 01:17:40.494779 [ERR]
> switch_core_sqldb.c:825
> > ERR: [select * from basic_calls where hostname='sip2' order by
> > call_created_epoch]
> > [STATE: 42S02 CODE 1146 ERROR: [unixODBC][MySQL][ODBC 3.51
> > Driver][mysqld-5.1.41-3ubuntu12.10-log]Table 'freeswitch.basic_calls'
> > doesn't exist
> > ]
> > show calls
> > -ERR SQL Error [STATE: 42S02 CODE 1146 ERROR: [unixODBC][MySQL][ODBC 3.51
> > Driver][mysqld-5.1.41-3ubuntu12.10-log]Table 'freeswitch.basic_calls'
> > doesn't exist
> > ]
> > freeswitch at default> 2011-08-12 01:17:41.314777 [ERR]
> switch_core_sqldb.c:825
> > ERR: [select * from basic_calls where hostname='sip2' order by
> > call_created_epoch]
> > [STATE: 42S02 CODE 1146 ERROR: [unixODBC][MySQL][ODBC 3.51
> > Driver][mysqld-5.1.41-3ubuntu12.10-log]Table 'freeswitch.basic_calls'
> > doesn't exist
> >
> >
> >
> >
> > -Avi
> >
> > On Fri, Aug 12, 2011 at 1:12 AM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> that's good, the error should be what triggered the creation of the
> view,
> >>
> >> On Thu, Aug 11, 2011 at 3:22 PM, Avi Marcus <avi at avimarcus.net> wrote:
> >> > I did 1) git pull 2) make sync, rather than doing a make clean, 3)
> >> > restarted
> >> > and 4) got:
> >> > 2011-08-11 23:20:45.654766 [ERR] switch_core_sqldb.c:825 ERR: [select
> *
> >> > from
> >> > basic_calls where hostname='sip2' order by call_created_epoch]
> >> > [STATE: 42S02 CODE 1146 ERROR: [unixODBC][MySQL][ODBC 3.51
> >> > Driver][mysqld-5.1.41-3ubuntu12.10-log]Table 'freeswitch.basic_calls'
> >> > doesn't exist
> >> > Would a make clean be any different?
> >> > -Avi
> >> >
> >> > On Thu, Aug 11, 2011 at 11:04 PM, Anthony Minessale
> >> > <anthony.minessale at gmail.com> wrote:
> >> >>
> >> >> i see a small way it may happen, update again and see if it works.
> >> >>
> >> >>
> >> >> On Thu, Aug 11, 2011 at 9:30 AM, mazilo <
> Nabble at slickdeals.endjunk.com>
> >> >> wrote:
> >> >> > Avi,
> >> >> >
> >> >> > I also just upgraded FS since .. 03/26/2011 ago. Now, my Seagate
> >> >> > DockStar is
> >> >> > hosting the same FS version you have. Notice that I don't know much
> >> >> > about
> >> >> > using ODBC, but my current FS version has been compiled with the
> >> >> > /--enable-core-odbc-support/ option. So, I haven't done anything to
> >> >> > create
> >> >> > any SQL database to use with my FS. However, then I pressed F4 key
> >> >> > while
> >> >> > I
> >> >> > was on a call to ext. 3000 (conference call), it showed me with the
> >> >> > call
> >> >> > in
> >> >> > progress and didn't show me the error message as you posted above.
> I
> >> >> > don't
> >> >> > know if having no SQL database file has anything to do with it.
> >> >> >
> >> >> > -----
> >> >> > FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY
> >> >> > consumes 3
> >> >> > Watts of electricity.
> >> >> > --
> >> >> > View this message in context:
> >> >> >
> >> >> >
> http://freeswitch-users.2379917.n2.nabble.com/odbc-basic-calls-tp6676013p6676584.html
> >> >> > Sent from the freeswitch-users mailing list archive at Nabble.com.
> >> >> >
> >> >> > _______________________________________________
> >> >> > Join us at ClueCon 2011, Aug 9-11, Chicago
> >> >> > http://www.cluecon.com 877-7-4ACLUE
> >> >> >
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >> >
> >> >>
> >> >>
> >> >>
> >> >> --
> >> >> Anthony Minessale II
> >> >>
> >> >> 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
> >> >> googletalk:conf+888 at conference.freeswitch.org
> >> >> pstn:+19193869900
> >> >>
> >> >> _______________________________________________
> >> >> Join us at ClueCon 2011, Aug 9-11, Chicago
> >> >> http://www.cluecon.com 877-7-4ACLUE
> >> >>
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > Join us at ClueCon 2011, Aug 9-11, Chicago
> >> > http://www.cluecon.com 877-7-4ACLUE
> >> >
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> 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
> >> googletalk:conf+888 at conference.freeswitch.org
> >> pstn:+19193869900
> >>
> >> _______________________________________________
> >> Join us at ClueCon 2011, Aug 9-11, Chicago
> >> http://www.cluecon.com 877-7-4ACLUE
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110812/5493d97f/attachment-0001.html 

From anthony.minessale at gmail.com  Fri Aug 12 02:45:45 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 11 Aug 2011 17:45:45 -0500
Subject: [Freeswitch-users] odbc basic_calls
In-Reply-To: <CALNZuwkhBhA3U-ykA3G_YoPic41uwpLMWZ-q+7KmMQJvxu8LCQ@mail.gmail.com>
References: <CALNZuwk20r8=rmu9rOOd-aChOS6XOGVm7mxKNhd1-DnC+id7Hg@mail.gmail.com>
	<1313073015901-6676584.post@n2.nabble.com>
	<CAKbxfG_RcTt6Uw44C8QB9BE27F7mSuVqxiu=m5gbXX7Y6p0Ofw@mail.gmail.com>
	<CALNZuwm9aAsH9z=hZgAHcQCx8f=4kNAw+s1h-2mk=EmhMJxQrw@mail.gmail.com>
	<CAKbxfG8HCdPAhSFuF30dw-q1N8XXVD=GADCqJPhzw6CZ+14Tbg@mail.gmail.com>
	<CALNZuwkh5cQd2qP=xhj3jN7SF81E0x49JYRR1pU--7rCbXrp7Q@mail.gmail.com>
	<CAKbxfG-qCbZ3p9U4hpgVbpjL+Yj6eTGwiuH3Qio=4OCEpW1=mw@mail.gmail.com>
	<CALNZuwkhBhA3U-ykA3G_YoPic41uwpLMWZ-q+7KmMQJvxu8LCQ@mail.gmail.com>
Message-ID: <CAKbxfG9u5TPEZFmPDMYyY1MhX9wnjf_cp=hTytEbZRi3Lo=LrA@mail.gmail.com>

Maybe MySQL has a log of its own somewhere?
 On Aug 11, 2011 5:43 PM, "Avi Marcus" <avi at avimarcus.net> wrote:
> Where would those logs be?
> A "grep -ir basic_calls /var/logs/*" returned nothing and I don't see how
to
> get beyond /log 7, although I'm sure I've done it before.
> -Avi
>
>
> On Fri, Aug 12, 2011 at 1:33 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> the create stmt for the view must be offending it.
>> Maybe you can dig up the error in your logs?
>>
>>
>> On Thu, Aug 11, 2011 at 5:18 PM, Avi Marcus <avi at avimarcus.net> wrote:
>> > I didn't think to try it twice. But no,
>> > freeswitch at default> show calls
>> > -ERR SQL Error [STATE: 42S02 CODE 1146 ERROR: [unixODBC][MySQL][ODBC
3.51
>> > Driver][mysqld-5.1.41-3ubuntu12.10-log]Table 'freeswitch.basic_calls'
>> > doesn't exist
>> > ]
>> > freeswitch at default> 2011-08-12 01:17:40.494779 [ERR]
>> switch_core_sqldb.c:825
>> > ERR: [select * from basic_calls where hostname='sip2' order by
>> > call_created_epoch]
>> > [STATE: 42S02 CODE 1146 ERROR: [unixODBC][MySQL][ODBC 3.51
>> > Driver][mysqld-5.1.41-3ubuntu12.10-log]Table 'freeswitch.basic_calls'
>> > doesn't exist
>> > ]
>> > show calls
>> > -ERR SQL Error [STATE: 42S02 CODE 1146 ERROR: [unixODBC][MySQL][ODBC
3.51
>> > Driver][mysqld-5.1.41-3ubuntu12.10-log]Table 'freeswitch.basic_calls'
>> > doesn't exist
>> > ]
>> > freeswitch at default> 2011-08-12 01:17:41.314777 [ERR]
>> switch_core_sqldb.c:825
>> > ERR: [select * from basic_calls where hostname='sip2' order by
>> > call_created_epoch]
>> > [STATE: 42S02 CODE 1146 ERROR: [unixODBC][MySQL][ODBC 3.51
>> > Driver][mysqld-5.1.41-3ubuntu12.10-log]Table 'freeswitch.basic_calls'
>> > doesn't exist
>> >
>> >
>> >
>> >
>> > -Avi
>> >
>> > On Fri, Aug 12, 2011 at 1:12 AM, Anthony Minessale
>> > <anthony.minessale at gmail.com> wrote:
>> >>
>> >> that's good, the error should be what triggered the creation of the
>> view,
>> >>
>> >> On Thu, Aug 11, 2011 at 3:22 PM, Avi Marcus <avi at avimarcus.net> wrote:
>> >> > I did 1) git pull 2) make sync, rather than doing a make clean, 3)
>> >> > restarted
>> >> > and 4) got:
>> >> > 2011-08-11 23:20:45.654766 [ERR] switch_core_sqldb.c:825 ERR:
[select
>> *
>> >> > from
>> >> > basic_calls where hostname='sip2' order by call_created_epoch]
>> >> > [STATE: 42S02 CODE 1146 ERROR: [unixODBC][MySQL][ODBC 3.51
>> >> > Driver][mysqld-5.1.41-3ubuntu12.10-log]Table
'freeswitch.basic_calls'
>> >> > doesn't exist
>> >> > Would a make clean be any different?
>> >> > -Avi
>> >> >
>> >> > On Thu, Aug 11, 2011 at 11:04 PM, Anthony Minessale
>> >> > <anthony.minessale at gmail.com> wrote:
>> >> >>
>> >> >> i see a small way it may happen, update again and see if it works.
>> >> >>
>> >> >>
>> >> >> On Thu, Aug 11, 2011 at 9:30 AM, mazilo <
>> Nabble at slickdeals.endjunk.com>
>> >> >> wrote:
>> >> >> > Avi,
>> >> >> >
>> >> >> > I also just upgraded FS since .. 03/26/2011 ago. Now, my Seagate
>> >> >> > DockStar is
>> >> >> > hosting the same FS version you have. Notice that I don't know
much
>> >> >> > about
>> >> >> > using ODBC, but my current FS version has been compiled with the
>> >> >> > /--enable-core-odbc-support/ option. So, I haven't done anything
to
>> >> >> > create
>> >> >> > any SQL database to use with my FS. However, then I pressed F4
key
>> >> >> > while
>> >> >> > I
>> >> >> > was on a call to ext. 3000 (conference call), it showed me with
the
>> >> >> > call
>> >> >> > in
>> >> >> > progress and didn't show me the error message as you posted
above.
>> I
>> >> >> > don't
>> >> >> > know if having no SQL database file has anything to do with it.
>> >> >> >
>> >> >> > -----
>> >> >> > FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY
>> >> >> > consumes 3
>> >> >> > Watts of electricity.
>> >> >> > --
>> >> >> > View this message in context:
>> >> >> >
>> >> >> >
>>
http://freeswitch-users.2379917.n2.nabble.com/odbc-basic-calls-tp6676013p6676584.html
>> >> >> > Sent from the freeswitch-users mailing list archive at
Nabble.com.
>> >> >> >
>> >> >> > _______________________________________________
>> >> >> > Join us at ClueCon 2011, Aug 9-11, Chicago
>> >> >> > http://www.cluecon.com 877-7-4ACLUE
>> >> >> >
>> >> >> > FreeSWITCH-users mailing list
>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >
>> >> >> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> > http://www.freeswitch.org
>> >> >> >
>> >> >>
>> >> >>
>> >> >>
>> >> >> --
>> >> >> Anthony Minessale II
>> >> >>
>> >> >> 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
>> >> >> googletalk:conf+888 at conference.freeswitch.org
>> >> >> pstn:+19193869900
>> >> >>
>> >> >> _______________________________________________
>> >> >> Join us at ClueCon 2011, Aug 9-11, Chicago
>> >> >> http://www.cluecon.com 877-7-4ACLUE
>> >> >>
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > Join us at ClueCon 2011, Aug 9-11, Chicago
>> >> > http://www.cluecon.com 877-7-4ACLUE
>> >> >
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >>
>> >>
>> >> --
>> >> Anthony Minessale II
>> >>
>> >> 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
>> >> googletalk:conf+888 at conference.freeswitch.org
>> >> pstn:+19193869900
>> >>
>> >> _______________________________________________
>> >> Join us at ClueCon 2011, Aug 9-11, Chicago
>> >> http://www.cluecon.com 877-7-4ACLUE
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > Join us at ClueCon 2011, Aug 9-11, Chicago
>> > http://www.cluecon.com 877-7-4ACLUE
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> 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
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110811/fda5e1b0/attachment.html 

From avi at avimarcus.net  Fri Aug 12 02:58:11 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 12 Aug 2011 01:58:11 +0300
Subject: [Freeswitch-users] odbc basic_calls
In-Reply-To: <CAKbxfG9u5TPEZFmPDMYyY1MhX9wnjf_cp=hTytEbZRi3Lo=LrA@mail.gmail.com>
References: <CALNZuwk20r8=rmu9rOOd-aChOS6XOGVm7mxKNhd1-DnC+id7Hg@mail.gmail.com>
	<1313073015901-6676584.post@n2.nabble.com>
	<CAKbxfG_RcTt6Uw44C8QB9BE27F7mSuVqxiu=m5gbXX7Y6p0Ofw@mail.gmail.com>
	<CALNZuwm9aAsH9z=hZgAHcQCx8f=4kNAw+s1h-2mk=EmhMJxQrw@mail.gmail.com>
	<CAKbxfG8HCdPAhSFuF30dw-q1N8XXVD=GADCqJPhzw6CZ+14Tbg@mail.gmail.com>
	<CALNZuwkh5cQd2qP=xhj3jN7SF81E0x49JYRR1pU--7rCbXrp7Q@mail.gmail.com>
	<CAKbxfG-qCbZ3p9U4hpgVbpjL+Yj6eTGwiuH3Qio=4OCEpW1=mw@mail.gmail.com>
	<CALNZuwkhBhA3U-ykA3G_YoPic41uwpLMWZ-q+7KmMQJvxu8LCQ@mail.gmail.com>
	<CAKbxfG9u5TPEZFmPDMYyY1MhX9wnjf_cp=hTytEbZRi3Lo=LrA@mail.gmail.com>
Message-ID: <CALNZuwkH5V7w4bQ5=xYhO-SNFQyj3rhsTR3QkH3owuXWtCfqHQ@mail.gmail.com>

I turned on verbose logging which is supposed to show failed error too, but
I've got nothing.
Perhaps the create isn't being triggered? Is there a way to see that debug
in the fs_cli somehow?

-Avi


On Fri, Aug 12, 2011 at 1:45 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Maybe MySQL has a log of its own somewhere?
>  On Aug 11, 2011 5:43 PM, "Avi Marcus" <avi at avimarcus.net> wrote:
> > Where would those logs be?
> > A "grep -ir basic_calls /var/logs/*" returned nothing and I don't see how
> to
> > get beyond /log 7, although I'm sure I've done it before.
> > -Avi
> >
> >
> > On Fri, Aug 12, 2011 at 1:33 AM, Anthony Minessale <
> > anthony.minessale at gmail.com> wrote:
> >
> >> the create stmt for the view must be offending it.
> >> Maybe you can dig up the error in your logs?
> >>
> >>
> >> On Thu, Aug 11, 2011 at 5:18 PM, Avi Marcus <avi at avimarcus.net> wrote:
> >> > I didn't think to try it twice. But no,
> >> > freeswitch at default> show calls
> >> > -ERR SQL Error [STATE: 42S02 CODE 1146 ERROR: [unixODBC][MySQL][ODBC
> 3.51
> >> > Driver][mysqld-5.1.41-3ubuntu12.10-log]Table 'freeswitch.basic_calls'
> >> > doesn't exist
> >> > ]
> >> > freeswitch at default> 2011-08-12 01:17:40.494779 [ERR]
> >> switch_core_sqldb.c:825
> >> > ERR: [select * from basic_calls where hostname='sip2' order by
> >> > call_created_epoch]
> >> > [STATE: 42S02 CODE 1146 ERROR: [unixODBC][MySQL][ODBC 3.51
> >> > Driver][mysqld-5.1.41-3ubuntu12.10-log]Table 'freeswitch.basic_calls'
> >> > doesn't exist
> >> > ]
> >> > show calls
> >> > -ERR SQL Error [STATE: 42S02 CODE 1146 ERROR: [unixODBC][MySQL][ODBC
> 3.51
> >> > Driver][mysqld-5.1.41-3ubuntu12.10-log]Table 'freeswitch.basic_calls'
> >> > doesn't exist
> >> > ]
> >> > freeswitch at default> 2011-08-12 01:17:41.314777 [ERR]
> >> switch_core_sqldb.c:825
> >> > ERR: [select * from basic_calls where hostname='sip2' order by
> >> > call_created_epoch]
> >> > [STATE: 42S02 CODE 1146 ERROR: [unixODBC][MySQL][ODBC 3.51
> >> > Driver][mysqld-5.1.41-3ubuntu12.10-log]Table 'freeswitch.basic_calls'
> >> > doesn't exist
> >> >
> >> >
> >> >
> >> >
> >> > -Avi
> >> >
> >> > On Fri, Aug 12, 2011 at 1:12 AM, Anthony Minessale
> >> > <anthony.minessale at gmail.com> wrote:
> >> >>
> >> >> that's good, the error should be what triggered the creation of the
> >> view,
> >> >>
> >> >> On Thu, Aug 11, 2011 at 3:22 PM, Avi Marcus <avi at avimarcus.net>
> wrote:
> >> >> > I did 1) git pull 2) make sync, rather than doing a make clean, 3)
> >> >> > restarted
> >> >> > and 4) got:
> >> >> > 2011-08-11 23:20:45.654766 [ERR] switch_core_sqldb.c:825 ERR:
> [select
> >> *
> >> >> > from
> >> >> > basic_calls where hostname='sip2' order by call_created_epoch]
> >> >> > [STATE: 42S02 CODE 1146 ERROR: [unixODBC][MySQL][ODBC 3.51
> >> >> > Driver][mysqld-5.1.41-3ubuntu12.10-log]Table
> 'freeswitch.basic_calls'
> >> >> > doesn't exist
> >> >> > Would a make clean be any different?
> >> >> > -Avi
> >> >> >
> >> >> > On Thu, Aug 11, 2011 at 11:04 PM, Anthony Minessale
> >> >> > <anthony.minessale at gmail.com> wrote:
> >> >> >>
> >> >> >> i see a small way it may happen, update again and see if it works.
> >> >> >>
> >> >> >>
> >> >> >> On Thu, Aug 11, 2011 at 9:30 AM, mazilo <
> >> Nabble at slickdeals.endjunk.com>
> >> >> >> wrote:
> >> >> >> > Avi,
> >> >> >> >
> >> >> >> > I also just upgraded FS since .. 03/26/2011 ago. Now, my Seagate
> >> >> >> > DockStar is
> >> >> >> > hosting the same FS version you have. Notice that I don't know
> much
> >> >> >> > about
> >> >> >> > using ODBC, but my current FS version has been compiled with the
> >> >> >> > /--enable-core-odbc-support/ option. So, I haven't done anything
> to
> >> >> >> > create
> >> >> >> > any SQL database to use with my FS. However, then I pressed F4
> key
> >> >> >> > while
> >> >> >> > I
> >> >> >> > was on a call to ext. 3000 (conference call), it showed me with
> the
> >> >> >> > call
> >> >> >> > in
> >> >> >> > progress and didn't show me the error message as you posted
> above.
> >> I
> >> >> >> > don't
> >> >> >> > know if having no SQL database file has anything to do with it.
> >> >> >> >
> >> >> >> > -----
> >> >> >> > FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY
> >> >> >> > consumes 3
> >> >> >> > Watts of electricity.
> >> >> >> > --
> >> >> >> > View this message in context:
> >> >> >> >
> >> >> >> >
> >>
> http://freeswitch-users.2379917.n2.nabble.com/odbc-basic-calls-tp6676013p6676584.html
> >> >> >> > Sent from the freeswitch-users mailing list archive at
> Nabble.com.
> >> >> >> >
> >> >> >> > _______________________________________________
> >> >> >> > Join us at ClueCon 2011, Aug 9-11, Chicago
> >> >> >> > http://www.cluecon.com 877-7-4ACLUE
> >> >> >> >
> >> >> >> > FreeSWITCH-users mailing list
> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >
> >> >> >> > UNSUBSCRIBE:
> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> > http://www.freeswitch.org
> >> >> >> >
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >> --
> >> >> >> Anthony Minessale II
> >> >> >>
> >> >> >> 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
> >> >> >> googletalk:conf+888 at conference.freeswitch.org
> >> >> >> pstn:+19193869900
> >> >> >>
> >> >> >> _______________________________________________
> >> >> >> Join us at ClueCon 2011, Aug 9-11, Chicago
> >> >> >> http://www.cluecon.com 877-7-4ACLUE
> >> >> >>
> >> >> >> FreeSWITCH-users mailing list
> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>
> >> >> >> UNSUBSCRIBE:
> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >> > _______________________________________________
> >> >> > Join us at ClueCon 2011, Aug 9-11, Chicago
> >> >> > http://www.cluecon.com 877-7-4ACLUE
> >> >> >
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> > UNSUBSCRIBE:
> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >>
> >> >>
> >> >>
> >> >> --
> >> >> Anthony Minessale II
> >> >>
> >> >> 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
> >> >> googletalk:conf+888 at conference.freeswitch.org
> >> >> pstn:+19193869900
> >> >>
> >> >> _______________________________________________
> >> >> Join us at ClueCon 2011, Aug 9-11, Chicago
> >> >> http://www.cluecon.com 877-7-4ACLUE
> >> >>
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> UNSUBSCRIBE:
> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > Join us at ClueCon 2011, Aug 9-11, Chicago
> >> > http://www.cluecon.com 877-7-4ACLUE
> >> >
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> 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
> >> googletalk:conf+888 at conference.freeswitch.org
> >> pstn:+19193869900
> >>
> >> _______________________________________________
> >> Join us at ClueCon 2011, Aug 9-11, Chicago
> >> http://www.cluecon.com 877-7-4ACLUE
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110812/fed527ec/attachment-0001.html 

From tculjaga at gmail.com  Fri Aug 12 03:33:07 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 12 Aug 2011 01:33:07 +0200
Subject: [Freeswitch-users] FS performance using ESL
In-Reply-To: <CAKbxfG-SWBzu0dBv_5B0v7710q7vLeeduT9L5V_cnceYHjrGJQ@mail.gmail.com>
References: <CAJoKKdU9VPzOS-C3jHCjG6Y0V-Jo_j7hqYwt6-Zh7ohb1pq3xQ@mail.gmail.com>
	<CAKbxfG-SWBzu0dBv_5B0v7710q7vLeeduT9L5V_cnceYHjrGJQ@mail.gmail.com>
Message-ID: <CAJoKKdUcA=9CvSdRPhaoMopK+f4K2W1DO0=0b5mck_KEjOc9pQ@mail.gmail.com>

Hi Anthony, thanks for your response ...


this is what i have:

        esl_filter(&handle, "unique-id",
esl_event_get_header(handle.info_event, "caller-unique-id"));
        esl_events(&handle, ESL_EVENT_TYPE_PLAIN, "CHANNEL_DATA
CHANNEL_EXECUTE_COMPLETE CHANNEL_HANGUP");

what do you suggest i put there ?


is the inbound method less costly ?




I modified testserver.c just a bit...

#include <sys/types.h>  /* include this before any other sys headers */
#include <sys/wait.h>   /* header for waitpid() and various macros */
#include <signal.h>     /* header for signal functions */
#include <stdio.h>      /* header for fprintf() */
#include <unistd.h>     /* header for fork() */
#include <stdlib.h>
#include <esl.h>

void sig_chld(int);     /* prototype for our SIGCHLD handler */

static void mycallback(esl_socket_t server_sock, esl_socket_t client_sock,
struct sockaddr_in *addr)
{
        esl_handle_t handle = {{0}};
        int done = 0;
        esl_status_t status;
        time_t exp = 0;

        if (fork() != 0) {
                close(client_sock);
                return;
        }

        esl_attach_handle(&handle, client_sock, addr);

        esl_log(ESL_LOG_INFO, "Connected! %d\n", handle.sock);

        esl_filter(&handle, "unique-id",
esl_event_get_header(handle.info_event, "caller-unique-id"));
        esl_events(&handle, ESL_EVENT_TYPE_PLAIN, "CHANNEL_DATA
CHANNEL_EXECUTE_COMPLETE CHANNEL_HANGUP");

        esl_send_recv(&handle, "linger");

        esl_execute(&handle, "answer", NULL, NULL);
        //esl_execute(&handle, "conference", "3000 at default", NULL);
        esl_execute(&handle, "playback", "/home/tculjaga/myWavFile.wav",
NULL);
        //esl_execute(&handle, "sleep", "1000", NULL);
        //esl_execute(&handle, "hangup", NULL, NULL);

        while((status = esl_recv_timed(&handle, 1000)) != ESL_FAIL) {
                if (done) {
                        if (time(NULL) >= exp) {
                                break;
                        }
                } else if (status == ESL_SUCCESS) {
                        const char *type =
esl_event_get_header(handle.last_event, "content-type");
                        if (type && !strcasecmp(type,
"text/disconnect-notice")) {
                                const char *dispo =
esl_event_get_header(handle.last_event, "content-disposition");
                                esl_log(ESL_LOG_INFO, "Got a disconnection
notice dispostion: [%s]\n", dispo ? dispo : "");
                                if (!strcmp(dispo, "linger")) {
                                        done = 1;
                                        esl_log(ESL_LOG_INFO, "Waiting 5
seconds for any remaining events.\n");
                                        exp = time(NULL) + 5;
                                }
                        }
                }
        }

        esl_log(ESL_LOG_INFO, "Disconnected! %d\n", handle.sock);
        esl_disconnect(&handle);

        close(client_sock);

        _exit(0);
}

/*
 * The signal handler function -- only gets called when a SIGCHLD
 * is received, ie when a child terminates
 */
void sig_chld(int signo)
{
    int status;

    /* Wait for any child without blocking */
    if (waitpid(-1, &status, WNOHANG) < 0)
    {
        /*
         * calling standard I/O functions like fprintf() in a
         * signal handler is not recommended, but probably OK
         * in toy programs like this one.
         */
        fprintf(stderr, "waitpid failed\n");
        return;
    }
}

int main(void)
{
        struct sigaction act;

        /* Assign sig_chld as our SIGCHLD handler */
        act.sa_handler = sig_chld;

        /* We don't want to block any other signals in this example */
        sigemptyset(&act.sa_mask);

        /*
         * We're only interested in children that have terminated, not ones
         * which have been stopped (eg user pressing control-Z at terminal)
         */
        act.sa_flags = SA_NOCLDSTOP;

        /*
         * Make these values effective. If we were writing a real
         * application, we would probably save the old value instead of
         * passing NULL.
         */
/*      if (sigaction(SIGCHLD, &act, NULL) < 0)
        {
                fprintf(stderr, "sigaction failed\n");
                return 1;
        }
*/
        signal(SIGCHLD, SIG_IGN);

        esl_global_set_default_logger(0);
        esl_listen("localhost", 8088, mycallback);

        return 0;
}




On Thu, Aug 11, 2011 at 9:59 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> try removing the filter and event subscriptions
> it's costly to consume all of the events especially at 75cps.
>
>
> On Thu, Aug 11, 2011 at 5:23 AM, Tihomir Culjaga <tculjaga at gmail.com>
> wrote:
> > hello,
> >
> > im wondering how much performance do we loose when using ESL instead of
> > running it via dialplan?
> >
> >
> > without ESL with a fine tuned FS and a short dialplan ( answer, playback
> > like 20 seconds file, hangup ) im able to service 75 CPS. On the same FS,
> > when i use ESL to answer the call, playback the same file and hangup, im
> not
> > able to run more than 2 CPS... this is a huge impact and i really can't
> > believe it.
> >
> > I'm using event-socket outbound e.g.:
> >
> > <action application="socket" data="127.0.0.1:8088 async full"/>
> >
> > my extension looks like:
> >
> > <extension name="ESL_C">
> >   <condition field="destination_number" expression="^(6666)$">
> >     <action application="socket" data="127.0.0.1:8088 async full"/>
> >     <action application="sleep" data="1000"/>
> >     <action application="hangup"/>
> >   </condition>
> > </extension>
> >
> >
> > im using testserver from lib/esl/ and i just removed the conference
> command
> > and added the playback one.... also i moved the esl_debug lvl to 0
> >
> >
> > anyhow, FS cannot run more than 2 CPS compared to 75 CPS when the
> playback
> > is done from the dialplan.
> >
> >
> > Please, can someone give me a clue on what is going on?
> > Maybe im doing something wrong?
> > how to get maximum FS performance using ESL ?
> >
> >
> >
> > Regards,
> > Tihomir.
> >
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110812/9ac9b714/attachment.html 

From tculjaga at gmail.com  Fri Aug 12 03:56:06 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 12 Aug 2011 01:56:06 +0200
Subject: [Freeswitch-users] FS performance using ESL
In-Reply-To: <CAJoKKdUcA=9CvSdRPhaoMopK+f4K2W1DO0=0b5mck_KEjOc9pQ@mail.gmail.com>
References: <CAJoKKdU9VPzOS-C3jHCjG6Y0V-Jo_j7hqYwt6-Zh7ohb1pq3xQ@mail.gmail.com>
	<CAKbxfG-SWBzu0dBv_5B0v7710q7vLeeduT9L5V_cnceYHjrGJQ@mail.gmail.com>
	<CAJoKKdUcA=9CvSdRPhaoMopK+f4K2W1DO0=0b5mck_KEjOc9pQ@mail.gmail.com>
Message-ID: <CAJoKKdXR64qNfCYj0uyYFda4YLavwd_NFn3bH6Td+2eSGo7Y1w@mail.gmail.com>

is there any other method than esl to controll calls on FS from an eternal
application?
will mod_curl or mod_xml_curl get better performance?

T.

On Fri, Aug 12, 2011 at 1:33 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> Hi Anthony, thanks for your response ...
>
>
> this is what i have:
>
>         esl_filter(&handle, "unique-id",
> esl_event_get_header(handle.info_event, "caller-unique-id"));
>         esl_events(&handle, ESL_EVENT_TYPE_PLAIN, "CHANNEL_DATA
> CHANNEL_EXECUTE_COMPLETE CHANNEL_HANGUP");
>
> what do you suggest i put there ?
>
>
> is the inbound method less costly ?
>
>
>
>
> I modified testserver.c just a bit...
>
> #include <sys/types.h>  /* include this before any other sys headers */
> #include <sys/wait.h>   /* header for waitpid() and various macros */
> #include <signal.h>     /* header for signal functions */
> #include <stdio.h>      /* header for fprintf() */
> #include <unistd.h>     /* header for fork() */
> #include <stdlib.h>
> #include <esl.h>
>
> void sig_chld(int);     /* prototype for our SIGCHLD handler */
>
> static void mycallback(esl_socket_t server_sock, esl_socket_t client_sock,
> struct sockaddr_in *addr)
> {
>         esl_handle_t handle = {{0}};
>         int done = 0;
>         esl_status_t status;
>         time_t exp = 0;
>
>         if (fork() != 0) {
>                 close(client_sock);
>                 return;
>         }
>
>         esl_attach_handle(&handle, client_sock, addr);
>
>         esl_log(ESL_LOG_INFO, "Connected! %d\n", handle.sock);
>
>         esl_filter(&handle, "unique-id",
> esl_event_get_header(handle.info_event, "caller-unique-id"));
>         esl_events(&handle, ESL_EVENT_TYPE_PLAIN, "CHANNEL_DATA
> CHANNEL_EXECUTE_COMPLETE CHANNEL_HANGUP");
>
>         esl_send_recv(&handle, "linger");
>
>         esl_execute(&handle, "answer", NULL, NULL);
>         //esl_execute(&handle, "conference", "3000 at default", NULL);
>         esl_execute(&handle, "playback", "/home/tculjaga/myWavFile.wav",
> NULL);
>         //esl_execute(&handle, "sleep", "1000", NULL);
>         //esl_execute(&handle, "hangup", NULL, NULL);
>
>         while((status = esl_recv_timed(&handle, 1000)) != ESL_FAIL) {
>                 if (done) {
>                         if (time(NULL) >= exp) {
>                                 break;
>                         }
>                 } else if (status == ESL_SUCCESS) {
>                         const char *type =
> esl_event_get_header(handle.last_event, "content-type");
>                         if (type && !strcasecmp(type,
> "text/disconnect-notice")) {
>                                 const char *dispo =
> esl_event_get_header(handle.last_event, "content-disposition");
>                                 esl_log(ESL_LOG_INFO, "Got a disconnection
> notice dispostion: [%s]\n", dispo ? dispo : "");
>                                 if (!strcmp(dispo, "linger")) {
>                                         done = 1;
>                                         esl_log(ESL_LOG_INFO, "Waiting 5
> seconds for any remaining events.\n");
>                                         exp = time(NULL) + 5;
>                                 }
>                         }
>                 }
>         }
>
>         esl_log(ESL_LOG_INFO, "Disconnected! %d\n", handle.sock);
>         esl_disconnect(&handle);
>
>         close(client_sock);
>
>         _exit(0);
> }
>
> /*
>  * The signal handler function -- only gets called when a SIGCHLD
>  * is received, ie when a child terminates
>  */
> void sig_chld(int signo)
> {
>     int status;
>
>     /* Wait for any child without blocking */
>     if (waitpid(-1, &status, WNOHANG) < 0)
>     {
>         /*
>          * calling standard I/O functions like fprintf() in a
>          * signal handler is not recommended, but probably OK
>          * in toy programs like this one.
>          */
>         fprintf(stderr, "waitpid failed\n");
>         return;
>     }
> }
>
> int main(void)
> {
>         struct sigaction act;
>
>         /* Assign sig_chld as our SIGCHLD handler */
>         act.sa_handler = sig_chld;
>
>         /* We don't want to block any other signals in this example */
>         sigemptyset(&act.sa_mask);
>
>         /*
>          * We're only interested in children that have terminated, not ones
>          * which have been stopped (eg user pressing control-Z at terminal)
>          */
>         act.sa_flags = SA_NOCLDSTOP;
>
>         /*
>          * Make these values effective. If we were writing a real
>          * application, we would probably save the old value instead of
>          * passing NULL.
>          */
> /*      if (sigaction(SIGCHLD, &act, NULL) < 0)
>         {
>                 fprintf(stderr, "sigaction failed\n");
>                 return 1;
>         }
> */
>         signal(SIGCHLD, SIG_IGN);
>
>         esl_global_set_default_logger(0);
>         esl_listen("localhost", 8088, mycallback);
>
>         return 0;
>
> }
>
>
>
>
> On Thu, Aug 11, 2011 at 9:59 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> try removing the filter and event subscriptions
>> it's costly to consume all of the events especially at 75cps.
>>
>>
>> On Thu, Aug 11, 2011 at 5:23 AM, Tihomir Culjaga <tculjaga at gmail.com>
>> wrote:
>> > hello,
>> >
>> > im wondering how much performance do we loose when using ESL instead of
>> > running it via dialplan?
>> >
>> >
>> > without ESL with a fine tuned FS and a short dialplan ( answer, playback
>> > like 20 seconds file, hangup ) im able to service 75 CPS. On the same
>> FS,
>> > when i use ESL to answer the call, playback the same file and hangup, im
>> not
>> > able to run more than 2 CPS... this is a huge impact and i really can't
>> > believe it.
>> >
>> > I'm using event-socket outbound e.g.:
>> >
>> > <action application="socket" data="127.0.0.1:8088 async full"/>
>> >
>> > my extension looks like:
>> >
>> > <extension name="ESL_C">
>> >   <condition field="destination_number" expression="^(6666)$">
>> >     <action application="socket" data="127.0.0.1:8088 async full"/>
>> >     <action application="sleep" data="1000"/>
>> >     <action application="hangup"/>
>> >   </condition>
>> > </extension>
>> >
>> >
>> > im using testserver from lib/esl/ and i just removed the conference
>> command
>> > and added the playback one.... also i moved the esl_debug lvl to 0
>> >
>> >
>> > anyhow, FS cannot run more than 2 CPS compared to 75 CPS when the
>> playback
>> > is done from the dialplan.
>> >
>> >
>> > Please, can someone give me a clue on what is going on?
>> > Maybe im doing something wrong?
>> > how to get maximum FS performance using ESL ?
>> >
>> >
>> >
>> > Regards,
>> > Tihomir.
>> >
>> >
>> > _______________________________________________
>> > Join us at ClueCon 2011, Aug 9-11, Chicago
>> > http://www.cluecon.com 877-7-4ACLUE
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> 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
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110812/b61ac371/attachment-0001.html 

From mrene_lists at avgs.ca  Fri Aug 12 09:28:55 2011
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Fri, 12 Aug 2011 01:28:55 -0400
Subject: [Freeswitch-users] Meetup Paris?
Message-ID: <E58D1B62-B7E9-4D15-A0AF-DC993AA70786@avgs.ca>

English version follows.

En cette fin de ClueCon, je pars sur Paris demain soir (donc j'arrive samedi en matin?, bien d?cal?), et j'y resterai jusqu'? mercredi.  Je propose d'organiser un petit meetup afin de joindre l'utile ? l'agr?able et de voir un peu le visage de la communaut? freeswitch parisienne. Est-ce quelqu'un peut proposer un bar sympa ou on pourrait faire un truc?

---

ClueCon is over and I'm leaving to Paris tomorrow (landing saturday morning, well jetlagged) and staying until Wednesday. I'm proposing to organize a little meetup in order to get to know the face of the parisian freeswitch community. Does anyone know a bar where we could meet up?

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca






From u2nsam at gmail.com  Fri Aug 12 10:37:42 2011
From: u2nsam at gmail.com (Sam)
Date: Fri, 12 Aug 2011 12:07:42 +0530
Subject: [Freeswitch-users] event
In-Reply-To: <CAE2ghw58FQVnvqQckzJzym6OuKFKwHefhVeAiqLB0BdoALaPeg@mail.gmail.com>
References: <CAE2ghw4GeQp-seD-c9nxa=W-wnkFbEQW6Ji_2sNqGBuaj8n+ug@mail.gmail.com>
	<CAFiqYumEKWBJd5_WR1A9cx72Rw5Digr6EVx7Q96bsBxRT6mJoA@mail.gmail.com>
	<CAE2ghw58FQVnvqQckzJzym6OuKFKwHefhVeAiqLB0BdoALaPeg@mail.gmail.com>
Message-ID: <CAE2ghw4eOkeZ---P56x7H5yThb2g1WPHSsG5raQaTJZn3jqd4w@mail.gmail.com>

The server has 3 static ip and 3 FS listening on different ips,
i have added <param name="auto-restart" value="false"/>

But still i get the below , and i want to stop it, how should i do it.

freeswitch at internal> 2011-08-12 11:48:08.265986 [INFO] mod_sofia.c:4919
EVENT_TRAP: IP change detected
2011-08-12 11:48:08.265986 [INFO] mod_sofia.c:4920 IP change detected
[192.168.53.189]->[192.168.53.188] []->[]
2011-08-12 11:48:08.405851 [NOTICE] sofia_glue.c:5192 Reload XML [Success]
2011-08-12 11:48:08.405851 [INFO] mod_enum.c:775 ENUM Reloaded
2011-08-12 11:48:08.405851 [INFO] switch_time.c:1028 Timezone reloaded 530
definitions
2011-08-12 11:48:08.925969 [DEBUG] sofia.c:1946 Write lock internal
2011-08-12 11:48:09.125988 [DEBUG] sofia.c:1946 Write lock external
2011-08-12 11:48:23.925913 [NOTICE] sofia.c:1953 Waiting for worker thread


Regards
Sam



On Mon, Aug 8, 2011 at 6:45 PM, Sam <u2nsam at gmail.com> wrote:

> Hi,
>
> The server is having 3 static ips on the ethernet interfaces and the FS is
> listening on single ip .
>
> I will try <param name="auto-restart" value="false"/> and check if it
> works.
>
> Regards
> Sam
>
>  On Mon, Aug 8, 2011 at 3:47 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> Well it looks like your machine changed its IP from .188 to .189. Did that
>> actually happen, perhaps if you're using DHCP? If it really did happen, then
>> FS needs to know so it can rebind on the new IP.
>>
>> If it didn't really happen (e.g. if your server is listening on both IPs)
>> look at the <param name="auto-restart" value="false"/> settting on the SIP
>> profile. See:
>>
>> http://wiki.freeswitch.org/wiki/Sofia#Forcing_SIP_profile_to_use_a_static_IP_address
>>
>> That parameter will make mod_sofia ignore the notification it receives
>> from the OS of a network address change.
>>
>> -Steve
>>
>>
>>
>>   On 8 August 2011 07:13, Sam <u2nsam at gmail.com> wrote:
>>
>>>   Hello,
>>>
>>> What makes the below event to occour and how to stop it reccouring.
>>>
>>> 2011-08-08 11:38:11.415945 [INFO] mod_sofia.c:4919 EVENT_TRAP: IP change
>>> detected
>>> 2011-08-08 11:38:11.415945 [INFO] mod_sofia.c:4920 IP change detected
>>> [192.168.53.188]->[192.168.53.189] []->[]
>>> 2011-08-08 11:38:11.615887 [NOTICE] sofia_glue.c:5192 Reload XML
>>> [Success]
>>> 2011-08-08 11:38:11.615887 [INFO] mod_enum.c:775 ENUM Reloaded
>>> 2011-08-08 11:38:11.615887 [INFO] switch_time.c:1028 Timezone reloaded
>>> 530 definitions
>>>
>>> Regards
>>> Sam
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110812/af100819/attachment.html 

From gerardo.barajas at gmail.com  Thu Aug 11 22:27:39 2011
From: gerardo.barajas at gmail.com (Gerardo Barajas)
Date: Thu, 11 Aug 2011 13:27:39 -0500
Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
In-Reply-To: <4E429D27.7040605@tiendalinux.com>
References: <4E4164C0.8030507@tiendalinux.com>
	<CAFiqYu=092K=O5xExqfmW2ZvBeWrxsRC_YAoNjkXSbMhe_zMEQ@mail.gmail.com>
	<4E429D27.7040605@tiendalinux.com>
Message-ID: <CAOQhXxFFSg-mT-NDGbXqoczZ-9CvwA4ivVkXSwqt+o0Prnhmng@mail.gmail.com>

?Why use Digium, when Sangoma Cards are fully compatible with FreeSwitch?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110811/82cf5067/attachment.html 

From peter.olsson at visionutveckling.se  Thu Aug 11 18:02:04 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 11 Aug 2011 16:02:04 +0200
Subject: [Freeswitch-users] odbc basic_calls
In-Reply-To: <CALNZuwkwgFmdmWxYMzkzGeW8u-TiRg2SbeJ1zFpOfxEPiyS2xg@mail.gmail.com>
References: <CALNZuwk20r8=rmu9rOOd-aChOS6XOGVm7mxKNhd1-DnC+id7Hg@mail.gmail.com>
	<CAKbxfG-8E=5Do_mLf62tokgL5eN8=LfH954tUNRawcNCtmMTLA@mail.gmail.com>,
	<CALNZuwkwgFmdmWxYMzkzGeW8u-TiRg2SbeJ1zFpOfxEPiyS2xg@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59EBABB8AD@cooper>

Could you try to create it manually? The view's definition is in switch_core_sqldb.c on line 1721.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Avi Marcus [avi at avimarcus.net]
Skickat: den 11 augusti 2011 15:48
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] odbc basic_calls

I don't see anything in the startup log about it checking the tables: http://pastebin.freeswitch.org/17013
This happens when I do "show calls". I know you changed something for show calls to even show 1 legged IVRs since my last update, but not having looked at the code, I don't see how that would be related.
I haven't heard of this basic_calls table before today.

-Avi Marcus

On Thu, Aug 11, 2011 at 4:33 PM, Anthony Minessale <anthony.minessale at gmail.com<mailto:anthony.minessale at gmail.com>> wrote:
its a view that should be created? perhaps there is an error on
startup creating the view in mysuckwell?


On Thu, Aug 11, 2011 at 6:28 AM, Avi Marcus <avi at avimarcus.net<mailto:avi at avimarcus.net>> wrote:
> I just upgraded FS since.. 7 weeks ago I think. Now: FreeSWITCH Version
> 1.0.head (git-9d98d49 2011-08-10 08-38-55 -0500)
> While testing the new build clean, when I hit F4 for show calls, I now get:
> freeswitch at internal> 2011-08-11 14:24:04.574797 [ERR]
> switch_core_sqldb.c:825 ERR: [select * from basic_calls where
> hostname='sip2' order by call_created_epoch]
> [STATE: 42S02 CODE 1146 ERROR: [unixODBC][MySQL][ODBC 3.51
> Driver][mysqld-5.1.41-3ubuntu12.10-log]Table 'freeswitch.basic_calls'
> doesn't exist
> Aren't the core odbc tables supposed to be auto-created?
> -Avi
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



--
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900<tel:%2B19193869900>

_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

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

!DSPAM:4e43de1032765001342651!


From avi at avimarcus.net  Fri Aug 12 16:05:44 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 12 Aug 2011 15:05:44 +0300
Subject: [Freeswitch-users] odbc basic_calls
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C59EBABB8AD@cooper>
References: <CALNZuwk20r8=rmu9rOOd-aChOS6XOGVm7mxKNhd1-DnC+id7Hg@mail.gmail.com>
	<CAKbxfG-8E=5Do_mLf62tokgL5eN8=LfH954tUNRawcNCtmMTLA@mail.gmail.com>
	<CALNZuwkwgFmdmWxYMzkzGeW8u-TiRg2SbeJ1zFpOfxEPiyS2xg@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59EBABB8AD@cooper>
Message-ID: <CALNZuw=uWC8kGww5Gjo_P7mxJ_QyrCDB1ipmamrG4QXVXb8TwQ@mail.gmail.com>

Error in query: Unknown column 'a.sent_callee_name' in 'field list'
So table channels.. no, I don't see that in there. I see callee_name in
there though. Did it miss some channel table alterations?

Huh, tables complete, db_data, interfaces and a few more all seem new. I've
been running odbc in mysql for a year? but I've never seen these until now.

-Avi Marcus

p.s. move this to jira now that it's back up..?


On Thu, Aug 11, 2011 at 5:02 PM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> Could you try to create it manually? The view's definition is in
> switch_core_sqldb.c on line 1721.
>
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [
> freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Avi Marcus [
> avi at avimarcus.net]
> Skickat: den 11 augusti 2011 15:48
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] odbc basic_calls
>
> I don't see anything in the startup log about it checking the tables:
> http://pastebin.freeswitch.org/17013
> This happens when I do "show calls". I know you changed something for show
> calls to even show 1 legged IVRs since my last update, but not having looked
> at the code, I don't see how that would be related.
> I haven't heard of this basic_calls table before today.
>
> -Avi Marcus
>
> On Thu, Aug 11, 2011 at 4:33 PM, Anthony Minessale <
> anthony.minessale at gmail.com<mailto:anthony.minessale at gmail.com>> wrote:
> its a view that should be created? perhaps there is an error on
> startup creating the view in mysuckwell?
>
>
> On Thu, Aug 11, 2011 at 6:28 AM, Avi Marcus <avi at avimarcus.net<mailto:
> avi at avimarcus.net>> wrote:
> > I just upgraded FS since.. 7 weeks ago I think. Now: FreeSWITCH Version
> > 1.0.head (git-9d98d49 2011-08-10 08-38-55 -0500)
> > While testing the new build clean, when I hit F4 for show calls, I now
> get:
> > freeswitch at internal> 2011-08-11 14:24:04.574797 [ERR]
> > switch_core_sqldb.c:825 ERR: [select * from basic_calls where
> > hostname='sip2' order by call_created_epoch]
> > [STATE: 42S02 CODE 1146 ERROR: [unixODBC][MySQL][ODBC 3.51
> > Driver][mysqld-5.1.41-3ubuntu12.10-log]Table 'freeswitch.basic_calls'
> > doesn't exist
> > Aren't the core odbc tables supposed to be auto-created?
> > -Avi
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org<mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com<mailto:
> MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<mailto:
> PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org<mailto:
> sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<mailto:
> googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900<tel:%2B19193869900>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> !DSPAM:4e43de1032765001342651!
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110812/63858bec/attachment-0001.html 

From vipkilla at gmail.com  Fri Aug 12 16:16:55 2011
From: vipkilla at gmail.com (vip killa)
Date: Fri, 12 Aug 2011 08:16:55 -0400
Subject: [Freeswitch-users] collect dtmf and store in channel variable
In-Reply-To: <CAKbxfG8A3XNoEhXfQwWNPwEjCKtU=sY_xWUpAEABZbzcqyxdNg@mail.gmail.com>
References: <CAC-LwPOmcTxMZYUZKny2mfryY1MVvqJ9mWrtqS+VGV+G_cis2w@mail.gmail.com>
	<23723.1313079736@ccs.covici.com>
	<CAC-LwPOSH8FxVoJB1PejS=+j==JY515u=_FDFo7M_0LeAG32nA@mail.gmail.com>
	<CALNZuw=kwNEcAaN+-oxrkOM4OxWiAc2Koryct1YbuhkMA28_MA@mail.gmail.com>
	<CAC-LwPOpgRcwhAm1Q3Y9VQ+X5qn+Wx1tQGxzesOsWm7EPJCKvQ@mail.gmail.com>
	<CALNZuw=ncGwazLjZ-gRdG1=L1oa_0p_26sE_j7mzj0u5DkT0DQ@mail.gmail.com>
	<CAC-LwPP9aWzJs1C9C-t5f+oqSZcf3rF0WSaFN4cSWZ+J29+5ug@mail.gmail.com>
	<CALNZuwnF7hoysisyo2TQ40TFc17sfxk_0AH7PgBHwLzfnrqT+w@mail.gmail.com>
	<CAC-LwPP8_SgoLwjoYKec6buMN0Jz14=F86AS0QpTJfRobXTesQ@mail.gmail.com>
	<CALNZuwkBHptWyKCWYb=X-ApcP79BxFWeySCq7SEKvFJhMdtDNQ@mail.gmail.com>
	<CAKbxfG8A3XNoEhXfQwWNPwEjCKtU=sY_xWUpAEABZbzcqyxdNg@mail.gmail.com>
Message-ID: <CAC-LwPObMxhBsTHHQnEiWhsXO8AARQucfadvbO-JiY8CHinjsA@mail.gmail.com>

I just looked at the CDR CSVs, I don't see the digits dialed in there...
there is nothing in wiki on how to use this either. How can I see the
digits_dialed
variable in the CDR?

On Thu, Aug 11, 2011 at 6:32 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> digits_dialed variable available in the cdr will contain any digits
> dialed the whole call duration.
>
>
> On Thu, Aug 11, 2011 at 3:15 PM, Avi Marcus <avi at avimarcus.net> wrote:
> > According to the wiki, apparently, you can't capture digits from
> > bind_digit_action. So you'll have to use that to trigger
> > a Dialplan_Tools_play_and_get_digits which automatically sets a variable,
> or
> > getdigits within lua/js (only?) and then save it.
> >
> > -Avi
> >
> >
> > On Thu, Aug 11, 2011 at 10:18 PM, vip killa <vipkilla at gmail.com> wrote:
> >>
> >> I can't seem to figure out how to store what is collected
> >> from bind_digit_action into a channel variable... any ideas?
> >>
> >> On Thu, Aug 11, 2011 at 3:07 PM, Avi Marcus <avi at avimarcus.net> wrote:
> >>>
> >>> So some sort of bind_digit_action to collect and use that to set a
> hangup
> >>> hook. I then execute_extension from the action will be the easiest way
> to do
> >>> it, because you have several actions to accomplish.
> >>> However, doing this while the other person is on the hook.. sounds like
> a
> >>> bad idea.
> >>> -Avi
> >>>
> >>> On Thu, Aug 11, 2011 at 9:13 PM, vip killa <vipkilla at gmail.com> wrote:
> >>>>
> >>>> thanks but i'm not interested in paying someone to do this. but yes i
> >>>> simply want to collect the digits during the call from the B leg then
> store
> >>>> them in a variable which will rename the file accordingly from the
> hangup
> >>>> hook
> >>>>
> >>>> On Thu, Aug 11, 2011 at 2:07 PM, Avi Marcus <avi at avimarcus.net>
> wrote:
> >>>>>
> >>>>> Oh, you don't want the info in the CDR, you want the info in the
> >>>>> recording. Does it have to be in the name, or is an audio tag enough?
> I'm
> >>>>> not sure if tags can work once the recording started...
> >>>>> OK, so you can set a hangup hook to rename the file and add the
> >>>>> variable with the account code to the end.
> >>>>> Whatever you use to capture the DTMF - bind_digit_action or
> >>>>> playandgetdigits or anything can set a variable.
> >>>>> -Avi Marcus
> >>>>>
> >>>>> p.s. If you want me to... actually do it for you, I'm available for
> >>>>> consulting.. email me offlist.
> >>>>>
> >>>>> On Thu, Aug 11, 2011 at 8:52 PM, vip killa <vipkilla at gmail.com>
> wrote:
> >>>>>>
> >>>>>> I apologize but the caller would not know the account number, it
> would
> >>>>>> be for internal use only. how could xml_cdrs and mod_cdr_csv be used
> to
> >>>>>> stamp a recording? how do you store the DTMF in a channel variable?
> >>>>>>
> >>>>>> On Thu, Aug 11, 2011 at 12:41 PM, Avi Marcus <avi at avimarcus.net>
> >>>>>> wrote:
> >>>>>>>
> >>>>>>> You can set any channel variable from the dialplan or from a lua
> >>>>>>> script.
> >>>>>>> I'd reccommend a standalone lua script before the agent gets on the
> >>>>>>> phone, which can verify their account number and then present it to
> the
> >>>>>>> agent even. But yes, you can do anything with the dtmf including
> setting a
> >>>>>>> variable which can be saved from the xml_cdrs or mod_cdr_csv or
> whatever
> >>>>>>> other cdr system you use.
> >>>>>>> -Avi
> >>>>>>>
> >>>>>>> On Thu, Aug 11, 2011 at 7:30 PM, vip killa <vipkilla at gmail.com>
> >>>>>>> wrote:
> >>>>>>>>
> >>>>>>>> indeed it may be easier but this is what the client is asking
> >>>>>>>> for.... i know it has to be possible... i just need some
> direction.
> >>>>>>>>
> >>>>>>>> On Thu, Aug 11, 2011 at 12:22 PM, <covici at ccs.covici.com> wrote:
> >>>>>>>>>
> >>>>>>>>> Could not the agent just type it on a screen -- it would seem to
> be
> >>>>>>>>> much
> >>>>>>>>> easier.
> >>>>>>>>>
> >>>>>>>>> vip killa <vipkilla at gmail.com> wrote:
> >>>>>>>>>
> >>>>>>>>> > Hi everyone,
> >>>>>>>>> > I'm trying to collect DTMF digits and store them in a channel
> >>>>>>>>> > variable so
> >>>>>>>>> > when the channel hangs up it uses those digits to "mark" (or
> >>>>>>>>> > rename) the
> >>>>>>>>> > recording of the call. The DTMF will be entered by the called
> >>>>>>>>> > party (i think
> >>>>>>>>> > that would be the B-leg?). I've been experimenting with
> >>>>>>>>> > "bind_meta_app" and
> >>>>>>>>> > "bind_digit_action", it seems like "bind_digit_action" may be
> the
> >>>>>>>>> > one i need
> >>>>>>>>> > to use but im not sure. I'll explain the scenario to make
> things
> >>>>>>>>> > more
> >>>>>>>>> > clear...we are trying to install this in a call center type
> >>>>>>>>> > environment
> >>>>>>>>> > where all calls are being recorded. A caller gets an agent, the
> >>>>>>>>> > caller gives
> >>>>>>>>> > the agent the account number of their case, the agent uses DTMF
> >>>>>>>>> > to mark the
> >>>>>>>>> > recording of the call with the account number of the caller's
> >>>>>>>>> > case. Does
> >>>>>>>>> > that make sense? Please let me know if this is possible,
> thanks.
> >>>>>>>>> >
> >>>>>>>>> > ----------------------------------------------------
> >>>>>>>>> > Alternatives:
> >>>>>>>>> >
> >>>>>>>>> > ----------------------------------------------------
> >>>>>>>>> > _______________________________________________
> >>>>>>>>> > Join us at ClueCon 2011, Aug 9-11, Chicago
> >>>>>>>>> > http://www.cluecon.com 877-7-4ACLUE
> >>>>>>>>> >
> >>>>>>>>> > FreeSWITCH-users mailing list
> >>>>>>>>> > FreeSWITCH-users at lists.freeswitch.org
> >>>>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>>>>> >
> >>>>>>>>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>>>>> > http://www.freeswitch.org
> >>>>>>>>>
> >>>>>>>>> --
> >>>>>>>>> Your life is like a penny.  You're going to lose it.  The
> question
> >>>>>>>>> is:
> >>>>>>>>> How do
> >>>>>>>>> you spend it?
> >>>>>>>>>
> >>>>>>>>>         John Covici
> >>>>>>>>>         covici at ccs.covici.com
> >>>>>>>>>
> >>>>>>>>> _______________________________________________
> >>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
> >>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
> >>>>>>>>>
> >>>>>>>>> FreeSWITCH-users mailing list
> >>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>>>>>
> >>>>>>>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>>>>> http://www.freeswitch.org
> >>>>>>>>
> >>>>>>>>
> >>>>>>>> _______________________________________________
> >>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
> >>>>>>>> http://www.cluecon.com 877-7-4ACLUE
> >>>>>>>>
> >>>>>>>> FreeSWITCH-users mailing list
> >>>>>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>>>>
> >>>>>>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>>>> http://www.freeswitch.org
> >>>>>>>>
> >>>>>>>
> >>>>>>>
> >>>>>>> _______________________________________________
> >>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
> >>>>>>> http://www.cluecon.com 877-7-4ACLUE
> >>>>>>>
> >>>>>>> FreeSWITCH-users mailing list
> >>>>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>>>
> >>>>>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>>> http://www.freeswitch.org
> >>>>>>>
> >>>>>>
> >>>>>>
> >>>>>> _______________________________________________
> >>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
> >>>>>> http://www.cluecon.com 877-7-4ACLUE
> >>>>>>
> >>>>>> FreeSWITCH-users mailing list
> >>>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>>
> >>>>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>> http://www.freeswitch.org
> >>>>>>
> >>>>>
> >>>>>
> >>>>> _______________________________________________
> >>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
> >>>>> http://www.cluecon.com 877-7-4ACLUE
> >>>>>
> >>>>> FreeSWITCH-users mailing list
> >>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>
> >>>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>> http://www.freeswitch.org
> >>>>>
> >>>>
> >>>>
> >>>> _______________________________________________
> >>>> Join us at ClueCon 2011, Aug 9-11, Chicago
> >>>> http://www.cluecon.com 877-7-4ACLUE
> >>>>
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>>
> >>>
> >>>
> >>> _______________________________________________
> >>> Join us at ClueCon 2011, Aug 9-11, Chicago
> >>> http://www.cluecon.com 877-7-4ACLUE
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>
> >> _______________________________________________
> >> Join us at ClueCon 2011, Aug 9-11, Chicago
> >> http://www.cluecon.com 877-7-4ACLUE
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110812/997c8125/attachment-0001.html 

From vipkilla at gmail.com  Fri Aug 12 16:23:16 2011
From: vipkilla at gmail.com (vip killa)
Date: Fri, 12 Aug 2011 08:23:16 -0400
Subject: [Freeswitch-users] collect dtmf and store in channel variable
In-Reply-To: <CAC-LwPObMxhBsTHHQnEiWhsXO8AARQucfadvbO-JiY8CHinjsA@mail.gmail.com>
References: <CAC-LwPOmcTxMZYUZKny2mfryY1MVvqJ9mWrtqS+VGV+G_cis2w@mail.gmail.com>
	<23723.1313079736@ccs.covici.com>
	<CAC-LwPOSH8FxVoJB1PejS=+j==JY515u=_FDFo7M_0LeAG32nA@mail.gmail.com>
	<CALNZuw=kwNEcAaN+-oxrkOM4OxWiAc2Koryct1YbuhkMA28_MA@mail.gmail.com>
	<CAC-LwPOpgRcwhAm1Q3Y9VQ+X5qn+Wx1tQGxzesOsWm7EPJCKvQ@mail.gmail.com>
	<CALNZuw=ncGwazLjZ-gRdG1=L1oa_0p_26sE_j7mzj0u5DkT0DQ@mail.gmail.com>
	<CAC-LwPP9aWzJs1C9C-t5f+oqSZcf3rF0WSaFN4cSWZ+J29+5ug@mail.gmail.com>
	<CALNZuwnF7hoysisyo2TQ40TFc17sfxk_0AH7PgBHwLzfnrqT+w@mail.gmail.com>
	<CAC-LwPP8_SgoLwjoYKec6buMN0Jz14=F86AS0QpTJfRobXTesQ@mail.gmail.com>
	<CALNZuwkBHptWyKCWYb=X-ApcP79BxFWeySCq7SEKvFJhMdtDNQ@mail.gmail.com>
	<CAKbxfG8A3XNoEhXfQwWNPwEjCKtU=sY_xWUpAEABZbzcqyxdNg@mail.gmail.com>
	<CAC-LwPObMxhBsTHHQnEiWhsXO8AARQucfadvbO-JiY8CHinjsA@mail.gmail.com>
Message-ID: <CAC-LwPN_bJEYuK3Za28C5ip3JiDRErtKVb9q0C=yNy1Qzwx+BA@mail.gmail.com>

I apologize, I got that to work by adding ${digits_dialed} to
cdr_csv.conf.xml
Still, it only records the digits from the A leg, anyway to make it record
digits from the B leg?

On Fri, Aug 12, 2011 at 8:16 AM, vip killa <vipkilla at gmail.com> wrote:

> I just looked at the CDR CSVs, I don't see the digits dialed in there...
> there is nothing in wiki on how to use this either. How can I see the digits_dialed
> variable in the CDR?
>
>
> On Thu, Aug 11, 2011 at 6:32 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> digits_dialed variable available in the cdr will contain any digits
>> dialed the whole call duration.
>>
>>
>> On Thu, Aug 11, 2011 at 3:15 PM, Avi Marcus <avi at avimarcus.net> wrote:
>> > According to the wiki, apparently, you can't capture digits from
>> > bind_digit_action. So you'll have to use that to trigger
>> > a Dialplan_Tools_play_and_get_digits which automatically sets a
>> variable, or
>> > getdigits within lua/js (only?) and then save it.
>> >
>> > -Avi
>> >
>> >
>> > On Thu, Aug 11, 2011 at 10:18 PM, vip killa <vipkilla at gmail.com> wrote:
>> >>
>> >> I can't seem to figure out how to store what is collected
>> >> from bind_digit_action into a channel variable... any ideas?
>> >>
>> >> On Thu, Aug 11, 2011 at 3:07 PM, Avi Marcus <avi at avimarcus.net> wrote:
>> >>>
>> >>> So some sort of bind_digit_action to collect and use that to set a
>> hangup
>> >>> hook. I then execute_extension from the action will be the easiest way
>> to do
>> >>> it, because you have several actions to accomplish.
>> >>> However, doing this while the other person is on the hook.. sounds
>> like a
>> >>> bad idea.
>> >>> -Avi
>> >>>
>> >>> On Thu, Aug 11, 2011 at 9:13 PM, vip killa <vipkilla at gmail.com>
>> wrote:
>> >>>>
>> >>>> thanks but i'm not interested in paying someone to do this. but yes i
>> >>>> simply want to collect the digits during the call from the B leg then
>> store
>> >>>> them in a variable which will rename the file accordingly from the
>> hangup
>> >>>> hook
>> >>>>
>> >>>> On Thu, Aug 11, 2011 at 2:07 PM, Avi Marcus <avi at avimarcus.net>
>> wrote:
>> >>>>>
>> >>>>> Oh, you don't want the info in the CDR, you want the info in the
>> >>>>> recording. Does it have to be in the name, or is an audio tag
>> enough? I'm
>> >>>>> not sure if tags can work once the recording started...
>> >>>>> OK, so you can set a hangup hook to rename the file and add the
>> >>>>> variable with the account code to the end.
>> >>>>> Whatever you use to capture the DTMF - bind_digit_action or
>> >>>>> playandgetdigits or anything can set a variable.
>> >>>>> -Avi Marcus
>> >>>>>
>> >>>>> p.s. If you want me to... actually do it for you, I'm available for
>> >>>>> consulting.. email me offlist.
>> >>>>>
>> >>>>> On Thu, Aug 11, 2011 at 8:52 PM, vip killa <vipkilla at gmail.com>
>> wrote:
>> >>>>>>
>> >>>>>> I apologize but the caller would not know the account number, it
>> would
>> >>>>>> be for internal use only. how could xml_cdrs and mod_cdr_csv be
>> used to
>> >>>>>> stamp a recording? how do you store the DTMF in a channel variable?
>> >>>>>>
>> >>>>>> On Thu, Aug 11, 2011 at 12:41 PM, Avi Marcus <avi at avimarcus.net>
>> >>>>>> wrote:
>> >>>>>>>
>> >>>>>>> You can set any channel variable from the dialplan or from a lua
>> >>>>>>> script.
>> >>>>>>> I'd reccommend a standalone lua script before the agent gets on
>> the
>> >>>>>>> phone, which can verify their account number and then present it
>> to the
>> >>>>>>> agent even. But yes, you can do anything with the dtmf including
>> setting a
>> >>>>>>> variable which can be saved from the xml_cdrs or mod_cdr_csv or
>> whatever
>> >>>>>>> other cdr system you use.
>> >>>>>>> -Avi
>> >>>>>>>
>> >>>>>>> On Thu, Aug 11, 2011 at 7:30 PM, vip killa <vipkilla at gmail.com>
>> >>>>>>> wrote:
>> >>>>>>>>
>> >>>>>>>> indeed it may be easier but this is what the client is asking
>> >>>>>>>> for.... i know it has to be possible... i just need some
>> direction.
>> >>>>>>>>
>> >>>>>>>> On Thu, Aug 11, 2011 at 12:22 PM, <covici at ccs.covici.com> wrote:
>> >>>>>>>>>
>> >>>>>>>>> Could not the agent just type it on a screen -- it would seem to
>> be
>> >>>>>>>>> much
>> >>>>>>>>> easier.
>> >>>>>>>>>
>> >>>>>>>>> vip killa <vipkilla at gmail.com> wrote:
>> >>>>>>>>>
>> >>>>>>>>> > Hi everyone,
>> >>>>>>>>> > I'm trying to collect DTMF digits and store them in a channel
>> >>>>>>>>> > variable so
>> >>>>>>>>> > when the channel hangs up it uses those digits to "mark" (or
>> >>>>>>>>> > rename) the
>> >>>>>>>>> > recording of the call. The DTMF will be entered by the called
>> >>>>>>>>> > party (i think
>> >>>>>>>>> > that would be the B-leg?). I've been experimenting with
>> >>>>>>>>> > "bind_meta_app" and
>> >>>>>>>>> > "bind_digit_action", it seems like "bind_digit_action" may be
>> the
>> >>>>>>>>> > one i need
>> >>>>>>>>> > to use but im not sure. I'll explain the scenario to make
>> things
>> >>>>>>>>> > more
>> >>>>>>>>> > clear...we are trying to install this in a call center type
>> >>>>>>>>> > environment
>> >>>>>>>>> > where all calls are being recorded. A caller gets an agent,
>> the
>> >>>>>>>>> > caller gives
>> >>>>>>>>> > the agent the account number of their case, the agent uses
>> DTMF
>> >>>>>>>>> > to mark the
>> >>>>>>>>> > recording of the call with the account number of the caller's
>> >>>>>>>>> > case. Does
>> >>>>>>>>> > that make sense? Please let me know if this is possible,
>> thanks.
>> >>>>>>>>> >
>> >>>>>>>>> > ----------------------------------------------------
>> >>>>>>>>> > Alternatives:
>> >>>>>>>>> >
>> >>>>>>>>> > ----------------------------------------------------
>> >>>>>>>>> > _______________________________________________
>> >>>>>>>>> > Join us at ClueCon 2011, Aug 9-11, Chicago
>> >>>>>>>>> > http://www.cluecon.com 877-7-4ACLUE
>> >>>>>>>>> >
>> >>>>>>>>> > FreeSWITCH-users mailing list
>> >>>>>>>>> > FreeSWITCH-users at lists.freeswitch.org
>> >>>>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>>>>>> >
>> >>>>>>>>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>>>>>>> > http://www.freeswitch.org
>> >>>>>>>>>
>> >>>>>>>>> --
>> >>>>>>>>> Your life is like a penny.  You're going to lose it.  The
>> question
>> >>>>>>>>> is:
>> >>>>>>>>> How do
>> >>>>>>>>> you spend it?
>> >>>>>>>>>
>> >>>>>>>>>         John Covici
>> >>>>>>>>>         covici at ccs.covici.com
>> >>>>>>>>>
>> >>>>>>>>> _______________________________________________
>> >>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> >>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>> >>>>>>>>>
>> >>>>>>>>> FreeSWITCH-users mailing list
>> >>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>>>>>>
>> >>>>>>>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>>>>>>> http://www.freeswitch.org
>> >>>>>>>>
>> >>>>>>>>
>> >>>>>>>> _______________________________________________
>> >>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> >>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>> >>>>>>>>
>> >>>>>>>> FreeSWITCH-users mailing list
>> >>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>>>>>
>> >>>>>>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>>>>>> http://www.freeswitch.org
>> >>>>>>>>
>> >>>>>>>
>> >>>>>>>
>> >>>>>>> _______________________________________________
>> >>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> >>>>>>> http://www.cluecon.com 877-7-4ACLUE
>> >>>>>>>
>> >>>>>>> FreeSWITCH-users mailing list
>> >>>>>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>>>>
>> >>>>>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>>>>> http://www.freeswitch.org
>> >>>>>>>
>> >>>>>>
>> >>>>>>
>> >>>>>> _______________________________________________
>> >>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> >>>>>> http://www.cluecon.com 877-7-4ACLUE
>> >>>>>>
>> >>>>>> FreeSWITCH-users mailing list
>> >>>>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>>>
>> >>>>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>>>> http://www.freeswitch.org
>> >>>>>>
>> >>>>>
>> >>>>>
>> >>>>> _______________________________________________
>> >>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> >>>>> http://www.cluecon.com 877-7-4ACLUE
>> >>>>>
>> >>>>> FreeSWITCH-users mailing list
>> >>>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>>>
>> >>>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>>> http://www.freeswitch.org
>> >>>>>
>> >>>>
>> >>>>
>> >>>> _______________________________________________
>> >>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> >>>> http://www.cluecon.com 877-7-4ACLUE
>> >>>>
>> >>>> FreeSWITCH-users mailing list
>> >>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> http://www.freeswitch.org
>> >>>>
>> >>>
>> >>>
>> >>> _______________________________________________
>> >>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> >>> http://www.cluecon.com 877-7-4ACLUE
>> >>>
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>>
>> >>
>> >>
>> >> _______________________________________________
>> >> Join us at ClueCon 2011, Aug 9-11, Chicago
>> >> http://www.cluecon.com 877-7-4ACLUE
>> >>
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> > _______________________________________________
>> > Join us at ClueCon 2011, Aug 9-11, Chicago
>> > http://www.cluecon.com 877-7-4ACLUE
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> 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
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110812/57d9f370/attachment-0001.html 

From michel.daggelinckx at gmail.com  Fri Aug 12 17:48:54 2011
From: michel.daggelinckx at gmail.com (Michel Daggelinckx)
Date: Fri, 12 Aug 2011 15:48:54 +0200
Subject: [Freeswitch-users] Meetup Paris?
In-Reply-To: <E58D1B62-B7E9-4D15-A0AF-DC993AA70786@avgs.ca>
References: <E58D1B62-B7E9-4D15-A0AF-DC993AA70786@avgs.ca>
Message-ID: <CAFYZzbvqVs_gFNKuoORNxTBd31Ya6AxSz3bz-MV3D-P8DCQJGg@mail.gmail.com>

A European version of cluecon would be nice.

Michel

On Fri, Aug 12, 2011 at 7:28 AM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> English version follows.
>
> En cette fin de ClueCon, je pars sur Paris demain soir (donc j'arrive
> samedi en matin?, bien d?cal?), et j'y resterai jusqu'? mercredi.  Je
> propose d'organiser un petit meetup afin de joindre l'utile ? l'agr?able et
> de voir un peu le visage de la communaut? freeswitch parisienne. Est-ce
> quelqu'un peut proposer un bar sympa ou on pourrait faire un truc?
>
> ---
>
> ClueCon is over and I'm leaving to Paris tomorrow (landing saturday
> morning, well jetlagged) and staying until Wednesday. I'm proposing to
> organize a little meetup in order to get to know the face of the parisian
> freeswitch community. Does anyone know a bar where we could meet up?
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110812/4d56e8ef/attachment.html 

From michal.bielicki at seventhsignal.de  Fri Aug 12 18:33:34 2011
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Fri, 12 Aug 2011 16:33:34 +0200
Subject: [Freeswitch-users] Meetup Paris?
In-Reply-To: <CAFYZzbvqVs_gFNKuoORNxTBd31Ya6AxSz3bz-MV3D-P8DCQJGg@mail.gmail.com>
References: <E58D1B62-B7E9-4D15-A0AF-DC993AA70786@avgs.ca>
	<CAFYZzbvqVs_gFNKuoORNxTBd31Ya6AxSz3bz-MV3D-P8DCQJGg@mail.gmail.com>
Message-ID: <B61CBF19-1706-4370-A2D2-546561708847@seventhsignal.de>

There used to be something close organised by the German Unix User Group every Year. But it died off. It would be interesting to see how much interest would be there to reawake it, since I am evetl. thinking of doing something like that in germany next year.
Am 12.08.2011 um 15:48 schrieb Michel Daggelinckx:

> A European version of cluecon would be nice.
> 
> Michel
> 
> On Fri, Aug 12, 2011 at 7:28 AM, Mathieu Rene <mrene_lists at avgs.ca> wrote:
> English version follows.
> 
> En cette fin de ClueCon, je pars sur Paris demain soir (donc j'arrive samedi en matin?, bien d?cal?), et j'y resterai jusqu'? mercredi.  Je propose d'organiser un petit meetup afin de joindre l'utile ? l'agr?able et de voir un peu le visage de la communaut? freeswitch parisienne. Est-ce quelqu'un peut proposer un bar sympa ou on pourrait faire un truc?
> 
> ---
> 
> ClueCon is over and I'm leaving to Paris tomorrow (landing saturday morning, well jetlagged) and staying until Wednesday. I'm proposing to organize a little meetup in order to get to know the face of the parisian freeswitch community. Does anyone know a bar where we could meet up?
> 
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
> 
> 
> 
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Michal Bielicki
Gesch?ftsf?hrer / CEO

Seventh Signal Ltd. & Co. KG
Weigandufer 45, B?ro 115, D-12059 Berlin
Voice: +49 30 60988730

Amtsgericht Charlottenburg HRA 44413 B
Ust.-ID: DE266981999
Gesch?ftsf?hrer: Michal Bielicki
Pers?nlich Haftende Gesellschafterin:
Seventh Signal Ltd, 69 Great Hampton St. Birmingham, 
B18 6EW, GB, Company Nr.: 06889439
WWW.: http://www.seventhsignal.de



----

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110812/96380c0b/attachment.html 

From gmaruzz at gmail.com  Fri Aug 12 20:29:18 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 12 Aug 2011 18:29:18 +0200
Subject: [Freeswitch-users] Meetup Paris?
In-Reply-To: <B61CBF19-1706-4370-A2D2-546561708847@seventhsignal.de>
References: <E58D1B62-B7E9-4D15-A0AF-DC993AA70786@avgs.ca>
	<CAFYZzbvqVs_gFNKuoORNxTBd31Ya6AxSz3bz-MV3D-P8DCQJGg@mail.gmail.com>
	<B61CBF19-1706-4370-A2D2-546561708847@seventhsignal.de>
Message-ID: <CALXCt0pcAUVq_qRKqkYEq4zZeiQzo9so+tdQGuZnTe1+ZWXHXg@mail.gmail.com>

Yep, is due time to begin push FS interest in Europe.
I'm just arrived in Milan, so I'll miss the Paris bistrot, but I'll
see Math next week in Milan.
Let's start to conspire for a FS Europe!

-giovanni

On 8/12/11, Michal Bielicki <michal.bielicki at seventhsignal.de> wrote:
> There used to be something close organised by the German Unix User Group
> every Year. But it died off. It would be interesting to see how much
> interest would be there to reawake it, since I am evetl. thinking of doing
> something like that in germany next year.
> Am 12.08.2011 um 15:48 schrieb Michel Daggelinckx:
>
>> A European version of cluecon would be nice.
>>
>> Michel
>>
>> On Fri, Aug 12, 2011 at 7:28 AM, Mathieu Rene <mrene_lists at avgs.ca> wrote:
>> English version follows.
>>
>> En cette fin de ClueCon, je pars sur Paris demain soir (donc j'arrive
>> samedi en matin?, bien d?cal?), et j'y resterai jusqu'? mercredi.  Je
>> propose d'organiser un petit meetup afin de joindre l'utile ? l'agr?able
>> et de voir un peu le visage de la communaut? freeswitch parisienne. Est-ce
>> quelqu'un peut proposer un bar sympa ou on pourrait faire un truc?
>>
>> ---
>>
>> ClueCon is over and I'm leaving to Paris tomorrow (landing saturday
>> morning, well jetlagged) and staying until Wednesday. I'm proposing to
>> organize a little meetup in order to get to know the face of the parisian
>> freeswitch community. Does anyone know a bar where we could meet up?
>>
>> Mathieu Rene
>> Avant-Garde Solutions Inc
>> Office: + 1 (514) 664-1044 x100
>> Cell: +1 (514) 664-1044 x200
>> mrene at avgs.ca
>>
>>
>>
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> Michal Bielicki
> Gesch?ftsf?hrer / CEO
>
> Seventh Signal Ltd. & Co. KG
> Weigandufer 45, B?ro 115, D-12059 Berlin
> Voice: +49 30 60988730
>
> Amtsgericht Charlottenburg HRA 44413 B
> Ust.-ID: DE266981999
> Gesch?ftsf?hrer: Michal Bielicki
> Pers?nlich Haftende Gesellschafterin:
> Seventh Signal Ltd, 69 Great Hampton St. Birmingham,
> B18 6EW, GB, Company Nr.: 06889439
> WWW.: http://www.seventhsignal.de
>
>
>
> ----
>
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From sdame at 207me.com  Thu Aug 11 21:37:39 2011
From: sdame at 207me.com (Stephen Dame)
Date: Thu, 11 Aug 2011 13:37:39 -0400
Subject: [Freeswitch-users] Playback .wma/.wmv
Message-ID: <ykia4qwbow2ys8eddfcwhdjy.1313084259315@email.android.com>



Sent from my U.S. Cellular? Android phone

Jeff Lenk <jeff at jefflenk.com> wrote:

>Not at this time. AFAIK
>
>--
>View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Playback-wma-wmv-tp6674880p6677105.html
>Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>_______________________________________________
>Join us at ClueCon 2011, Aug 9-11, Chicago
>http://www.cluecon.com 877-7-4ACLUE
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org

From alec.taylor6 at gmail.com  Fri Aug 12 19:55:13 2011
From: alec.taylor6 at gmail.com (Alec Taylor)
Date: Sat, 13 Aug 2011 01:55:13 +1000
Subject: [Freeswitch-users] Conference calls through web-interface with
	moderation using FreeSWITCH?
Message-ID: <CAO+9iGdBYXQ2f8BFB+gSriP7bb_rKaFVVwaJzK2YmkfPWEAbdA@mail.gmail.com>

Good Morning,

I have been researching this for a while, basically I'd like to have a
website with the following functionality:
? One-click call-in to show (after setting username, best-case
scenario: sign-in through Drupal, use that name for conference-call)
? Web-interface only (Flash/Flex, Javascript/JQuery or Java), without
any additional software/addons/plugins to install
? Moderation: host of conference call can quieten/mute or even kick
people from the conference call if they're being rowdy

So far I have setup an IceCAST server, broadcasting through edcast in
an mp3 stream. Viewers of my website can now listen-in on the /radio/
sub-page.

How do I setup the aforementioned [3] features using FreeSWITCH? ? Do
I need [Free, Open-Source] products other than FreeSWITCH to get this
done, if so, which?

Thanks for all suggestions,

Alec Taylor


From alec.taylor6 at gmail.com  Fri Aug 12 20:47:29 2011
From: alec.taylor6 at gmail.com (Alec Taylor)
Date: Sat, 13 Aug 2011 02:47:29 +1000
Subject: [Freeswitch-users] Meetup Paris?
In-Reply-To: <CALXCt0pcAUVq_qRKqkYEq4zZeiQzo9so+tdQGuZnTe1+ZWXHXg@mail.gmail.com>
References: <E58D1B62-B7E9-4D15-A0AF-DC993AA70786@avgs.ca>
	<CAFYZzbvqVs_gFNKuoORNxTBd31Ya6AxSz3bz-MV3D-P8DCQJGg@mail.gmail.com>
	<B61CBF19-1706-4370-A2D2-546561708847@seventhsignal.de>
	<CALXCt0pcAUVq_qRKqkYEq4zZeiQzo9so+tdQGuZnTe1+ZWXHXg@mail.gmail.com>
Message-ID: <CAO+9iGe4M9eep9pANrFK-sLHzB=342BYMQbBzzzYop2MKUV2yQ@mail.gmail.com>

Want me to setup a conference site for you guys?

Cost = references

:]

On Sat, Aug 13, 2011 at 2:29 AM, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
> Yep, is due time to begin push FS interest in Europe.
> I'm just arrived in Milan, so I'll miss the Paris bistrot, but I'll
> see Math next week in Milan.
> Let's start to conspire for a FS Europe!
>
> -giovanni
>
> On 8/12/11, Michal Bielicki <michal.bielicki at seventhsignal.de> wrote:
>> There used to be something close organised by the German Unix User Group
>> every Year. But it died off. It would be interesting to see how much
>> interest would be there to reawake it, since I am evetl. thinking of doing
>> something like that in germany next year.
>> Am 12.08.2011 um 15:48 schrieb Michel Daggelinckx:
>>
>>> A European version of cluecon would be nice.
>>>
>>> Michel
>>>
>>> On Fri, Aug 12, 2011 at 7:28 AM, Mathieu Rene <mrene_lists at avgs.ca> wrote:
>>> English version follows.
>>>
>>> En cette fin de ClueCon, je pars sur Paris demain soir (donc j'arrive
>>> samedi en matin?, bien d?cal?), et j'y resterai jusqu'? mercredi. ?Je
>>> propose d'organiser un petit meetup afin de joindre l'utile ? l'agr?able
>>> et de voir un peu le visage de la communaut? freeswitch parisienne. Est-ce
>>> quelqu'un peut proposer un bar sympa ou on pourrait faire un truc?
>>>
>>> ---
>>>
>>> ClueCon is over and I'm leaving to Paris tomorrow (landing saturday
>>> morning, well jetlagged) and staying until Wednesday. I'm proposing to
>>> organize a little meetup in order to get to know the face of the parisian
>>> freeswitch community. Does anyone know a bar where we could meet up?
>>>
>>> Mathieu Rene
>>> Avant-Garde Solutions Inc
>>> Office: + 1 (514) 664-1044 x100
>>> Cell: +1 (514) 664-1044 x200
>>> mrene at avgs.ca
>>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> Michal Bielicki
>> Gesch?ftsf?hrer / CEO
>>
>> Seventh Signal Ltd. & Co. KG
>> Weigandufer 45, B?ro 115, D-12059 Berlin
>> Voice: +49 30 60988730
>>
>> Amtsgericht Charlottenburg HRA 44413 B
>> Ust.-ID: DE266981999
>> Gesch?ftsf?hrer: Michal Bielicki
>> Pers?nlich Haftende Gesellschafterin:
>> Seventh Signal Ltd, 69 Great Hampton St. Birmingham,
>> B18 6EW, GB, Company Nr.: 06889439
>> WWW.: http://www.seventhsignal.de
>>
>>
>>
>> ----
>>
>>
>
> --
> Sent from my mobile device
>
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From cjbujold at accra.ca  Fri Aug 12 20:50:06 2011
From: cjbujold at accra.ca (Charles Bujold)
Date: Fri, 12 Aug 2011 13:50:06 -0300
Subject: [Freeswitch-users] Newbie - error when calling with new Yealink
	T20P phone
Message-ID: <00e601cc590f$e3db66d0$ab923470$@accra.ca>

Just installed a new phone Yealink T20P  at another location and it
registers with Freeswitch.  However when I try to check voice messages or
make calls I get the following error.  If somebody could help me identify
and fix the issue it would be most appreciated.  (xxx hides the IP)

 

Thanks

 

CJB

 

2011-08-10 10:56:57.711199 [DEBUG] switch_rtp.c:3105 Correct ip/port
confirmed.

2011-08-10 10:57:30.191234 [WARNING] sofia_reg.c:1339 SIP auth challenge
(REGISTER) on sofia profile 'internal' for [401 at sip.accra.com] from ip
142.166.219.xxx

2011-08-10 11:00:31.171255 [WARNING] sofia_reg.c:1339 SIP auth challenge
(REGISTER) on sofia profile 'internal' for [401 at sip.accra.com] from ip
142.166.219. xxx

2011-08-10 11:03:28.691243 [WARNING] sofia_reg.c:1339 SIP auth challenge
(REGISTER) on sofia profile 'internal' for [400 at sip.accra.com] from ip
142.166.219. xxx

2011-08-10 11:03:40.271249 [DEBUG] sofia.c:7069 IP 142.166.219. xxx Rejected
by acl "domains". Falling back to Digest auth.

2011-08-10 11:03:40.271249 [WARNING] sofia_reg.c:1339 SIP auth challenge
(INVITE) on sofia profile 'internal' for [*98 at sip.accra.com] from ip
142.166.219. xxx

2011-08-10 11:03:40.291256 [WARNING] sofia_reg.c:1339 SIP auth challenge
(REGISTER) on sofia profile 'internal' for [401 at sip.accra.com] from ip
142.166.219. xxx

2011-08-10 11:03:40.431248 [DEBUG] sofia.c:7069 IP 142.166.219. xxx Rejected
by acl "domains". Falling back to Digest auth.

2011-08-10 11:03:40.431248 [NOTICE] switch_channel.c:904 New Channel
sofia/internal/401 at sip.accra.com [5aec6826-e15b-4ce8-bffe-6a218fc0030f]

2011-08-10 11:03:40.431248 [DEBUG] switch_core_state_machine.c:330
(sofia/internal/401 at sip.accra.com) Running State Change CS_NEW

2011-08-10 11:03:40.431248 [DEBUG] switch_core_state_machine.c:348
(sofia/internal/401 at sip.accra.com) State NEW

2011-08-10 11:03:40.431248 [DEBUG] sofia.c:5119 Channel
sofia/internal/401 at sip.accra.com entering state [received][100]

2011-08-10 11:03:40.431248 [DEBUG] sofia.c:5130 Remote SDP:

v=0

o=- 20001 20001 IN IP4 142.166.219. xxx

s=SDP data

c=IN IP4 142.166.219. xxx

t=0 0

m=audio 11782 RTP/AVP 18 101

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

 

2011-08-10 11:03:40.431248 [DEBUG] sofia_glue.c:4731 Audio Codec Compare
[G729:18:8000:20:8000]/[G722:9:8000:20:64000]

2011-08-10 11:03:40.431248 [DEBUG] sofia_glue.c:4731 Audio Codec Compare
[G729:18:8000:20:8000]/[PCMU:0:8000:20:64000]

2011-08-10 11:03:40.431248 [DEBUG] sofia_glue.c:4731 Audio Codec Compare
[G729:18:8000:20:8000]/[PCMA:8:8000:20:64000]

2011-08-10 11:03:40.431248 [DEBUG] sofia_glue.c:4731 Audio Codec Compare
[G729:18:8000:20:8000]/[GSM:3:8000:20:13200]

2011-08-10 11:03:40.431248 [DEBUG] sofia_glue.c:4731 Audio Codec Compare
[telephone-event:101:8000:20:0]/[G722:9:8000:20:64000]

2011-08-10 11:03:40.431248 [DEBUG] sofia_glue.c:4731 Audio Codec Compare
[telephone-event:101:8000:20:0]/[PCMU:0:8000:20:64000]

2011-08-10 11:03:40.431248 [DEBUG] sofia_glue.c:4731 Audio Codec Compare
[telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000]

2011-08-10 11:03:40.431248 [DEBUG] sofia_glue.c:4731 Audio Codec Compare
[telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200]

2011-08-10 11:03:40.431248 [DEBUG] sofia_glue.c:4845 Set 2833 dtmf send/recv
payload to 101

2011-08-10 11:03:40.431248 [DEBUG] switch_channel.c:2767
(sofia/internal/401 at sip.accra.com) Callstate Change DOWN -> HANGUP

2011-08-10 11:03:40.431248 [NOTICE] sofia.c:5374 Hangup
sofia/internal/401 at sip.accra.com [CS_NEW] [INCOMPATIBLE_DESTINATION]

2011-08-10 11:03:40.431248 [DEBUG] switch_channel.c:2783 Send signal
sofia/internal/401 at sip.accra.com [KILL]

2011-08-10 11:03:40.431248 [DEBUG] switch_core_session.c:1156 Send signal
sofia/internal/401 at sip.accra.com [BREAK]

2011-08-10 11:03:40.431248 [DEBUG] switch_core_state_machine.c:330
(sofia/internal/401 at sip.accra.com) Running State Change CS_HANGUP

2011-08-10 11:03:40.431248 [DEBUG] switch_core_state_machine.c:580
(sofia/internal/401 at sip.accra.com) State HANGUP

2011-08-10 11:03:40.431248 [DEBUG] mod_sofia.c:458 Channel
sofia/internal/401 at sip.accra.com hanging up, cause: INCOMPATIBLE_DESTINATION

2011-08-10 11:03:40.431248 [DEBUG] mod_sofia.c:522 Responding to INVITE
with: 488

2011-08-10 11:03:40.431248 [DEBUG] switch_core_state_machine.c:46
sofia/internal/401 at sip.accra.com Standard HANGUP, cause:
INCOMPATIBLE_DESTINATION

2011-08-10 11:03:40.431248 [DEBUG] switch_core_state_machine.c:580
(sofia/internal/401 at sip.accra.com) State HANGUP going to sleep

2011-08-10 11:03:40.431248 [DEBUG] switch_core_state_machine.c:361
(sofia/internal/401 at sip.accra.com) State Change CS_HANGUP -> CS_REPORTING

2011-08-10 11:03:40.431248 [DEBUG] switch_core_session.c:1156 Send signal
sofia/internal/401 at sip.accra.com [BREAK]

2011-08-10 11:03:40.431248 [DEBUG] switch_core_state_machine.c:330
(sofia/internal/401 at sip.accra.com) Running State Change CS_REPORTING

2011-08-10 11:03:40.431248 [DEBUG] switch_core_state_machine.c:640
(sofia/internal/401 at sip.accra.com) State REPORTING

2011-08-10 11:03:40.611244 [DEBUG] switch_core_state_machine.c:53
sofia/internal/401 at sip.accra.com Standard REPORTING, cause:
INCOMPATIBLE_DESTINATION

2011-08-10 11:03:40.611244 [DEBUG] switch_core_state_machine.c:640
(sofia/internal/401 at sip.accra.com) State REPORTING going to sleep

2011-08-10 11:03:40.611244 [DEBUG] switch_core_state_machine.c:355
(sofia/internal/401 at sip.accra.com) State Change CS_REPORTING -> CS_DESTROY

2011-08-10 11:03:40.611244 [DEBUG] switch_core_session.c:1156 Send signal
sofia/internal/401 at sip.accra.com [BREAK]

2011-08-10 11:03:40.611244 [DEBUG] switch_core_session.c:1328 Session 64
(sofia/internal/401 at sip.accra.com) Locked, Waiting on external entities

2011-08-10 11:03:40.611244 [NOTICE] switch_core_session.c:1346 Session 64
(sofia/internal/401 at sip.accra.com) Ended

2011-08-10 11:03:40.611244 [NOTICE] switch_core_session.c:1348 Close Channel
sofia/internal/401 at sip.accra.com [CS_DESTROY]

2011-08-10 11:03:40.611244 [DEBUG] switch_core_state_machine.c:469
(sofia/internal/401 at sip.accra.com) Callstate Change HANGUP -> DOWN

2011-08-10 11:03:40.611244 [DEBUG] switch_core_state_machine.c:472
(sofia/internal/401 at sip.accra.com) Running State Change CS_DESTROY

2011-08-10 11:03:40.611244 [DEBUG] switch_core_state_machine.c:482
(sofia/internal/401 at sip.accra.com) State DESTROY

2011-08-10 11:03:40.611244 [DEBUG] mod_sofia.c:363
sofia/internal/401 at sip.accra.com SOFIA DESTROY

2011-08-10 11:03:40.611244 [DEBUG] switch_core_state_machine.c:60
sofia/internal/401 at sip.accra.com Standard DESTROY

2011-08-10 11:03:40.611244 [DEBUG] switch_core_state_machine.c:482
(sofia/internal/401 at sip.accra.com) State DESTROY going to sleep

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110812/a860096e/attachment.html 

From vipkilla at gmail.com  Fri Aug 12 21:12:22 2011
From: vipkilla at gmail.com (vip killa)
Date: Fri, 12 Aug 2011 13:12:22 -0400
Subject: [Freeswitch-users] collect dtmf and store in channel variable
In-Reply-To: <CAC-LwPN_bJEYuK3Za28C5ip3JiDRErtKVb9q0C=yNy1Qzwx+BA@mail.gmail.com>
References: <CAC-LwPOmcTxMZYUZKny2mfryY1MVvqJ9mWrtqS+VGV+G_cis2w@mail.gmail.com>
	<23723.1313079736@ccs.covici.com>
	<CAC-LwPOSH8FxVoJB1PejS=+j==JY515u=_FDFo7M_0LeAG32nA@mail.gmail.com>
	<CALNZuw=kwNEcAaN+-oxrkOM4OxWiAc2Koryct1YbuhkMA28_MA@mail.gmail.com>
	<CAC-LwPOpgRcwhAm1Q3Y9VQ+X5qn+Wx1tQGxzesOsWm7EPJCKvQ@mail.gmail.com>
	<CALNZuw=ncGwazLjZ-gRdG1=L1oa_0p_26sE_j7mzj0u5DkT0DQ@mail.gmail.com>
	<CAC-LwPP9aWzJs1C9C-t5f+oqSZcf3rF0WSaFN4cSWZ+J29+5ug@mail.gmail.com>
	<CALNZuwnF7hoysisyo2TQ40TFc17sfxk_0AH7PgBHwLzfnrqT+w@mail.gmail.com>
	<CAC-LwPP8_SgoLwjoYKec6buMN0Jz14=F86AS0QpTJfRobXTesQ@mail.gmail.com>
	<CALNZuwkBHptWyKCWYb=X-ApcP79BxFWeySCq7SEKvFJhMdtDNQ@mail.gmail.com>
	<CAKbxfG8A3XNoEhXfQwWNPwEjCKtU=sY_xWUpAEABZbzcqyxdNg@mail.gmail.com>
	<CAC-LwPObMxhBsTHHQnEiWhsXO8AARQucfadvbO-JiY8CHinjsA@mail.gmail.com>
	<CAC-LwPN_bJEYuK3Za28C5ip3JiDRErtKVb9q0C=yNy1Qzwx+BA@mail.gmail.com>
Message-ID: <CAC-LwPN0DJRjtgYEnyOavYBB4-NqdzU4Ou-ecWwXEteFdiVK6A@mail.gmail.com>

I've been informed that i need to use xml_cdr to log the B-leg. But i dont
see a way to log the digits_dialed variable using xml_cdr. with cdr_csv you
use templates. is there something similar with xml_cdr? Thanks for everyones
help so far.

On Fri, Aug 12, 2011 at 8:23 AM, vip killa <vipkilla at gmail.com> wrote:

> I apologize, I got that to work by adding ${digits_dialed} to
> cdr_csv.conf.xml
> Still, it only records the digits from the A leg, anyway to make it record
> digits from the B leg?
>
> On Fri, Aug 12, 2011 at 8:16 AM, vip killa <vipkilla at gmail.com> wrote:
>
>> I just looked at the CDR CSVs, I don't see the digits dialed in there...
>> there is nothing in wiki on how to use this either. How can I see the digits_dialed
>> variable in the CDR?
>>
>>
>> On Thu, Aug 11, 2011 at 6:32 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> digits_dialed variable available in the cdr will contain any digits
>>> dialed the whole call duration.
>>>
>>>
>>> On Thu, Aug 11, 2011 at 3:15 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>> > According to the wiki, apparently, you can't capture digits from
>>> > bind_digit_action. So you'll have to use that to trigger
>>> > a Dialplan_Tools_play_and_get_digits which automatically sets a
>>> variable, or
>>> > getdigits within lua/js (only?) and then save it.
>>> >
>>> > -Avi
>>> >
>>> >
>>> > On Thu, Aug 11, 2011 at 10:18 PM, vip killa <vipkilla at gmail.com>
>>> wrote:
>>> >>
>>> >> I can't seem to figure out how to store what is collected
>>> >> from bind_digit_action into a channel variable... any ideas?
>>> >>
>>> >> On Thu, Aug 11, 2011 at 3:07 PM, Avi Marcus <avi at avimarcus.net>
>>> wrote:
>>> >>>
>>> >>> So some sort of bind_digit_action to collect and use that to set a
>>> hangup
>>> >>> hook. I then execute_extension from the action will be the easiest
>>> way to do
>>> >>> it, because you have several actions to accomplish.
>>> >>> However, doing this while the other person is on the hook.. sounds
>>> like a
>>> >>> bad idea.
>>> >>> -Avi
>>> >>>
>>> >>> On Thu, Aug 11, 2011 at 9:13 PM, vip killa <vipkilla at gmail.com>
>>> wrote:
>>> >>>>
>>> >>>> thanks but i'm not interested in paying someone to do this. but yes
>>> i
>>> >>>> simply want to collect the digits during the call from the B leg
>>> then store
>>> >>>> them in a variable which will rename the file accordingly from the
>>> hangup
>>> >>>> hook
>>> >>>>
>>> >>>> On Thu, Aug 11, 2011 at 2:07 PM, Avi Marcus <avi at avimarcus.net>
>>> wrote:
>>> >>>>>
>>> >>>>> Oh, you don't want the info in the CDR, you want the info in the
>>> >>>>> recording. Does it have to be in the name, or is an audio tag
>>> enough? I'm
>>> >>>>> not sure if tags can work once the recording started...
>>> >>>>> OK, so you can set a hangup hook to rename the file and add the
>>> >>>>> variable with the account code to the end.
>>> >>>>> Whatever you use to capture the DTMF - bind_digit_action or
>>> >>>>> playandgetdigits or anything can set a variable.
>>> >>>>> -Avi Marcus
>>> >>>>>
>>> >>>>> p.s. If you want me to... actually do it for you, I'm available for
>>> >>>>> consulting.. email me offlist.
>>> >>>>>
>>> >>>>> On Thu, Aug 11, 2011 at 8:52 PM, vip killa <vipkilla at gmail.com>
>>> wrote:
>>> >>>>>>
>>> >>>>>> I apologize but the caller would not know the account number, it
>>> would
>>> >>>>>> be for internal use only. how could xml_cdrs and mod_cdr_csv be
>>> used to
>>> >>>>>> stamp a recording? how do you store the DTMF in a channel
>>> variable?
>>> >>>>>>
>>> >>>>>> On Thu, Aug 11, 2011 at 12:41 PM, Avi Marcus <avi at avimarcus.net>
>>> >>>>>> wrote:
>>> >>>>>>>
>>> >>>>>>> You can set any channel variable from the dialplan or from a lua
>>> >>>>>>> script.
>>> >>>>>>> I'd reccommend a standalone lua script before the agent gets on
>>> the
>>> >>>>>>> phone, which can verify their account number and then present it
>>> to the
>>> >>>>>>> agent even. But yes, you can do anything with the dtmf including
>>> setting a
>>> >>>>>>> variable which can be saved from the xml_cdrs or mod_cdr_csv or
>>> whatever
>>> >>>>>>> other cdr system you use.
>>> >>>>>>> -Avi
>>> >>>>>>>
>>> >>>>>>> On Thu, Aug 11, 2011 at 7:30 PM, vip killa <vipkilla at gmail.com>
>>> >>>>>>> wrote:
>>> >>>>>>>>
>>> >>>>>>>> indeed it may be easier but this is what the client is asking
>>> >>>>>>>> for.... i know it has to be possible... i just need some
>>> direction.
>>> >>>>>>>>
>>> >>>>>>>> On Thu, Aug 11, 2011 at 12:22 PM, <covici at ccs.covici.com>
>>> wrote:
>>> >>>>>>>>>
>>> >>>>>>>>> Could not the agent just type it on a screen -- it would seem
>>> to be
>>> >>>>>>>>> much
>>> >>>>>>>>> easier.
>>> >>>>>>>>>
>>> >>>>>>>>> vip killa <vipkilla at gmail.com> wrote:
>>> >>>>>>>>>
>>> >>>>>>>>> > Hi everyone,
>>> >>>>>>>>> > I'm trying to collect DTMF digits and store them in a channel
>>> >>>>>>>>> > variable so
>>> >>>>>>>>> > when the channel hangs up it uses those digits to "mark" (or
>>> >>>>>>>>> > rename) the
>>> >>>>>>>>> > recording of the call. The DTMF will be entered by the called
>>> >>>>>>>>> > party (i think
>>> >>>>>>>>> > that would be the B-leg?). I've been experimenting with
>>> >>>>>>>>> > "bind_meta_app" and
>>> >>>>>>>>> > "bind_digit_action", it seems like "bind_digit_action" may be
>>> the
>>> >>>>>>>>> > one i need
>>> >>>>>>>>> > to use but im not sure. I'll explain the scenario to make
>>> things
>>> >>>>>>>>> > more
>>> >>>>>>>>> > clear...we are trying to install this in a call center type
>>> >>>>>>>>> > environment
>>> >>>>>>>>> > where all calls are being recorded. A caller gets an agent,
>>> the
>>> >>>>>>>>> > caller gives
>>> >>>>>>>>> > the agent the account number of their case, the agent uses
>>> DTMF
>>> >>>>>>>>> > to mark the
>>> >>>>>>>>> > recording of the call with the account number of the caller's
>>> >>>>>>>>> > case. Does
>>> >>>>>>>>> > that make sense? Please let me know if this is possible,
>>> thanks.
>>> >>>>>>>>> >
>>> >>>>>>>>> > ----------------------------------------------------
>>> >>>>>>>>> > Alternatives:
>>> >>>>>>>>> >
>>> >>>>>>>>> > ----------------------------------------------------
>>> >>>>>>>>> > _______________________________________________
>>> >>>>>>>>> > Join us at ClueCon 2011, Aug 9-11, Chicago
>>> >>>>>>>>> > http://www.cluecon.com 877-7-4ACLUE
>>> >>>>>>>>> >
>>> >>>>>>>>> > FreeSWITCH-users mailing list
>>> >>>>>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>> >>>>>>>>> >
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>>>>>>> >
>>> >>>>>>>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>>>>>>>> > http://www.freeswitch.org
>>> >>>>>>>>>
>>> >>>>>>>>> --
>>> >>>>>>>>> Your life is like a penny.  You're going to lose it.  The
>>> question
>>> >>>>>>>>> is:
>>> >>>>>>>>> How do
>>> >>>>>>>>> you spend it?
>>> >>>>>>>>>
>>> >>>>>>>>>         John Covici
>>> >>>>>>>>>         covici at ccs.covici.com
>>> >>>>>>>>>
>>> >>>>>>>>> _______________________________________________
>>> >>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> >>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>> >>>>>>>>>
>>> >>>>>>>>> FreeSWITCH-users mailing list
>>> >>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>> >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>>>>>>>
>>> >>>>>>>>> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>>>>>>>> http://www.freeswitch.org
>>> >>>>>>>>
>>> >>>>>>>>
>>> >>>>>>>> _______________________________________________
>>> >>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> >>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>> >>>>>>>>
>>> >>>>>>>> FreeSWITCH-users mailing list
>>> >>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>> >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>>>>>>
>>> >>>>>>>> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>>>>>>> http://www.freeswitch.org
>>> >>>>>>>>
>>> >>>>>>>
>>> >>>>>>>
>>> >>>>>>> _______________________________________________
>>> >>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> >>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>> >>>>>>>
>>> >>>>>>> FreeSWITCH-users mailing list
>>> >>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>> >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>>>>>
>>> >>>>>>> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>>>>>> http://www.freeswitch.org
>>> >>>>>>>
>>> >>>>>>
>>> >>>>>>
>>> >>>>>> _______________________________________________
>>> >>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> >>>>>> http://www.cluecon.com 877-7-4ACLUE
>>> >>>>>>
>>> >>>>>> FreeSWITCH-users mailing list
>>> >>>>>> FreeSWITCH-users at lists.freeswitch.org
>>> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>>>>
>>> >>>>>> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>>>>> http://www.freeswitch.org
>>> >>>>>>
>>> >>>>>
>>> >>>>>
>>> >>>>> _______________________________________________
>>> >>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> >>>>> http://www.cluecon.com 877-7-4ACLUE
>>> >>>>>
>>> >>>>> FreeSWITCH-users mailing list
>>> >>>>> FreeSWITCH-users at lists.freeswitch.org
>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>>>
>>> >>>>> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>>>> http://www.freeswitch.org
>>> >>>>>
>>> >>>>
>>> >>>>
>>> >>>> _______________________________________________
>>> >>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> >>>> http://www.cluecon.com 877-7-4ACLUE
>>> >>>>
>>> >>>> FreeSWITCH-users mailing list
>>> >>>> FreeSWITCH-users at lists.freeswitch.org
>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>>> http://www.freeswitch.org
>>> >>>>
>>> >>>
>>> >>>
>>> >>> _______________________________________________
>>> >>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> >>> http://www.cluecon.com 877-7-4ACLUE
>>> >>>
>>> >>> FreeSWITCH-users mailing list
>>> >>> FreeSWITCH-users at lists.freeswitch.org
>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>> http://www.freeswitch.org
>>> >>>
>>> >>
>>> >>
>>> >> _______________________________________________
>>> >> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> >> http://www.cluecon.com 877-7-4ACLUE
>>> >>
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >>
>>> >
>>> >
>>> > _______________________________________________
>>> > Join us at ClueCon 2011, Aug 9-11, Chicago
>>> > http://www.cluecon.com 877-7-4ACLUE
>>> >
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> 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
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110812/f9b439d2/attachment-0001.html 

From vipkilla at gmail.com  Fri Aug 12 21:14:59 2011
From: vipkilla at gmail.com (vip killa)
Date: Fri, 12 Aug 2011 13:14:59 -0400
Subject: [Freeswitch-users] collect dtmf and store in channel variable
In-Reply-To: <CAC-LwPN0DJRjtgYEnyOavYBB4-NqdzU4Ou-ecWwXEteFdiVK6A@mail.gmail.com>
References: <CAC-LwPOmcTxMZYUZKny2mfryY1MVvqJ9mWrtqS+VGV+G_cis2w@mail.gmail.com>
	<23723.1313079736@ccs.covici.com>
	<CAC-LwPOSH8FxVoJB1PejS=+j==JY515u=_FDFo7M_0LeAG32nA@mail.gmail.com>
	<CALNZuw=kwNEcAaN+-oxrkOM4OxWiAc2Koryct1YbuhkMA28_MA@mail.gmail.com>
	<CAC-LwPOpgRcwhAm1Q3Y9VQ+X5qn+Wx1tQGxzesOsWm7EPJCKvQ@mail.gmail.com>
	<CALNZuw=ncGwazLjZ-gRdG1=L1oa_0p_26sE_j7mzj0u5DkT0DQ@mail.gmail.com>
	<CAC-LwPP9aWzJs1C9C-t5f+oqSZcf3rF0WSaFN4cSWZ+J29+5ug@mail.gmail.com>
	<CALNZuwnF7hoysisyo2TQ40TFc17sfxk_0AH7PgBHwLzfnrqT+w@mail.gmail.com>
	<CAC-LwPP8_SgoLwjoYKec6buMN0Jz14=F86AS0QpTJfRobXTesQ@mail.gmail.com>
	<CALNZuwkBHptWyKCWYb=X-ApcP79BxFWeySCq7SEKvFJhMdtDNQ@mail.gmail.com>
	<CAKbxfG8A3XNoEhXfQwWNPwEjCKtU=sY_xWUpAEABZbzcqyxdNg@mail.gmail.com>
	<CAC-LwPObMxhBsTHHQnEiWhsXO8AARQucfadvbO-JiY8CHinjsA@mail.gmail.com>
	<CAC-LwPN_bJEYuK3Za28C5ip3JiDRErtKVb9q0C=yNy1Qzwx+BA@mail.gmail.com>
	<CAC-LwPN0DJRjtgYEnyOavYBB4-NqdzU4Ou-ecWwXEteFdiVK6A@mail.gmail.com>
Message-ID: <CAC-LwPPNq0-CDjiNMTVeD3LVKyK1XmtP0ToiFQkhqroPy+KppA@mail.gmail.com>

lol, nevermind, i missed the variable! thanks for everyone's help, consider
this issue resolved!

On Fri, Aug 12, 2011 at 1:12 PM, vip killa <vipkilla at gmail.com> wrote:

> I've been informed that i need to use xml_cdr to log the B-leg. But i dont
> see a way to log the digits_dialed variable using xml_cdr. with cdr_csv you
> use templates. is there something similar with xml_cdr? Thanks for everyones
> help so far.
>
>
> On Fri, Aug 12, 2011 at 8:23 AM, vip killa <vipkilla at gmail.com> wrote:
>
>> I apologize, I got that to work by adding ${digits_dialed} to
>> cdr_csv.conf.xml
>> Still, it only records the digits from the A leg, anyway to make it record
>> digits from the B leg?
>>
>> On Fri, Aug 12, 2011 at 8:16 AM, vip killa <vipkilla at gmail.com> wrote:
>>
>>> I just looked at the CDR CSVs, I don't see the digits dialed in there...
>>> there is nothing in wiki on how to use this either. How can I see the digits_dialed
>>> variable in the CDR?
>>>
>>>
>>> On Thu, Aug 11, 2011 at 6:32 PM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>>> digits_dialed variable available in the cdr will contain any digits
>>>> dialed the whole call duration.
>>>>
>>>>
>>>> On Thu, Aug 11, 2011 at 3:15 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>>> > According to the wiki, apparently, you can't capture digits from
>>>> > bind_digit_action. So you'll have to use that to trigger
>>>> > a Dialplan_Tools_play_and_get_digits which automatically sets a
>>>> variable, or
>>>> > getdigits within lua/js (only?) and then save it.
>>>> >
>>>> > -Avi
>>>> >
>>>> >
>>>> > On Thu, Aug 11, 2011 at 10:18 PM, vip killa <vipkilla at gmail.com>
>>>> wrote:
>>>> >>
>>>> >> I can't seem to figure out how to store what is collected
>>>> >> from bind_digit_action into a channel variable... any ideas?
>>>> >>
>>>> >> On Thu, Aug 11, 2011 at 3:07 PM, Avi Marcus <avi at avimarcus.net>
>>>> wrote:
>>>> >>>
>>>> >>> So some sort of bind_digit_action to collect and use that to set a
>>>> hangup
>>>> >>> hook. I then execute_extension from the action will be the easiest
>>>> way to do
>>>> >>> it, because you have several actions to accomplish.
>>>> >>> However, doing this while the other person is on the hook.. sounds
>>>> like a
>>>> >>> bad idea.
>>>> >>> -Avi
>>>> >>>
>>>> >>> On Thu, Aug 11, 2011 at 9:13 PM, vip killa <vipkilla at gmail.com>
>>>> wrote:
>>>> >>>>
>>>> >>>> thanks but i'm not interested in paying someone to do this. but yes
>>>> i
>>>> >>>> simply want to collect the digits during the call from the B leg
>>>> then store
>>>> >>>> them in a variable which will rename the file accordingly from the
>>>> hangup
>>>> >>>> hook
>>>> >>>>
>>>> >>>> On Thu, Aug 11, 2011 at 2:07 PM, Avi Marcus <avi at avimarcus.net>
>>>> wrote:
>>>> >>>>>
>>>> >>>>> Oh, you don't want the info in the CDR, you want the info in the
>>>> >>>>> recording. Does it have to be in the name, or is an audio tag
>>>> enough? I'm
>>>> >>>>> not sure if tags can work once the recording started...
>>>> >>>>> OK, so you can set a hangup hook to rename the file and add the
>>>> >>>>> variable with the account code to the end.
>>>> >>>>> Whatever you use to capture the DTMF - bind_digit_action or
>>>> >>>>> playandgetdigits or anything can set a variable.
>>>> >>>>> -Avi Marcus
>>>> >>>>>
>>>> >>>>> p.s. If you want me to... actually do it for you, I'm available
>>>> for
>>>> >>>>> consulting.. email me offlist.
>>>> >>>>>
>>>> >>>>> On Thu, Aug 11, 2011 at 8:52 PM, vip killa <vipkilla at gmail.com>
>>>> wrote:
>>>> >>>>>>
>>>> >>>>>> I apologize but the caller would not know the account number, it
>>>> would
>>>> >>>>>> be for internal use only. how could xml_cdrs and mod_cdr_csv be
>>>> used to
>>>> >>>>>> stamp a recording? how do you store the DTMF in a channel
>>>> variable?
>>>> >>>>>>
>>>> >>>>>> On Thu, Aug 11, 2011 at 12:41 PM, Avi Marcus <avi at avimarcus.net>
>>>> >>>>>> wrote:
>>>> >>>>>>>
>>>> >>>>>>> You can set any channel variable from the dialplan or from a lua
>>>> >>>>>>> script.
>>>> >>>>>>> I'd reccommend a standalone lua script before the agent gets on
>>>> the
>>>> >>>>>>> phone, which can verify their account number and then present it
>>>> to the
>>>> >>>>>>> agent even. But yes, you can do anything with the dtmf including
>>>> setting a
>>>> >>>>>>> variable which can be saved from the xml_cdrs or mod_cdr_csv or
>>>> whatever
>>>> >>>>>>> other cdr system you use.
>>>> >>>>>>> -Avi
>>>> >>>>>>>
>>>> >>>>>>> On Thu, Aug 11, 2011 at 7:30 PM, vip killa <vipkilla at gmail.com>
>>>> >>>>>>> wrote:
>>>> >>>>>>>>
>>>> >>>>>>>> indeed it may be easier but this is what the client is asking
>>>> >>>>>>>> for.... i know it has to be possible... i just need some
>>>> direction.
>>>> >>>>>>>>
>>>> >>>>>>>> On Thu, Aug 11, 2011 at 12:22 PM, <covici at ccs.covici.com>
>>>> wrote:
>>>> >>>>>>>>>
>>>> >>>>>>>>> Could not the agent just type it on a screen -- it would seem
>>>> to be
>>>> >>>>>>>>> much
>>>> >>>>>>>>> easier.
>>>> >>>>>>>>>
>>>> >>>>>>>>> vip killa <vipkilla at gmail.com> wrote:
>>>> >>>>>>>>>
>>>> >>>>>>>>> > Hi everyone,
>>>> >>>>>>>>> > I'm trying to collect DTMF digits and store them in a
>>>> channel
>>>> >>>>>>>>> > variable so
>>>> >>>>>>>>> > when the channel hangs up it uses those digits to "mark" (or
>>>> >>>>>>>>> > rename) the
>>>> >>>>>>>>> > recording of the call. The DTMF will be entered by the
>>>> called
>>>> >>>>>>>>> > party (i think
>>>> >>>>>>>>> > that would be the B-leg?). I've been experimenting with
>>>> >>>>>>>>> > "bind_meta_app" and
>>>> >>>>>>>>> > "bind_digit_action", it seems like "bind_digit_action" may
>>>> be the
>>>> >>>>>>>>> > one i need
>>>> >>>>>>>>> > to use but im not sure. I'll explain the scenario to make
>>>> things
>>>> >>>>>>>>> > more
>>>> >>>>>>>>> > clear...we are trying to install this in a call center type
>>>> >>>>>>>>> > environment
>>>> >>>>>>>>> > where all calls are being recorded. A caller gets an agent,
>>>> the
>>>> >>>>>>>>> > caller gives
>>>> >>>>>>>>> > the agent the account number of their case, the agent uses
>>>> DTMF
>>>> >>>>>>>>> > to mark the
>>>> >>>>>>>>> > recording of the call with the account number of the
>>>> caller's
>>>> >>>>>>>>> > case. Does
>>>> >>>>>>>>> > that make sense? Please let me know if this is possible,
>>>> thanks.
>>>> >>>>>>>>> >
>>>> >>>>>>>>> > ----------------------------------------------------
>>>> >>>>>>>>> > Alternatives:
>>>> >>>>>>>>> >
>>>> >>>>>>>>> > ----------------------------------------------------
>>>> >>>>>>>>> > _______________________________________________
>>>> >>>>>>>>> > Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> >>>>>>>>> > http://www.cluecon.com 877-7-4ACLUE
>>>> >>>>>>>>> >
>>>> >>>>>>>>> > FreeSWITCH-users mailing list
>>>> >>>>>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> >>>>>>>>> >
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >>>>>>>>> >
>>>> >>>>>>>>> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >>>>>>>>> > http://www.freeswitch.org
>>>> >>>>>>>>>
>>>> >>>>>>>>> --
>>>> >>>>>>>>> Your life is like a penny.  You're going to lose it.  The
>>>> question
>>>> >>>>>>>>> is:
>>>> >>>>>>>>> How do
>>>> >>>>>>>>> you spend it?
>>>> >>>>>>>>>
>>>> >>>>>>>>>         John Covici
>>>> >>>>>>>>>         covici at ccs.covici.com
>>>> >>>>>>>>>
>>>> >>>>>>>>> _______________________________________________
>>>> >>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> >>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>> >>>>>>>>>
>>>> >>>>>>>>> FreeSWITCH-users mailing list
>>>> >>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >>>>>>>>>
>>>> >>>>>>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >>>>>>>>> http://www.freeswitch.org
>>>> >>>>>>>>
>>>> >>>>>>>>
>>>> >>>>>>>> _______________________________________________
>>>> >>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> >>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>> >>>>>>>>
>>>> >>>>>>>> FreeSWITCH-users mailing list
>>>> >>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >>>>>>>>
>>>> >>>>>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >>>>>>>> http://www.freeswitch.org
>>>> >>>>>>>>
>>>> >>>>>>>
>>>> >>>>>>>
>>>> >>>>>>> _______________________________________________
>>>> >>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> >>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>> >>>>>>>
>>>> >>>>>>> FreeSWITCH-users mailing list
>>>> >>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >>>>>>>
>>>> >>>>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >>>>>>> http://www.freeswitch.org
>>>> >>>>>>>
>>>> >>>>>>
>>>> >>>>>>
>>>> >>>>>> _______________________________________________
>>>> >>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> >>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>> >>>>>>
>>>> >>>>>> FreeSWITCH-users mailing list
>>>> >>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >>>>>>
>>>> >>>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >>>>>> http://www.freeswitch.org
>>>> >>>>>>
>>>> >>>>>
>>>> >>>>>
>>>> >>>>> _______________________________________________
>>>> >>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> >>>>> http://www.cluecon.com 877-7-4ACLUE
>>>> >>>>>
>>>> >>>>> FreeSWITCH-users mailing list
>>>> >>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >>>>>
>>>> >>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >>>>> http://www.freeswitch.org
>>>> >>>>>
>>>> >>>>
>>>> >>>>
>>>> >>>> _______________________________________________
>>>> >>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> >>>> http://www.cluecon.com 877-7-4ACLUE
>>>> >>>>
>>>> >>>> FreeSWITCH-users mailing list
>>>> >>>> FreeSWITCH-users at lists.freeswitch.org
>>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >>>> http://www.freeswitch.org
>>>> >>>>
>>>> >>>
>>>> >>>
>>>> >>> _______________________________________________
>>>> >>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> >>> http://www.cluecon.com 877-7-4ACLUE
>>>> >>>
>>>> >>> FreeSWITCH-users mailing list
>>>> >>> FreeSWITCH-users at lists.freeswitch.org
>>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >>> http://www.freeswitch.org
>>>> >>>
>>>> >>
>>>> >>
>>>> >> _______________________________________________
>>>> >> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> >> http://www.cluecon.com 877-7-4ACLUE
>>>> >>
>>>> >> FreeSWITCH-users mailing list
>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> http://www.freeswitch.org
>>>> >>
>>>> >
>>>> >
>>>> > _______________________________________________
>>>> > Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> > http://www.cluecon.com 877-7-4ACLUE
>>>> >
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>> >
>>>> >
>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> 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
>>>> googletalk:conf+888 at conference.freeswitch.org
>>>> pstn:+19193869900
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110812/dde35b93/attachment-0001.html 

From juanito1982 at gmail.com  Fri Aug 12 22:42:38 2011
From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=)
Date: Fri, 12 Aug 2011 20:42:38 +0200
Subject: [Freeswitch-users] Newbie - error when calling with new Yealink
 T20P phone
In-Reply-To: <00e601cc590f$e3db66d0$ab923470$@accra.ca>
References: <00e601cc590f$e3db66d0$ab923470$@accra.ca>
Message-ID: <CAHLYDCuacB4AyRu3EnyjuG9-HanwTSyevtKHgFnV=wcq=B54zg@mail.gmail.com>

Try to add G729 to you inbound-codec-prefs var in vars.xml or to select one
codec different to G729 in your T20P. Remember you will cannot do G729
transcoding without buying some G729 licenses.

Regards

2011/8/12 Charles Bujold <cjbujold at accra.ca>

> Just installed a new phone Yealink T20P  at another location and it
> registers with Freeswitch.  However when I try to check voice messages or
> make calls I get the following error.  If somebody could help me identify
> and fix the issue it would be most appreciated.  (xxx hides the IP)****
>
> ** **
>
> Thanks****
>
> ** **
>
> CJB****
>
> ** **
>
> 2011-08-10 10:56:57.711199 [DEBUG] switch_rtp.c:3105 Correct ip/port
> confirmed.****
>
> 2011-08-10 10:57:30.191234 [WARNING] sofia_reg.c:1339 SIP auth challenge
> (REGISTER) on sofia profile 'internal' for [401 at sip.accra.com] from ip
> 142.166.219.xxx****
>
> 2011-08-10 11:00:31.171255 [WARNING] sofia_reg.c:1339 SIP auth challenge
> (REGISTER) on sofia profile 'internal' for [401 at sip.accra.com] from ip
> 142.166.219. xxx****
>
> 2011-08-10 11:03:28.691243 [WARNING] sofia_reg.c:1339 SIP auth challenge
> (REGISTER) on sofia profile 'internal' for [400 at sip.accra.com] from ip
> 142.166.219. xxx****
>
> 2011-08-10 11:03:40.271249 [DEBUG] sofia.c:7069 IP 142.166.219. xxx
> Rejected by acl "domains". Falling back to Digest auth.****
>
> 2011-08-10 11:03:40.271249 [WARNING] sofia_reg.c:1339 SIP auth challenge
> (INVITE) on sofia profile 'internal' for [*98 at sip.accra.com] from ip
> 142.166.219. xxx****
>
> 2011-08-10 11:03:40.291256 [WARNING] sofia_reg.c:1339 SIP auth challenge
> (REGISTER) on sofia profile 'internal' for [401 at sip.accra.com] from ip
> 142.166.219. xxx****
>
> 2011-08-10 11:03:40.431248 [DEBUG] sofia.c:7069 IP 142.166.219. xxx
> Rejected by acl "domains". Falling back to Digest auth.****
>
> 2011-08-10 11:03:40.431248 [NOTICE] switch_channel.c:904 New Channel
> sofia/internal/401 at sip.accra.com [5aec6826-e15b-4ce8-bffe-6a218fc0030f]***
> *
>
> 2011-08-10 11:03:40.431248 [DEBUG] switch_core_state_machine.c:330
> (sofia/internal/401 at sip.accra.com) Running State Change CS_NEW****
>
> 2011-08-10 11:03:40.431248 [DEBUG] switch_core_state_machine.c:348
> (sofia/internal/401 at sip.accra.com) State NEW****
>
> 2011-08-10 11:03:40.431248 [DEBUG] sofia.c:5119 Channel sofia/internal/
> 401 at sip.accra.com entering state [received][100]****
>
> 2011-08-10 11:03:40.431248 [DEBUG] sofia.c:5130 Remote SDP:****
>
> v=0****
>
> o=- 20001 20001 IN IP4 142.166.219. xxx****
>
> s=SDP data****
>
> c=IN IP4 142.166.219. xxx****
>
> t=0 0****
>
> m=audio 11782 RTP/AVP 18 101****
>
> a=rtpmap:18 G729/8000****
>
> a=fmtp:18 annexb=no****
>
> a=rtpmap:101 telephone-event/8000****
>
> a=fmtp:101 0-15****
>
> a=ptime:20****
>
> ** **
>
> 2011-08-10 11:03:40.431248 [DEBUG] sofia_glue.c:4731 Audio Codec Compare
> [G729:18:8000:20:8000]/[G722:9:8000:20:64000]****
>
> 2011-08-10 11:03:40.431248 [DEBUG] sofia_glue.c:4731 Audio Codec Compare
> [G729:18:8000:20:8000]/[PCMU:0:8000:20:64000]****
>
> 2011-08-10 11:03:40.431248 [DEBUG] sofia_glue.c:4731 Audio Codec Compare
> [G729:18:8000:20:8000]/[PCMA:8:8000:20:64000]****
>
> 2011-08-10 11:03:40.431248 [DEBUG] sofia_glue.c:4731 Audio Codec Compare
> [G729:18:8000:20:8000]/[GSM:3:8000:20:13200]****
>
> 2011-08-10 11:03:40.431248 [DEBUG] sofia_glue.c:4731 Audio Codec Compare
> [telephone-event:101:8000:20:0]/[G722:9:8000:20:64000]****
>
> 2011-08-10 11:03:40.431248 [DEBUG] sofia_glue.c:4731 Audio Codec Compare
> [telephone-event:101:8000:20:0]/[PCMU:0:8000:20:64000]****
>
> 2011-08-10 11:03:40.431248 [DEBUG] sofia_glue.c:4731 Audio Codec Compare
> [telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000]****
>
> 2011-08-10 11:03:40.431248 [DEBUG] sofia_glue.c:4731 Audio Codec Compare
> [telephone-event:101:8000:20:0]/[GSM:3:8000:20:13200]****
>
> 2011-08-10 11:03:40.431248 [DEBUG] sofia_glue.c:4845 Set 2833 dtmf
> send/recv payload to 101****
>
> 2011-08-10 11:03:40.431248 [DEBUG] switch_channel.c:2767 (sofia/internal/
> 401 at sip.accra.com) Callstate Change DOWN -> HANGUP****
>
> 2011-08-10 11:03:40.431248 [NOTICE] sofia.c:5374 Hangup sofia/internal/
> 401 at sip.accra.com [CS_NEW] [INCOMPATIBLE_DESTINATION]****
>
> 2011-08-10 11:03:40.431248 [DEBUG] switch_channel.c:2783 Send signal
> sofia/internal/401 at sip.accra.com [KILL]****
>
> 2011-08-10 11:03:40.431248 [DEBUG] switch_core_session.c:1156 Send signal
> sofia/internal/401 at sip.accra.com [BREAK]****
>
> 2011-08-10 11:03:40.431248 [DEBUG] switch_core_state_machine.c:330
> (sofia/internal/401 at sip.accra.com) Running State Change CS_HANGUP****
>
> 2011-08-10 11:03:40.431248 [DEBUG] switch_core_state_machine.c:580
> (sofia/internal/401 at sip.accra.com) State HANGUP****
>
> 2011-08-10 11:03:40.431248 [DEBUG] mod_sofia.c:458 Channel sofia/internal/
> 401 at sip.accra.com hanging up, cause: INCOMPATIBLE_DESTINATION****
>
> 2011-08-10 11:03:40.431248 [DEBUG] mod_sofia.c:522 Responding to INVITE
> with: 488****
>
> 2011-08-10 11:03:40.431248 [DEBUG] switch_core_state_machine.c:46
> sofia/internal/401 at sip.accra.com Standard HANGUP, cause:
> INCOMPATIBLE_DESTINATION****
>
> 2011-08-10 11:03:40.431248 [DEBUG] switch_core_state_machine.c:580
> (sofia/internal/401 at sip.accra.com) State HANGUP going to sleep****
>
> 2011-08-10 11:03:40.431248 [DEBUG] switch_core_state_machine.c:361
> (sofia/internal/401 at sip.accra.com) State Change CS_HANGUP -> CS_REPORTING*
> ***
>
> 2011-08-10 11:03:40.431248 [DEBUG] switch_core_session.c:1156 Send signal
> sofia/internal/401 at sip.accra.com [BREAK]****
>
> 2011-08-10 11:03:40.431248 [DEBUG] switch_core_state_machine.c:330
> (sofia/internal/401 at sip.accra.com) Running State Change CS_REPORTING****
>
> 2011-08-10 11:03:40.431248 [DEBUG] switch_core_state_machine.c:640
> (sofia/internal/401 at sip.accra.com) State REPORTING****
>
> 2011-08-10 11:03:40.611244 [DEBUG] switch_core_state_machine.c:53
> sofia/internal/401 at sip.accra.com Standard REPORTING, cause:
> INCOMPATIBLE_DESTINATION****
>
> 2011-08-10 11:03:40.611244 [DEBUG] switch_core_state_machine.c:640
> (sofia/internal/401 at sip.accra.com) State REPORTING going to sleep****
>
> 2011-08-10 11:03:40.611244 [DEBUG] switch_core_state_machine.c:355
> (sofia/internal/401 at sip.accra.com) State Change CS_REPORTING -> CS_DESTROY
> ****
>
> 2011-08-10 11:03:40.611244 [DEBUG] switch_core_session.c:1156 Send signal
> sofia/internal/401 at sip.accra.com [BREAK]****
>
> 2011-08-10 11:03:40.611244 [DEBUG] switch_core_session.c:1328 Session 64
> (sofia/internal/401 at sip.accra.com) Locked, Waiting on external entities***
> *
>
> 2011-08-10 11:03:40.611244 [NOTICE] switch_core_session.c:1346 Session 64
> (sofia/internal/401 at sip.accra.com) Ended****
>
> 2011-08-10 11:03:40.611244 [NOTICE] switch_core_session.c:1348 Close
> Channel sofia/internal/401 at sip.accra.com [CS_DESTROY]****
>
> 2011-08-10 11:03:40.611244 [DEBUG] switch_core_state_machine.c:469
> (sofia/internal/401 at sip.accra.com) Callstate Change HANGUP -> DOWN****
>
> 2011-08-10 11:03:40.611244 [DEBUG] switch_core_state_machine.c:472
> (sofia/internal/401 at sip.accra.com) Running State Change CS_DESTROY****
>
> 2011-08-10 11:03:40.611244 [DEBUG] switch_core_state_machine.c:482
> (sofia/internal/401 at sip.accra.com) State DESTROY****
>
> 2011-08-10 11:03:40.611244 [DEBUG] mod_sofia.c:363 sofia/internal/
> 401 at sip.accra.com SOFIA DESTROY****
>
> 2011-08-10 11:03:40.611244 [DEBUG] switch_core_state_machine.c:60
> sofia/internal/401 at sip.accra.com Standard DESTROY****
>
> 2011-08-10 11:03:40.611244 [DEBUG] switch_core_state_machine.c:482
> (sofia/internal/401 at sip.accra.com) State DESTROY going to sleep****
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110812/40f8b8f2/attachment.html 

From mi.ke at null.net  Sat Aug 13 13:46:23 2011
From: mi.ke at null.net (Mi Ke)
Date: Sat, 13 Aug 2011 09:46:23 +0000
Subject: [Freeswitch-users] getting disconnect cause for a leg after bridge
	in Lua
Message-ID: <20110813094624.167960@gmx.com>

Hi All,

 Is there any way to get a real disconnection cause for leg B in the following script ?


 if (session_a:ready() and session_b:ready()) then

 freeswitch.bridge(session_a,session_b)

 -- session_b gets disconnect here ...

 local session_b_hangup_cause = session_b:hangupCause()



 session_b_hangup_cause is always "SUCCESS" after debridging while log and CDR shows correct value - can get it to my script ?

 Thanks / Mike
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110813/4329e421/attachment.html 

From michal.bielicki at seventhsignal.de  Sat Aug 13 15:03:33 2011
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Sat, 13 Aug 2011 13:03:33 +0200
Subject: [Freeswitch-users] Meetup Paris?
In-Reply-To: <CAO+9iGe4M9eep9pANrFK-sLHzB=342BYMQbBzzzYop2MKUV2yQ@mail.gmail.com>
References: <E58D1B62-B7E9-4D15-A0AF-DC993AA70786@avgs.ca>
	<CAFYZzbvqVs_gFNKuoORNxTBd31Ya6AxSz3bz-MV3D-P8DCQJGg@mail.gmail.com>
	<B61CBF19-1706-4370-A2D2-546561708847@seventhsignal.de>
	<CALXCt0pcAUVq_qRKqkYEq4zZeiQzo9so+tdQGuZnTe1+ZWXHXg@mail.gmail.com>
	<CAO+9iGe4M9eep9pANrFK-sLHzB=342BYMQbBzzzYop2MKUV2yQ@mail.gmail.com>
Message-ID: <EF9CDA5D-34FB-4B13-9AFC-BA028E45048D@seventhsignal.de>

We have one, will publish details later. Sites are not an issue. People doing things are :)

Am 12.08.2011 um 18:47 schrieb Alec Taylor:

> Want me to setup a conference site for you guys?
> 
> Cost = references
> 
> :]
> 
> On Sat, Aug 13, 2011 at 2:29 AM, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
>> Yep, is due time to begin push FS interest in Europe.
>> I'm just arrived in Milan, so I'll miss the Paris bistrot, but I'll
>> see Math next week in Milan.
>> Let's start to conspire for a FS Europe!
>> 
>> -giovanni
>> 
>> On 8/12/11, Michal Bielicki <michal.bielicki at seventhsignal.de> wrote:
>>> There used to be something close organised by the German Unix User Group
>>> every Year. But it died off. It would be interesting to see how much
>>> interest would be there to reawake it, since I am evetl. thinking of doing
>>> something like that in germany next year.
>>> Am 12.08.2011 um 15:48 schrieb Michel Daggelinckx:
>>> 
>>>> A European version of cluecon would be nice.
>>>> 
>>>> Michel
>>>> 
>>>> On Fri, Aug 12, 2011 at 7:28 AM, Mathieu Rene <mrene_lists at avgs.ca> wrote:
>>>> English version follows.
>>>> 
>>>> En cette fin de ClueCon, je pars sur Paris demain soir (donc j'arrive
>>>> samedi en matin?, bien d?cal?), et j'y resterai jusqu'? mercredi.  Je
>>>> propose d'organiser un petit meetup afin de joindre l'utile ? l'agr?able
>>>> et de voir un peu le visage de la communaut? freeswitch parisienne. Est-ce
>>>> quelqu'un peut proposer un bar sympa ou on pourrait faire un truc?
>>>> 
>>>> ---
>>>> 
>>>> ClueCon is over and I'm leaving to Paris tomorrow (landing saturday
>>>> morning, well jetlagged) and staying until Wednesday. I'm proposing to
>>>> organize a little meetup in order to get to know the face of the parisian
>>>> freeswitch community. Does anyone know a bar where we could meet up?
>>>> 
>>>> Mathieu Rene
>>>> Avant-Garde Solutions Inc
>>>> Office: + 1 (514) 664-1044 x100
>>>> Cell: +1 (514) 664-1044 x200
>>>> mrene at avgs.ca
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> Michal Bielicki
>>> Gesch?ftsf?hrer / CEO
>>> 
>>> Seventh Signal Ltd. & Co. KG
>>> Weigandufer 45, B?ro 115, D-12059 Berlin
>>> Voice: +49 30 60988730
>>> 
>>> Amtsgericht Charlottenburg HRA 44413 B
>>> Ust.-ID: DE266981999
>>> Gesch?ftsf?hrer: Michal Bielicki
>>> Pers?nlich Haftende Gesellschafterin:
>>> Seventh Signal Ltd, 69 Great Hampton St. Birmingham,
>>> B18 6EW, GB, Company Nr.: 06889439
>>> WWW.: http://www.seventhsignal.de
>>> 
>>> 
>>> 
>>> ----
>>> 
>>> 
>> 
>> --
>> Sent from my mobile device
>> 
>> Sincerely,
>> 
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>> 
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Michal Bielicki
Gesch?ftsf?hrer / CEO

Seventh Signal Ltd. & Co. KG
Weigandufer 45, B?ro 115, D-12059 Berlin
Voice: +49 30 60988730

Amtsgericht Charlottenburg HRA 44413 B
Ust.-ID: DE266981999
Gesch?ftsf?hrer: Michal Bielicki
Pers?nlich Haftende Gesellschafterin:
Seventh Signal Ltd, 69 Great Hampton St. Birmingham, 
B18 6EW, GB, Company Nr.: 06889439
WWW.: http://www.seventhsignal.de



----

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110813/d1eafde2/attachment-0001.html 

From chrisbware at interfree.it  Sat Aug 13 17:41:46 2011
From: chrisbware at interfree.it (Chrisbware)
Date: Sat, 13 Aug 2011 15:41:46 +0200
Subject: [Freeswitch-users] Secure connection
Message-ID: <4E467F1A.6030600@interfree.it>

Just to have your opinion on it: I've a PHP site on a server, connected
to Freeswitch (mod_event_socket) on another server.
Which method would you recommend to protect data trasmitted between two
servers (they travel on internet)?


From 12ukwn at gmail.com  Sat Aug 13 18:08:17 2011
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Sat, 13 Aug 2011 16:08:17 +0200
Subject: [Freeswitch-users] Secure connection
In-Reply-To: <4E467F1A.6030600@interfree.it>
References: <4E467F1A.6030600@interfree.it>
Message-ID: <20110813160817.6a765e63@anubis.defcon1>

On Sat, 13 Aug 2011 15:41:46 +0200, Chrisbware <chrisbware at interfree.it> wrote:

> Just to have your opinion on it: I've a PHP site on a server, connected
> to Freeswitch (mod_event_socket) on another server.
> Which method would you recommend to protect data trasmitted between two
> servers (they travel on internet)?

stunnel in mode 2 (cli + svr certificates + full authentication)

-- 
In a great romance, each person basically plays a part that the
other really likes.
		-- Elizabeth Ashley


From alec.taylor6 at gmail.com  Sat Aug 13 20:03:09 2011
From: alec.taylor6 at gmail.com (Alec Taylor)
Date: Sun, 14 Aug 2011 02:03:09 +1000
Subject: [Freeswitch-users] Meetup Paris?
In-Reply-To: <EF9CDA5D-34FB-4B13-9AFC-BA028E45048D@seventhsignal.de>
References: <E58D1B62-B7E9-4D15-A0AF-DC993AA70786@avgs.ca>
	<CAFYZzbvqVs_gFNKuoORNxTBd31Ya6AxSz3bz-MV3D-P8DCQJGg@mail.gmail.com>
	<B61CBF19-1706-4370-A2D2-546561708847@seventhsignal.de>
	<CALXCt0pcAUVq_qRKqkYEq4zZeiQzo9so+tdQGuZnTe1+ZWXHXg@mail.gmail.com>
	<CAO+9iGe4M9eep9pANrFK-sLHzB=342BYMQbBzzzYop2MKUV2yQ@mail.gmail.com>
	<EF9CDA5D-34FB-4B13-9AFC-BA028E45048D@seventhsignal.de>
Message-ID: <CAO+9iGePMaeRfpeTrF-ES4vNUaGE5ku5N75gxr8EXThv6gdA0Q@mail.gmail.com>

k

On Sat, Aug 13, 2011 at 9:03 PM, Michal Bielicki
<michal.bielicki at seventhsignal.de> wrote:
> We have one, will publish details later. Sites are not an issue. People
> doing things are :)
> Am 12.08.2011 um 18:47 schrieb Alec Taylor:
>
> Want me to setup a conference site for you guys?
>
> Cost = references
>
> :]
>
> On Sat, Aug 13, 2011 at 2:29 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>
> Yep, is due time to begin push FS interest in Europe.
>
> I'm just arrived in Milan, so I'll miss the Paris bistrot, but I'll
>
> see Math next week in Milan.
>
> Let's start to conspire for a FS Europe!
>
> -giovanni
>
> On 8/12/11, Michal Bielicki <michal.bielicki at seventhsignal.de> wrote:
>
> There used to be something close organised by the German Unix User Group
>
> every Year. But it died off. It would be interesting to see how much
>
> interest would be there to reawake it, since I am evetl. thinking of doing
>
> something like that in germany next year.
>
> Am 12.08.2011 um 15:48 schrieb Michel Daggelinckx:
>
> A European version of cluecon would be nice.
>
> Michel
>
> On Fri, Aug 12, 2011 at 7:28 AM, Mathieu Rene <mrene_lists at avgs.ca> wrote:
>
> English version follows.
>
> En cette fin de ClueCon, je pars sur Paris demain soir (donc j'arrive
>
> samedi en matin?, bien d?cal?), et j'y resterai jusqu'? mercredi. ?Je
>
> propose d'organiser un petit meetup afin de joindre l'utile ? l'agr?able
>
> et de voir un peu le visage de la communaut? freeswitch parisienne. Est-ce
>
> quelqu'un peut proposer un bar sympa ou on pourrait faire un truc?
>
> ---
>
> ClueCon is over and I'm leaving to Paris tomorrow (landing saturday
>
> morning, well jetlagged) and staying until Wednesday. I'm proposing to
>
> organize a little meetup in order to get to know the face of the parisian
>
> freeswitch community. Does anyone know a bar where we could meet up?
>
> Mathieu Rene
>
> Avant-Garde Solutions Inc
>
> Office: + 1 (514) 664-1044 x100
>
> Cell: +1 (514) 664-1044 x200
>
> mrene at avgs.ca
>
>
>
>
>
> _______________________________________________
>
> Join us at ClueCon 2011, Aug 9-11, Chicago
>
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
> _______________________________________________
>
> Join us at ClueCon 2011, Aug 9-11, Chicago
>
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
> Michal Bielicki
>
> Gesch?ftsf?hrer / CEO
>
> Seventh Signal Ltd. & Co. KG
>
> Weigandufer 45, B?ro 115, D-12059 Berlin
>
> Voice: +49 30 60988730
>
> Amtsgericht Charlottenburg HRA 44413 B
>
> Ust.-ID: DE266981999
>
> Gesch?ftsf?hrer: Michal Bielicki
>
> Pers?nlich Haftende Gesellschafterin:
>
> Seventh Signal Ltd, 69 Great Hampton St. Birmingham,
>
> B18 6EW, GB, Company Nr.: 06889439
>
> WWW.: http://www.seventhsignal.de
>
>
>
> ----
>
>
>
> --
>
> Sent from my mobile device
>
> Sincerely,
>
> Giovanni Maruzzelli
>
> Cell : +39-347-2665618
>
> _______________________________________________
>
> Join us at ClueCon 2011, Aug 9-11, Chicago
>
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> Michal Bielicki
> Gesch?ftsf?hrer / CEO
> Seventh Signal Ltd. & Co. KG
> Weigandufer 45, B?ro 115,?D-12059 Berlin
> Voice: +49 30?60988730
> Amtsgericht Charlottenburg HRA 44413 B
> Ust.-ID: DE266981999
> Gesch?ftsf?hrer: Michal Bielicki
> Pers?nlich Haftende Gesellschafterin:
> Seventh Signal Ltd, 69 Great Hampton St. Birmingham,
> B18 6EW, GB, Company Nr.: 06889439
> WWW.:?http://www.seventhsignal.de
>
> ----
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From moises.silva at gmail.com  Sat Aug 13 20:48:15 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Sat, 13 Aug 2011 12:48:15 -0400
Subject: [Freeswitch-users] any IVR example in C/C++?
In-Reply-To: <4E43EE36.60209@hw.ac.uk>
References: <4E43EE36.60209@hw.ac.uk>
Message-ID: <CAA4nhyAnC0HfxJJPGpOdiBGaG__gBf6Bp9KVi6spjDQgjLTZ5w@mail.gmail.com>

On Thu, Aug 11, 2011 at 10:59 AM, xl127 <x.liu at hw.ac.uk> wrote:
> I am wondering how I could do this for a C/C++ application?
> And in the scripts languages I can set a callback method, e.g.
> ? session.setInputCallback(myInputCallback)
> but I didn't find how to do this in C/C++.

The default question here is, why do you need C/C++ for an IVR?
FreeSWITCH allows you to use simpler/safer languages to build IVR's.

You can certainly do it, but the reason you don't find examples is
probably because most people understand there is no need for C/C++
there. Having said that, you can take a look at the IVR/say/play API's
in switch_ivr_play_say.c to find out how to provide a callback to the
different API's thru the switch_input_args_t structure.

Moises Silva
Senior Software Engineer, Software Development Manager
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
L3R 9R6 Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com


From steveayre at gmail.com  Sat Aug 13 22:07:32 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 13 Aug 2011 19:07:32 +0100
Subject: [Freeswitch-users] Secure connection
In-Reply-To: <4E467F1A.6030600@interfree.it>
References: <4E467F1A.6030600@interfree.it>
Message-ID: <6B50778F-B8F2-49A0-AFB4-201B62AD488D@gmail.com>

It's a plaintext protocol so nothing you can do within FS to secure it. Firewall the ports to restrict external access and use a VPN if you need to encrypt the data.

Steve on iPhone

On 13 Aug 2011, at 14:41, Chrisbware <chrisbware at interfree.it> wrote:

> Just to have your opinion on it: I've a PHP site on a server, connected
> to Freeswitch (mod_event_socket) on another server.
> Which method would you recommend to protect data trasmitted between two
> servers (they travel on internet)?
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From curriegrad2004 at gmail.com  Sat Aug 13 22:49:55 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sat, 13 Aug 2011 11:49:55 -0700
Subject: [Freeswitch-users] Secure connection
In-Reply-To: <6B50778F-B8F2-49A0-AFB4-201B62AD488D@gmail.com>
References: <4E467F1A.6030600@interfree.it>
	<6B50778F-B8F2-49A0-AFB4-201B62AD488D@gmail.com>
Message-ID: <CAMWiMy-GFqHnfK=OB2SwKWF=tXNXT75K-8=ZvfMW7vE+XoGrTA@mail.gmail.com>

OpenVPN, IPSec +L2TP and PPTP comes to mind here for VPN solutions

On Sat, Aug 13, 2011 at 11:07 AM, Steven Ayre <steveayre at gmail.com> wrote:
> It's a plaintext protocol so nothing you can do within FS to secure it. Firewall the ports to restrict external access and use a VPN if you need to encrypt the data.
>
> Steve on iPhone
>
> On 13 Aug 2011, at 14:41, Chrisbware <chrisbware at interfree.it> wrote:
>
>> Just to have your opinion on it: I've a PHP site on a server, connected
>> to Freeswitch (mod_event_socket) on another server.
>> Which method would you recommend to protect data trasmitted between two
>> servers (they travel on internet)?
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From gmaruzz at gmail.com  Sun Aug 14 01:16:39 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sat, 13 Aug 2011 23:16:39 +0200
Subject: [Freeswitch-users] any IVR example in C/C++?
In-Reply-To: <CAA4nhyAnC0HfxJJPGpOdiBGaG__gBf6Bp9KVi6spjDQgjLTZ5w@mail.gmail.com>
References: <4E43EE36.60209@hw.ac.uk>
	<CAA4nhyAnC0HfxJJPGpOdiBGaG__gBf6Bp9KVi6spjDQgjLTZ5w@mail.gmail.com>
Message-ID: <CALXCt0paCyLs3ojFVeWk9uO7VuuuEgXd_owR_10o0p5L-WStag@mail.gmail.com>

Also, you can check mod-voicemail.c and the new mod-voicemail made by
Moc, those are the only IVR written in C that I know about (they're
written in C because voicemail is considered a base feature, and been
written in C assure stability because people does not fiddle with
them)

On 8/13/11, Moises Silva <moises.silva at gmail.com> wrote:
> On Thu, Aug 11, 2011 at 10:59 AM, xl127 <x.liu at hw.ac.uk> wrote:
>> I am wondering how I could do this for a C/C++ application?
>> And in the scripts languages I can set a callback method, e.g.
>> ? session.setInputCallback(myInputCallback)
>> but I didn't find how to do this in C/C++.
>
> The default question here is, why do you need C/C++ for an IVR?
> FreeSWITCH allows you to use simpler/safer languages to build IVR's.
>
> You can certainly do it, but the reason you don't find examples is
> probably because most people understand there is no need for C/C++
> there. Having said that, you can take a look at the IVR/say/play API's
> in switch_ivr_play_say.c to find out how to provide a callback to the
> different API's thru the switch_input_args_t structure.
>
> Moises Silva
> Senior Software Engineer, Software Development Manager
> Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
> L3R 9R6 Canada
> t. 1 905 474 1990 x128 | e. moy at sangoma.com
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From nbhatti at gmail.com  Sat Aug 13 21:43:11 2011
From: nbhatti at gmail.com (Muhammad Naseer Bhatti)
Date: Sat, 13 Aug 2011 20:43:11 +0300
Subject: [Freeswitch-users] Open Source Billing for FreeSWITCH to get public
	soon ..
Message-ID: <CACak_yx0UBoxVXiCBkWuRLJ0463QR+NRMMaiR_zhEb8n=_Fb9A@mail.gmail.com>

Hello,
I thought this would be the right time to let everyone know we are
going to release open source billing for FreeSWITCH. Completely
developed in PHP/MySQL.

Some of the major features includes:

    Both Pre-Paid and Post-Paid model
    Multiple administration access levels
    Multiple reseller level
    Easy rate/price management
    Route management
    Separate user interface to view their CDR(s) and billing information
    Authentication by IP/ANI and SIP registration
    Codec management for both user and switch
    CDR statistics
    Gateway statistics
    Admin/Reseller/User management
    Switch management from 1 GUI
    Balance and payment information

... and much more.

We need volunteers for testing. Please drop an email --[ vbilling [at]
digitallinx.com ]-- if anyone is interested.

Thanks.


From covici at ccs.covici.com  Sun Aug 14 03:58:45 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Sat, 13 Aug 2011 19:58:45 -0400
Subject: [Freeswitch-users] any IVR example in C/C++?
In-Reply-To: <CALXCt0paCyLs3ojFVeWk9uO7VuuuEgXd_owR_10o0p5L-WStag@mail.gmail.com>
References: <4E43EE36.60209@hw.ac.uk>
	<CAA4nhyAnC0HfxJJPGpOdiBGaG__gBf6Bp9KVi6spjDQgjLTZ5w@mail.gmail.com>
	<CALXCt0paCyLs3ojFVeWk9uO7VuuuEgXd_owR_10o0p5L-WStag@mail.gmail.com>
Message-ID: <31462.1313279925@ccs.covici.com>

Where can I find Mock's voicemail -- I don't see it in contrib?

Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:

> Also, you can check mod-voicemail.c and the new mod-voicemail made by
> Moc, those are the only IVR written in C that I know about (they're
> written in C because voicemail is considered a base feature, and been
> written in C assure stability because people does not fiddle with
> them)
> 
> On 8/13/11, Moises Silva <moises.silva at gmail.com> wrote:
> > On Thu, Aug 11, 2011 at 10:59 AM, xl127 <x.liu at hw.ac.uk> wrote:
> >> I am wondering how I could do this for a C/C++ application?
> >> And in the scripts languages I can set a callback method, e.g.
> >> ? session.setInputCallback(myInputCallback)
> >> but I didn't find how to do this in C/C++.
> >
> > The default question here is, why do you need C/C++ for an IVR?
> > FreeSWITCH allows you to use simpler/safer languages to build IVR's.
> >
> > You can certainly do it, but the reason you don't find examples is
> > probably because most people understand there is no need for C/C++
> > there. Having said that, you can take a look at the IVR/say/play API's
> > in switch_ivr_play_say.c to find out how to provide a callback to the
> > different API's thru the switch_input_args_t structure.
> >
> > Moises Silva
> > Senior Software Engineer, Software Development Manager
> > Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
> > L3R 9R6 Canada
> > t. 1 905 474 1990 x128 | e. moy at sangoma.com
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> -- 
> Sent from my mobile device
> 
> Sincerely,
> 
> Giovanni Maruzzelli
> Cell : +39-347-2665618
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From casteven at gmail.com  Sun Aug 14 07:04:39 2011
From: casteven at gmail.com (Campbell Steven)
Date: Sun, 14 Aug 2011 15:04:39 +1200
Subject: [Freeswitch-users] any IVR example in C/C++?
In-Reply-To: <31462.1313279925@ccs.covici.com>
References: <4E43EE36.60209@hw.ac.uk>
	<CAA4nhyAnC0HfxJJPGpOdiBGaG__gBf6Bp9KVi6spjDQgjLTZ5w@mail.gmail.com>
	<CALXCt0paCyLs3ojFVeWk9uO7VuuuEgXd_owR_10o0p5L-WStag@mail.gmail.com>
	<31462.1313279925@ccs.covici.com>
Message-ID: <CAPerZE93NU+ccKdmGEHufb11rRv0S+L1m0=riKibtgUX2VLU1A@mail.gmail.com>

Try here:

http://fisheye.freeswitch.org/browse/freeswitch.git/src/mod/applications/mod_protovm

Campbell

On Sun, Aug 14, 2011 at 11:58 AM,  <covici at ccs.covici.com> wrote:
> Where can I find Mock's voicemail -- I don't see it in contrib?
>
> Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
>
>> Also, you can check mod-voicemail.c and the new mod-voicemail made by
>> Moc, those are the only IVR written in C that I know about (they're
>> written in C because voicemail is considered a base feature, and been
>> written in C assure stability because people does not fiddle with
>> them)
>>
>> On 8/13/11, Moises Silva <moises.silva at gmail.com> wrote:
>> > On Thu, Aug 11, 2011 at 10:59 AM, xl127 <x.liu at hw.ac.uk> wrote:
>> >> I am wondering how I could do this for a C/C++ application?
>> >> And in the scripts languages I can set a callback method, e.g.
>> >> ? session.setInputCallback(myInputCallback)
>> >> but I didn't find how to do this in C/C++.
>> >
>> > The default question here is, why do you need C/C++ for an IVR?
>> > FreeSWITCH allows you to use simpler/safer languages to build IVR's.
>> >
>> > You can certainly do it, but the reason you don't find examples is
>> > probably because most people understand there is no need for C/C++
>> > there. Having said that, you can take a look at the IVR/say/play API's
>> > in switch_ivr_play_say.c to find out how to provide a callback to the
>> > different API's thru the switch_input_args_t structure.
>> >
>> > Moises Silva
>> > Senior Software Engineer, Software Development Manager
>> > Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
>> > L3R 9R6 Canada
>> > t. 1 905 474 1990 x128 | e. moy at sangoma.com
>> >
>> > _______________________________________________
>> > Join us at ClueCon 2011, Aug 9-11, Chicago
>> > http://www.cluecon.com 877-7-4ACLUE
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> --
>> Sent from my mobile device
>>
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> --
> Your life is like a penny. ?You're going to lose it. ?The question is:
> How do
> you spend it?
>
> ? ? ? ? John Covici
> ? ? ? ? covici at ccs.covici.com
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From mashudi72 at gmail.com  Sun Aug 14 07:22:57 2011
From: mashudi72 at gmail.com (mashudi72 -)
Date: Sun, 14 Aug 2011 10:22:57 +0700
Subject: [Freeswitch-users] Open Source Billing for FreeSWITCH to get
 public soon ..
In-Reply-To: <CACak_yx0UBoxVXiCBkWuRLJ0463QR+NRMMaiR_zhEb8n=_Fb9A@mail.gmail.com>
References: <CACak_yx0UBoxVXiCBkWuRLJ0463QR+NRMMaiR_zhEb8n=_Fb9A@mail.gmail.com>
Message-ID: <CAGpNWUnNNAQ5SwMdLvgkc5-71qkGEpwePA0LCv44OC1KTBn5eA@mail.gmail.com>

Dear Muhammad Naseer Bhatti,

Could you involved me as voluntier to test your billing application? do you
have the documentation for installation and operation that you could share?
thank you,

Mashudi
2011/8/14 Muhammad Naseer Bhatti <nbhatti at gmail.com>

> Hello,
> I thought this would be the right time to let everyone know we are
> going to release open source billing for FreeSWITCH. Completely
> developed in PHP/MySQL.
>
> Some of the major features includes:
>
>    Both Pre-Paid and Post-Paid model
>    Multiple administration access levels
>    Multiple reseller level
>    Easy rate/price management
>    Route management
>    Separate user interface to view their CDR(s) and billing information
>    Authentication by IP/ANI and SIP registration
>    Codec management for both user and switch
>    CDR statistics
>    Gateway statistics
>    Admin/Reseller/User management
>    Switch management from 1 GUI
>    Balance and payment information
>
> ... and much more.
>
> We need volunteers for testing. Please drop an email --[ vbilling [at]
> digitallinx.com ]-- if anyone is interested.
>
> Thanks.
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110814/91056c6d/attachment.html 

From brian at freeswitch.org  Sun Aug 14 07:25:42 2011
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Aug 2011 22:25:42 -0500
Subject: [Freeswitch-users] Playback .wma/.wmv
In-Reply-To: <CAC4JBECvNjZqC66HC9MqE9HVghHkTOn5xFSt_RLZ6gBk+TCOuw@mail.gmail.com>
References: <CAC4JBEDuBqK1KfsWG4rY1dd8ASk9zGzAgP76YqxbpJX9Dg-YmA@mail.gmail.com>
	<1313081473061-6677105.post@n2.nabble.com>
	<CAC4JBECvNjZqC66HC9MqE9HVghHkTOn5xFSt_RLZ6gBk+TCOuw@mail.gmail.com>
Message-ID: <ADE331F6-F99A-4FEF-B85C-821CDFDA585A@freeswitch.org>

Use something that is license compatible which I think gstreamer is but I am not aware that gstreamer comes with the codec for WMA/WMV.  ffmpeg comes with the codec but personally I would extract the wma encoder and decoder and form a standalone lib if possible to use for mod_wma or something.   Just my take.

/b

On Aug 11, 2011, at 12:15 PM, Duvid Rottenberg wrote:

> Thanks for the response. I guess I will have to write that myself (need to
> brush up my c programming skills), I am thinking of doing a mod using
> gstreamer. Does anyone have any recommendations?
> 
> On Thu, Aug 11, 2011 at 12:51 PM, Jeff Lenk <jeff at jefflenk.com> wrote:
> 
>> Not at this time. AFAIK
>> 
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/Playback-wma-wmv-tp6674880p6677105.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>> 
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Sun Aug 14 07:26:30 2011
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Aug 2011 22:26:30 -0500
Subject: [Freeswitch-users] event
In-Reply-To: <CAE2ghw4eOkeZ---P56x7H5yThb2g1WPHSsG5raQaTJZn3jqd4w@mail.gmail.com>
References: <CAE2ghw4GeQp-seD-c9nxa=W-wnkFbEQW6Ji_2sNqGBuaj8n+ug@mail.gmail.com>
	<CAFiqYumEKWBJd5_WR1A9cx72Rw5Digr6EVx7Q96bsBxRT6mJoA@mail.gmail.com>
	<CAE2ghw58FQVnvqQckzJzym6OuKFKwHefhVeAiqLB0BdoALaPeg@mail.gmail.com>
	<CAE2ghw4eOkeZ---P56x7H5yThb2g1WPHSsG5raQaTJZn3jqd4w@mail.gmail.com>
Message-ID: <2E125D40-E6B0-4E27-B9CF-5661CCB5A66A@freeswitch.org>

You need to restart to make this param be active. And this is a global param in the sofia.conf.xml not profile level.

/b

On Aug 12, 2011, at 1:37 AM, Sam wrote:

> The server has 3 static ip and 3 FS listening on different ips,
> i have added <param name="auto-restart" value="false"/>
> 
> But still i get the below , and i want to stop it, how should i do it.
> 
> freeswitch at internal> 2011-08-12 11:48:08.265986 [INFO] mod_sofia.c:4919
> EVENT_TRAP: IP change detected
> 2011-08-12 11:48:08.265986 [INFO] mod_sofia.c:4920 IP change detected
> [192.168.53.189]->[192.168.53.188] []->[]
> 2011-08-12 11:48:08.405851 [NOTICE] sofia_glue.c:5192 Reload XML [Success]
> 2011-08-12 11:48:08.405851 [INFO] mod_enum.c:775 ENUM Reloaded
> 2011-08-12 11:48:08.405851 [INFO] switch_time.c:1028 Timezone reloaded 530
> definitions
> 2011-08-12 11:48:08.925969 [DEBUG] sofia.c:1946 Write lock internal
> 2011-08-12 11:48:09.125988 [DEBUG] sofia.c:1946 Write lock external
> 2011-08-12 11:48:23.925913 [NOTICE] sofia.c:1953 Waiting for worker thread
> 
> 
> Regards
> Sam
> 
> 
> 
> On Mon, Aug 8, 2011 at 6:45 PM, Sam <u2nsam at gmail.com> wrote:
> 
>> Hi,
>> 
>> The server is having 3 static ips on the ethernet interfaces and the FS is
>> listening on single ip .
>> 
>> I will try <param name="auto-restart" value="false"/> and check if it
>> works.
>> 
>> Regards
>> Sam
>> 
>> On Mon, Aug 8, 2011 at 3:47 PM, Steven Ayre <steveayre at gmail.com> wrote:
>> 
>>> Well it looks like your machine changed its IP from .188 to .189. Did that
>>> actually happen, perhaps if you're using DHCP? If it really did happen, then
>>> FS needs to know so it can rebind on the new IP.
>>> 
>>> If it didn't really happen (e.g. if your server is listening on both IPs)
>>> look at the <param name="auto-restart" value="false"/> settting on the SIP
>>> profile. See:
>>> 
>>> http://wiki.freeswitch.org/wiki/Sofia#Forcing_SIP_profile_to_use_a_static_IP_address
>>> 
>>> That parameter will make mod_sofia ignore the notification it receives
>>> from the OS of a network address change.
>>> 
>>> -Steve
>>> 
>>> 
>>> 
>>>  On 8 August 2011 07:13, Sam <u2nsam at gmail.com> wrote:
>>> 
>>>>  Hello,
>>>> 
>>>> What makes the below event to occour and how to stop it reccouring.
>>>> 
>>>> 2011-08-08 11:38:11.415945 [INFO] mod_sofia.c:4919 EVENT_TRAP: IP change
>>>> detected
>>>> 2011-08-08 11:38:11.415945 [INFO] mod_sofia.c:4920 IP change detected
>>>> [192.168.53.188]->[192.168.53.189] []->[]
>>>> 2011-08-08 11:38:11.615887 [NOTICE] sofia_glue.c:5192 Reload XML
>>>> [Success]
>>>> 2011-08-08 11:38:11.615887 [INFO] mod_enum.c:775 ENUM Reloaded
>>>> 2011-08-08 11:38:11.615887 [INFO] switch_time.c:1028 Timezone reloaded
>>>> 530 definitions
>>>> 
>>>> Regards
>>>> Sam
>>>> 
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>> 
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>> 
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>>> 
>> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Sun Aug 14 07:37:50 2011
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Aug 2011 22:37:50 -0500
Subject: [Freeswitch-users] 400 Bad Record-Route Header
In-Reply-To: <4E41858C.5050104@puzzled.xs4all.nl>
References: <CAA5vsdMjDTu_04r4TXH2bVWG1Te2Za7dQz1Ns4JgUO-L0xK-8w@mail.gmail.com>
	<4E41858C.5050104@puzzled.xs4all.nl>
Message-ID: <ACB5D0F5-1D7F-4DD7-B230-96EB771963DD@freeswitch.org>

No its not 1.0.6 its just a bad Record-Route header! :P

/b

On Aug 9, 2011, at 2:07 PM, Patrick Lists wrote:

> I'd say version 1.0.6 is the problem. It's ancient and compared to 
> recent releases a Pretty Bad Idea. Please get latest git and try again. 
> If there still is a problem with latest git then file a bug.
> 
> Regards,
> Patrick

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110813/e8f55aba/attachment.html 

From brian at freeswitch.org  Sun Aug 14 07:39:23 2011
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Aug 2011 22:39:23 -0500
Subject: [Freeswitch-users] SIP proxy collect DTMF using FS
In-Reply-To: <CAJUJwtjqMyvVX1eteYBWFf41Mz+fZjnRZA_0BR2DZs0R5zh_tA@mail.gmail.com>
References: <CAJUJwtjqMyvVX1eteYBWFf41Mz+fZjnRZA_0BR2DZs0R5zh_tA@mail.gmail.com>
Message-ID: <0D92C5B0-CD84-47B1-A17C-A2B083B760E2@freeswitch.org>

What are you trying to do exactly because it sounds like you have selected the most painful way to accomplish it?

/b

On Aug 9, 2011, at 2:11 AM, Sam Govind wrote:

> Hi guys,
> 
> I'm looking to establish a scenario like this, any idea how to do it, if its
> possible.
> 
> 1- SIP proxy send call to FS where DTMF will be collected. (I'm thinking of
> using PlayAndGetDigits)
> 2- DTMF collected be sent back to SIP proxy while FS ends the call
> 3- Call at SIP proxy end keeps running for some other processing.
> 
> basically I just need FS to collect DTMF and send those back to SIP Proxy.
> 
> Any ideas are welcome.
> 
> Thanks.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110813/98fee992/attachment.html 

From brian at freeswitch.org  Sun Aug 14 07:40:24 2011
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Aug 2011 22:40:24 -0500
Subject: [Freeswitch-users] incompatible destination because of crypto
	on incoming calls
In-Reply-To: <69CD4836D15D45B683FFFBAB557B5060@omni1.local>
References: <69CD4836D15D45B683FFFBAB557B5060@omni1.local>
Message-ID: <C4C8EBF3-F2FB-468E-8ABF-9A2F3B34705A@freeswitch.org>

And you're sending crypto keys in the RTP/AVP instead of the RTP/SAVP as per the RFC?  

/b

On Aug 6, 2011, at 2:38 AM, Anestis Mavro wrote:

> Hi,
> 
> 
> 
> Suddenly (latest git) something happened with the incoming calls.
> 
> All calls have crypto enabled and they get dropped with "INCOMPATIBLE
> DESTINATION"
> 
> I have TLS disabled in vars.xml but still I see in debug:
> 
> 
> 
> Sofia_glue:2900 Set Local Key [1 AES_CM_128_HMAC_SHA1_32 inline: ...]
> 
> 
> 
> Show channels and show calls show now a lot of calls, but they don't exist;
> they don't get cleared.
> 
> 
> 
> Anybody any idea where this can come from?
> 
> 
> 
> Thank you
> 
> 
> 
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Sun Aug 14 07:41:57 2011
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Aug 2011 22:41:57 -0500
Subject: [Freeswitch-users] ACL: inside out
In-Reply-To: <1312472318582-6653218.post@n2.nabble.com>
References: <1312472318582-6653218.post@n2.nabble.com>
Message-ID: <DB7BB104-74A9-46D9-8368-D668F976B406@freeswitch.org>

 If you have all public IP's then you don't have to do anything.  Are you saying its registering on the External profile?  

/b

On Aug 4, 2011, at 10:38 AM, micha wrote:

> Hello!
> 
> How do I define what is inside and what is outside my network? My FreeSWITCH
> server and my devices that reside on my internal network have public IP
> addresses but in separate subnets.
> 
> I tried the following in acl.conf.xml:
> 
>   <list name="lan" default="deny">
>      <node type="allow" cidr="XXX.XXX.XXX.0/22"/>
>    </list>
> 	
> But my internal clients are still listed as external registrations.
> 
> 
> Where is my error?
> 
> Cheers,
> 
> Micha

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110813/879dae18/attachment.html 

From brian at freeswitch.org  Sun Aug 14 07:42:44 2011
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Aug 2011 22:42:44 -0500
Subject: [Freeswitch-users] freeswitch 1.0.7 required
In-Reply-To: <CAMWiMy_61z8UhxL6wG+rzVQ9jdhhKdH9m9zUHtUKdQ_bghCG-w@mail.gmail.com>
References: <CA+P_fE_UgehuXmdtj8DOtoHYToRrUQg7tjpkCswe0-SgPbD10Q@mail.gmail.com>
	<CA5F9A6E.1A0F0%krice@freeswitch.org>
	<CAMWiMy_61z8UhxL6wG+rzVQ9jdhhKdH9m9zUHtUKdQ_bghCG-w@mail.gmail.com>
Message-ID: <C48C77C9-B6B3-43D7-8282-EB3B04656C0F@freeswitch.org>

Wasn't really a joke per se... just  an RC... I did the same for 1.0.5 which became 1.0.6 :P

/b

On Aug 4, 2011, at 11:20 AM, curriegrad2004 wrote:

> 1.0.7 was more like a joke release. Of course us seasoned FS users
> know better ;)
> 
> On Wed, Aug 3, 2011 at 10:38 PM, Ken Rice <krice at freeswitch.org> wrote:
>> 1.0.X , X is a snapshot of git at some point in time... Its a release
>> version... The problem is people don?t test when the main developers say
>> they are getting ready for a new release, so they test as best as they can
>> an cut a release...
>> 
>> 20 minutes later, a dozen bugs are reported...  Because people expect
>> everyone else to test... The way to fix this was decided that there will be
>> OLD version (and by old I mean really old) and there will be head for the
>> most part... That forces people to test... And 99.99% of the time if you do
>> fine something broken ?make current? fixes the issue...
>> 
>> 
>> And yes 1.0.head is what comes from git... Notice that git-HEXNUMBER...
>> That?s the actual give hash version...
>> Have a nice day
>> 
>> K
>> 
>> 
>> On 8/4/11 12:33 AM, "Dhairya Vora" <dhairya.blogs at gmail.com> wrote:
>> 
>> Here (http://wiki.freeswitch.org/wiki/Download_FreeSWITCH) they say that in
>> Git, 1.0.7 is not available.
>> 
>> (FYI: here
>> (http://lists.freeswitch.org/pipermail/freeswitch-users/2011-January/067446.html)
>> they say that "git does not give odd number version." Really??)
>> 
>> By the way, I installed using git and I thought that it is 1.0.6. Now it
>> shows 1.0.head. just see the output.
>> ****************************************************************************************************
>> freeswitch at localhost.localdomain> version
>> 
>> FreeSWITCH Version 1.0.head (git-4b1bb61 2011-08-01 15-43-07 -0500)
>> ****************************************************************************************************
>> 
>> 
>> 
>> On Thu, Aug 4, 2011 at 10:51 AM, Ken Rice <krice at freeswitch.org> wrote:
>> 
>> Just get git head... That?s where you want to be anyway... Yes its stable...
>> But as with any new deployment you should test it to make sure it meets your
>> needs
>> 
>> 
>> 
>> 
>> On 8/4/11 12:18 AM, "Dhairya Vora" <dhairya.blogs at gmail.com
>> <http://dhairya.blogs at gmail.com> > wrote:
>> 
>> I am installing freeswitch. I read that latest version is 1.0.7 but in git
>> it is not available. They say that it is at http://latest.freeswitch.org/
>> but I am unable to open that url. Either server is down, or url is changed.
>> Any other location to download freeswitch 1.0.7 (or the latest version) ?
>> 
>> ________________________________
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <http://FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> 
>> ________________________________
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
> 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110813/b9b7371d/attachment.html 

From curriegrad2004 at gmail.com  Sun Aug 14 07:42:44 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sat, 13 Aug 2011 20:42:44 -0700
Subject: [Freeswitch-users] Open Source Billing for FreeSWITCH to get
 public soon ..
In-Reply-To: <CAGpNWUnNNAQ5SwMdLvgkc5-71qkGEpwePA0LCv44OC1KTBn5eA@mail.gmail.com>
References: <CACak_yx0UBoxVXiCBkWuRLJ0463QR+NRMMaiR_zhEb8n=_Fb9A@mail.gmail.com>
	<CAGpNWUnNNAQ5SwMdLvgkc5-71qkGEpwePA0LCv44OC1KTBn5eA@mail.gmail.com>
Message-ID: <CAMWiMy8=9C9mxdtPLkvoC=Kw+wWrV72N=872JVkz6TEpnyzRgA@mail.gmail.com>

It would also be a great idea if you can obtain commit access to our
contrib repo so you can have your code hosted there ;)

On Sat, Aug 13, 2011 at 8:22 PM, mashudi72 - <mashudi72 at gmail.com> wrote:
>
> Dear Muhammad Naseer Bhatti,
>
> Could you involved me as voluntier to test your billing application? do you
> have the documentation for installation and operation that you could share?
> thank you,
>
> Mashudi
> 2011/8/14 Muhammad Naseer Bhatti <nbhatti at gmail.com>
>>
>> Hello,
>> I thought this would be the right time to let everyone know we are
>> going to release open source billing for FreeSWITCH. Completely
>> developed in PHP/MySQL.
>>
>> Some of the major features includes:
>>
>> ? ?Both Pre-Paid and Post-Paid model
>> ? ?Multiple administration access levels
>> ? ?Multiple reseller level
>> ? ?Easy rate/price management
>> ? ?Route management
>> ? ?Separate user interface to view their CDR(s) and billing information
>> ? ?Authentication by IP/ANI and SIP registration
>> ? ?Codec management for both user and switch
>> ? ?CDR statistics
>> ? ?Gateway statistics
>> ? ?Admin/Reseller/User management
>> ? ?Switch management from 1 GUI
>> ? ?Balance and payment information
>>
>> ... and much more.
>>
>> We need volunteers for testing. Please drop an email --[ vbilling [at]
>> digitallinx.com ]-- if anyone is interested.
>>
>> Thanks.
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From brian at freeswitch.org  Sun Aug 14 07:45:10 2011
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Aug 2011 22:45:10 -0500
Subject: [Freeswitch-users] FS to a Sonus SIP trunk
In-Reply-To: <4E31369A.70605@hcu-hamburg.de>
References: <4E2ED4AB.1030305@hcu-hamburg.de> <4E31369A.70605@hcu-hamburg.de>
Message-ID: <FA0AA9C2-B07B-43B7-9D83-21816E695DC3@freeswitch.org>

Have you thought of opening a Jira?

/b

On Jul 28, 2011, at 5:14 AM, michael knop wrote:

> Update:
> 
> Call starts with good sound quality. After the following log message 
> sound is choppy:
> 
> 2011-07-28 12:07:22.639151 [DEBUG] sofia.c:5094 Duplicate SDP
> v=0
> o=Sonus_UAC 8739 8900 IN IP4 XXX.XXX.XXX.XXX
> s=SIP Media Capabilities
> c=IN IP4 YYY.YYY.YYY.YYY
> t=0 0
> m=audio 20320 RTP/AVP 8 0 18 100
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:100 telephone-event/8000
> a=fmtp:100 0-15
> a=maxptime:10
> 
> /micha
> 
> Am 26.07.2011 16:52, schrieb michael knop:
>> Hi all!
>> 
>> I?m trying to connect my FS to a Sonus SIP trunk. I followed the
>> instruction at
>> 
>>    http://wiki.freeswitch.org/wiki/RTP_Issues#Sonus
>> 
>> but it did not work. At the beginning of a call voice quality is good.
>> After a while it changes to choppy.
>> 
>> I don?t know if it?s the same problem: When I call the Tetris extension
>> via Sonus SIP trunk the sound is too fast and I?m getting log entries
>> like the following one:
>> 
>> [...]
>> 2011-07-26 11:52:27.682259 [WARNING] mod_sofia.c:1106 Asynchronous PTIME
>> not supported, changing our end from 20 to 10
>> 2011-07-26 11:52:27.682259 [DEBUG] sofia_glue.c:2737 Changing Codec from
>> PCMA at 20ms@8000hz to PCMA at 10ms@8000hz
>> 2011-07-26 11:52:27.722150 [WARNING] switch_time.c:516 Increasing global
>> timer resolution to 10ms to handle interval 10
>> 2011-07-26 11:52:27.722150 [DEBUG] switch_rtp.c:1521 RE-Starting timer
>> [soft] 80 bytes per 10ms
>> 2011-07-26 11:52:27.722150 [DEBUG] sofia_glue.c:2819 Set Codec
>> sofia/external/+4940... at 193...:5060 PCMA/8000 10 ms 80 samples 64000 bits
>> 2011-07-26 11:52:27.722150 [DEBUG] switch_core_io.c:1074 Engaging Write
>> Buffer at 160 bytes to accommodate 320->160
>> [...]
>> 
>> This problem is fixed by adding the following line to
>> conf/sip_profiles/external.xml:
>> 
>> <param name="rtp-autofix-timing" value="false"/>
>> 
>> Any hints?
>> 
>> /micha
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Sun Aug 14 07:46:26 2011
From: brian at freeswitch.org (Brian West)
Date: Sat, 13 Aug 2011 22:46:26 -0500
Subject: [Freeswitch-users] Discovered problem with PocketSphinx build?
In-Reply-To: <4097DFE6-DA8E-47BC-A8DC-90AF44E0C575@bryansmart.com>
References: <4097DFE6-DA8E-47BC-A8DC-90AF44E0C575@bryansmart.com>
Message-ID: <0AE6DD0E-B9D6-4937-98C9-572260A06830@freeswitch.org>

You will have to install the wsj1 model yourself because they don't ship that anymore.

/b

On Jul 27, 2011, at 5:46 PM, Bryan Smart wrote:

> 
> 2011-07-27 21:43:14.651803 [WARNING] mod_pocketsphinx.c:147 Can't open speech model /usr/local/freeswitch/grammar/model/wsj1.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110813/6ac25abd/attachment.html 

From covici at ccs.covici.com  Sun Aug 14 08:55:00 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Sun, 14 Aug 2011 00:55:00 -0400
Subject: [Freeswitch-users] any IVR example in C/C++?
In-Reply-To: <CAPerZE93NU+ccKdmGEHufb11rRv0S+L1m0=riKibtgUX2VLU1A@mail.gmail.com>
References: <4E43EE36.60209@hw.ac.uk>
	<CAA4nhyAnC0HfxJJPGpOdiBGaG__gBf6Bp9KVi6spjDQgjLTZ5w@mail.gmail.com>
	<CALXCt0paCyLs3ojFVeWk9uO7VuuuEgXd_owR_10o0p5L-WStag@mail.gmail.com>
	<31462.1313279925@ccs.covici.com>
	<CAPerZE93NU+ccKdmGEHufb11rRv0S+L1m0=riKibtgUX2VLU1A@mail.gmail.com>
Message-ID: <5416.1313297700@ccs.covici.com>

And how would I do a git for that module?

Campbell Steven <casteven at gmail.com> wrote:

> Try here:
> 
> http://fisheye.freeswitch.org/browse/freeswitch.git/src/mod/applications/mod_protovm
> 
> Campbell
> 
> On Sun, Aug 14, 2011 at 11:58 AM,  <covici at ccs.covici.com> wrote:
> > Where can I find Mock's voicemail -- I don't see it in contrib?
> >
> > Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
> >
> >> Also, you can check mod-voicemail.c and the new mod-voicemail made by
> >> Moc, those are the only IVR written in C that I know about (they're
> >> written in C because voicemail is considered a base feature, and been
> >> written in C assure stability because people does not fiddle with
> >> them)
> >>
> >> On 8/13/11, Moises Silva <moises.silva at gmail.com> wrote:
> >> > On Thu, Aug 11, 2011 at 10:59 AM, xl127 <x.liu at hw.ac.uk> wrote:
> >> >> I am wondering how I could do this for a C/C++ application?
> >> >> And in the scripts languages I can set a callback method, e.g.
> >> >> ? session.setInputCallback(myInputCallback)
> >> >> but I didn't find how to do this in C/C++.
> >> >
> >> > The default question here is, why do you need C/C++ for an IVR?
> >> > FreeSWITCH allows you to use simpler/safer languages to build IVR's.
> >> >
> >> > You can certainly do it, but the reason you don't find examples is
> >> > probably because most people understand there is no need for C/C++
> >> > there. Having said that, you can take a look at the IVR/say/play API's
> >> > in switch_ivr_play_say.c to find out how to provide a callback to the
> >> > different API's thru the switch_input_args_t structure.
> >> >
> >> > Moises Silva
> >> > Senior Software Engineer, Software Development Manager
> >> > Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
> >> > L3R 9R6 Canada
> >> > t. 1 905 474 1990 x128 | e. moy at sangoma.com
> >> >
> >> > _______________________________________________
> >> > Join us at ClueCon 2011, Aug 9-11, Chicago
> >> > http://www.cluecon.com 877-7-4ACLUE
> >> >
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >>
> >> --
> >> Sent from my mobile device
> >>
> >> Sincerely,
> >>
> >> Giovanni Maruzzelli
> >> Cell : +39-347-2665618
> >>
> >> _______________________________________________
> >> Join us at ClueCon 2011, Aug 9-11, Chicago
> >> http://www.cluecon.com 877-7-4ACLUE
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > --
> > Your life is like a penny. ?You're going to lose it. ?The question is:
> > How do
> > you spend it?
> >
> > ? ? ? ? John Covici
> > ? ? ? ? covici at ccs.covici.com
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From mattdfong at gmail.com  Sun Aug 14 10:10:32 2011
From: mattdfong at gmail.com (Matthew Fong)
Date: Sat, 13 Aug 2011 23:10:32 -0700
Subject: [Freeswitch-users] mod_rtmp & flex client
Message-ID: <CAA5vsdNFhNdXZv10+Cd6CKCQWsgTP6Eg9NsVwaRcvzsDxydioQ@mail.gmail.com>

I am trying to get the mod_rtmp to work with the flex client, but I am
receiving the following errors when I try to make a test call

2011-08-14 06:04:32.744112 [NOTICE] rtmp_sig.c:121 Sent connect reply
2011-08-14 06:04:38.764112 [INFO] rtmp_sig.c:136 Replied to createStream (1)
2011-08-14 06:04:38.764112 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled
control packet (type=0x3)
2011-08-14 06:04:38.764112 [INFO] rtmp_sig.c:136 Replied to createStream (2)
2011-08-14 06:04:38.764112 [NOTICE] switch_channel.c:904 New Channel
rtmp/default/5000 [18578182-a702-49bc-9590-c53ec2d16b72]
2011-08-14 06:04:38.764112 [ERR] rtmp_sig.c:305 Couldn't create call.
2011-08-14 06:04:38.764112 [INFO] mod_dialplan_xml.c:336 Processing
 <0000000000>->5000 in context public
2011-08-14 06:04:38.764112 [NOTICE] switch_core_state_machine.c:194
rtmp/default/5000 has executed the last dialplan instruction, hanging up.
2011-08-14 06:04:38.764112 [NOTICE] switch_core_state_machine.c:196 Hangup
rtmp/default/5000 [CS_EXECUTE] [NORMAL_CLEARING]
2011-08-14 06:04:38.764112 [NOTICE] switch_core_session.c:1346 Session 1
(rtmp/default/5000) Ended
2011-08-14 06:04:38.764112 [NOTICE] switch_core_session.c:1348 Close Channel
rtmp/default/5000 [CS_DESTROY]
2011-08-14 06:04:39.784111 [INFO] rtmp_sig.c:159 Sending audio
2011-08-14 06:04:39.784111 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled
control packet (type=0x3)
2011-08-14 06:04:39.784111 [INFO] rtmp_sig.c:274 Got publish on stream 2.
2011-08-14 06:04:39.784111 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled
control packet (type=0x3)

I am using a version checked-out from yesterday, Ubuntu 10.4 64-bit and
flash debug version. Does anyone know what I am doing wrong? Or has anyone
gotten it to work? Also I tried the FS conference call test using the hosted
version on conference.freeswitch.org and it seems the first few seconds of
audio are distorted. Is there a way to fix this? Thanks... mod_rtmp seems
very promising.

--matt
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110813/ebbe6111/attachment.html 

From devel at omninet.eu  Sun Aug 14 11:43:48 2011
From: devel at omninet.eu (Anestis Mavro)
Date: Sun, 14 Aug 2011 10:43:48 +0300
Subject: [Freeswitch-users] incompatible destination because of crypto
	incoming calls
In-Reply-To: <C4C8EBF3-F2FB-468E-8ABF-9A2F3B34705A@freeswitch.org>
References: <69CD4836D15D45B683FFFBAB557B5060@omni1.local>
	<C4C8EBF3-F2FB-468E-8ABF-9A2F3B34705A@freeswitch.org>
Message-ID: <942B67E68E554F0B9067531E61AE812D@omni1.local>


Yes, this happened after updating to latest git around August 5th or 6th.
I went back to 3e2c662a (Aug 4th) and the problem disappeared.
A few days later I tried two more times to go for the latest git, but I
received the same error. 

I am now with the latest git (94c9cbf6) and the problem has gone.

It seems that something was broken for a few days but is fixed now.

Thank you for your reply.

Anestis


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Sunday, August 14, 2011 6:40 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] incompatible destination because of cryptoon
incoming calls

And you're sending crypto keys in the RTP/AVP instead of the RTP/SAVP as per
the RFC?  

/b

On Aug 6, 2011, at 2:38 AM, Anestis Mavro wrote:

> Hi,
> 
> 
> 
> Suddenly (latest git) something happened with the incoming calls.
> 
> All calls have crypto enabled and they get dropped with "INCOMPATIBLE
> DESTINATION"
> 
> I have TLS disabled in vars.xml but still I see in debug:
> 
> 
> 
> Sofia_glue:2900 Set Local Key [1 AES_CM_128_HMAC_SHA1_32 inline: ...]
> 
> 
> 
> Show channels and show calls show now a lot of calls, but they don't
exist;
> they don't get cleared.
> 
> 
> 
> Anybody any idea where this can come from?
> 
> 
> 
> Thank you
> 
> 
> 
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

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

__________ Information from ESET NOD32 Antivirus, version of virus signature
database 5054 (20100423) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com
 
 

__________ Information from ESET NOD32 Antivirus, version of virus signature
database 5054 (20100423) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com
 



From fieldpeak at gmail.com  Sun Aug 14 14:18:32 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Sun, 14 Aug 2011 18:18:32 +0800
Subject: [Freeswitch-users] Mod_rad_auth issue for FS working with
 FreeRadius server
In-Reply-To: <CAJoKKdXW+vrYYMMM163XFewWyzcHW14MLkyTr53oaVnKP4-9jQ@mail.gmail.com>
References: <CAOTmbMzvd5OHL8z=_8vu6Orp2_prGxaw1gmXYCM2=2+pgzs_WA@mail.gmail.com>
	<CAJoKKdX01Qc8ibs25PUELi8GrzMDNOrHb_bQvYdq3ZqA4XLy4w@mail.gmail.com>
	<CAOTmbMxL3pLOuw655fyT5VzAON_N0PrDN+-i-++=oZoYU6+Fmg@mail.gmail.com>
	<CAOTmbMyvyDA-CRpaMC=WRGVkiyTG=Rs+n4hyeQWdyTZpqysSzw@mail.gmail.com>
	<CAJoKKdXSwF02S9tBkorGwOjj57AtkRpU0c177qXASEpodKVk=w@mail.gmail.com>
	<CAOTmbMzu2qDYh2T24JSpCYn1xH2J4iGiK7UkUB4Kq=-KdcDWGQ@mail.gmail.com>
	<CAJoKKdU66jK3f+mOEV90Y3zrmzpTSdWnwGsWimZxir42khiJ-w@mail.gmail.com>
	<CAOTmbMxW=EDr3-Ayo2R_MsCCTUZ4PnDkkRUuR2FisH9tK5xkWA@mail.gmail.com>
	<CAJoKKdVSkQOV3Kou_ro5AQZTyaqH6r0WRE-hZdrA=ye8ARE_vQ@mail.gmail.com>
	<CAOTmbMyg5_q-pFFtpzOooUcSZ8Gkb38y4sk1pYeJLYqguA=Jbg@mail.gmail.com>
	<CAJoKKdWqjt46Gvbj=4Tz-apQb+EV2UHJTsRBTE_bJZ95ogo2DA@mail.gmail.com>
	<CAOTmbMx8_aDoB2Q9ocuVtuXoQf0hbTJ0=VL+RtaWiZQgAw=YTg@mail.gmail.com>
	<CAJoKKdXW+vrYYMMM163XFewWyzcHW14MLkyTr53oaVnKP4-9jQ@mail.gmail.com>
Message-ID: <CAOTmbMyq3r9q=obmguJs6K_0WPYSFYOiPTPNJdjsgP2HKAHiwA@mail.gmail.com>

Hi Thiomir,

Thanks for your clarification, understood...

Cheers!
Charles

2011/8/11 Tihomir Culjaga <tculjaga at gmail.com>

> hello,
>
> the example down below is just an example. In the real application you will
> be using channel variables instead of direct input.
> anyhow everything depends of what is your application intended for and how
> you would like to behave.
>
> Right now, you cannot authorize registrations as there is no event handler
> built into the module. Its on the roadmap but not gonna happen in next few
> weeks.
>
> what you can do is to authorize calls (INVITEs) by triggering the
> application within the dialplan. Also, FS extensions have their own ANI and
> you can authorize by ANI. If this is not enough, you can try to fetch the
> calling user password from the database and populate a session variable....
> than use this variable to trigger radius authorization.
>
> Anyhow, i think this is quite easy to do ... if you don't manage to do it
> on your own, drop me an e-mail and i can help ya.
>
>
> Cheers,
> Thiomir.
>
>
>
>
> On Tue, Aug 9, 2011 at 2:58 PM, fieldpeak <fieldpeak at gmail.com> wrote:
>
>> Hi Tihomir,
>>
>> As my understanding, when using mod_rad_auth, we have to send both
>> username and password to FreeRadius, like the example in wiki below (marked
>> in yellow), the example is for a fixed password, however in real world, we
>> have to dynamically inject the password as per user on-the-fly, e.g. user
>> 1001 's password is 1234, user 1002's password is 2345 etc. in other word,
>> we have to dynamically get the specific user's password and inject to the
>> dial plan. Can you please advise how we should write the dial plan for the
>> real case? Thanks in avdvance.
>>
>> P.S. What I'm concerning are both REGISTERATON and INVITE...how can we do
>> the auth by Freeradius...
>>
>> <extension name="unitest_rad-ANI-balance">
>>     <condition field="destination_number" expression="^602$">
>>       <action application="log" data="INFO  PRIJE RAD_AUTH "/>
>>
>>       <action inline="true" application="set" data="CALLID=h323-conf-id=${uuid}"/>
>>       <action inline="true" application="set" data="SERVICENUM=h323-prompt-id=${destination_number}"/>
>>       <action inline="true" application="set" data="TRANSACTIONID=h323-ivr-out=transactionID:1234"/>
>>  <!--      <action inline="true" application="set" data="CALLINGNUMBER=${caller_id_number}"/> -->
>>       <action inline="true" application="set" data="CALLINGNUMBER=38516060333"/>
>>       <action inline="true" application="set" data="USERNAME=38516060333"/>
>>  <!--      <action inline="true" application="set" data="USERNAME=209354"/> -->
>>       <action inline="true" application="set" data="PASSWD=003282"/>
>>       <action inline="true" application="set" data="DIALED_NUMBER=16094191500"/>
>>
>>
>>
>> Regards,
>> Charles
>>
>>
>> 2011/8/9 Tihomir Culjaga <tculjaga at gmail.com>
>>
>>> im glad it works :=)
>>>
>>> T.
>>>
>>>
>>> On Mon, Aug 8, 2011 at 8:18 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>>>
>>>> Hi Tihomir,
>>>>
>>>> The issue has been resolved by correcting the client secrect,
>>>> appreciated very much for your kindly help!
>>>>
>>>> Regards,
>>>> Charles
>>>>
>>>> 2011/8/7 Tihomir Culjaga <tculjaga at gmail.com>
>>>>
>>>>> are u sure you are using the correct secret on both client and server ?
>>>>>
>>>>>
>>>>> On Fri, Aug 5, 2011 at 10:12 AM, fieldpeak <fieldpeak at gmail.com>wrote:
>>>>>
>>>>>> Hi Tihomir,
>>>>>>
>>>>>> Thanks for your advise, i've added below to rad_auth.conf.xml (vsas
>>>>>> section), as well as tried auth-type to 0(local) and 1(system), however, the
>>>>>> issue still exist.
>>>>>>
>>>>>>
>>>>>> <param name="NAS-Port-Type" id="61" value="0" pec="0" expr="0"
>>>>>> direction="in"/>
>>>>>> <param name="Login-User" id="6" value="1" pec="0" expr="0"
>>>>>> direction="in"/>
>>>>>> <param name="Auth-Type" id="1000" value="0" pec="0" expr="0"
>>>>>> direction="in"/>
>>>>>>
>>>>>> FreeRadius output:
>>>>>>
>>>>>> Found Auth-Type = PAP
>>>>>> # Executing group from file /usr/local/etc/raddb/sites-enabled/default
>>>>>>
>>>>>> +- entering group PAP {...}
>>>>>> [pap] login attempt with password "Q?????? ??????p???F?+??a"
>>>>>> [pap] Using clear text password "1111"
>>>>>> [pap] Passwords don't match
>>>>>> ++[pap] returns reject
>>>>>> Failed to authenticate the user.
>>>>>>   WARNING: Unprintable characters in the password.        Double-check the shared secret on the server and the NAS!
>>>>>>
>>>>>> Using Post-Auth-Type Reject
>>>>>> # Executing group from file /usr/local/etc/raddb/sites-enabled/default
>>>>>>
>>>>>> +- entering group REJECT {...}
>>>>>> [attr_filter.access_reject]     expand: %{User-Name} -> 1001
>>>>>>  attr_filter: Matched entry DEFAULT at line 11
>>>>>> ++[attr_filter.access_reject] returns updated
>>>>>> Delaying reject of request 38 for 1 seconds
>>>>>>
>>>>>> Regards,
>>>>>> Charles
>>>>>>
>>>>>>
>>>>>> 2011/8/5 Tihomir Culjaga <tculjaga at gmail.com>
>>>>>>
>>>>>>> add to rad_auth.conf.xml
>>>>>>>
>>>>>>> <param name="NAS-Port-Type" id="61" value="0" pec="0" expr="0"
>>>>>>> direction="in"/>
>>>>>>> <param name="Login-User" id="6" value="1" pec="0" expr="0"
>>>>>>> direction="in"/>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> as for Auth Type im not sure if you need it ... this is up to your
>>>>>>> server.
>>>>>>> According to dictionary file you need to set it as follows:
>>>>>>>
>>>>>>> <param name="Auth-Type" id="1000" value="?" pec="0" expr="0"
>>>>>>> direction="in"/>
>>>>>>>
>>>>>>> the value (set as ?) is one of the folowing. Again, not sure what is
>>>>>>> required by your server.
>>>>>>>
>>>>>>> VALUE           Auth-Type               Local                   0
>>>>>>> VALUE           Auth-Type               System                  1
>>>>>>> VALUE           Auth-Type               SecurID                 2
>>>>>>> VALUE           Auth-Type               Crypt-Local             3
>>>>>>> VALUE           Auth-Type               Reject                  4
>>>>>>>
>>>>>>> #
>>>>>>> #       Cistron extensions
>>>>>>> #
>>>>>>> VALUE           Auth-Type               Pam                     253
>>>>>>> VALUE           Auth-Type               Accept                  254
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> regards,
>>>>>>> Tihomir.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Wed, Aug 3, 2011 at 6:32 AM, fieldpeak <fieldpeak at gmail.com>wrote:
>>>>>>>
>>>>>>>> Hi Tihomir,
>>>>>>>>
>>>>>>>> Sorry, i missed your mail in gmail before, just now saw it, and
>>>>>>>> after using your dictionary.all, the dictionary issue was resolved, very
>>>>>>>> appreciated for your kindly help! however, it did not fully functional yet,
>>>>>>>>
>>>>>>>> Attached are configuration files that i used, when i dial 601 to
>>>>>>>> trigger to auth, the freeradius server shows log below, the supecious log is
>>>>>>>> the value User-Password, it should be '1111' that i've set in the mysql db
>>>>>>>> of freeradisu server for the user 1001 .
>>>>>>>>
>>>>>>>> i searched in google, for "known good" password issue, i suggest
>>>>>>>> change user-password to cleartext-password, however, i did not find where it
>>>>>>>> is.
>>>>>>>> and also the Auth-Type, where to configure it...
>>>>>>>>
>>>>>>>> Freeradius server log:
>>>>>>>>
>>>>>>>> rad_recv: Access-Request packet from host 127.0.0.1 port 52684,
>>>>>>>> id=49, length=111
>>>>>>>>         User-Name = "1001"
>>>>>>>>         User-Password = "?\210\365@\263\t\306\343\243iT?\311C\t\002
>>>>>>>> "
>>>>>>>>         Called-Station-Id = "888"
>>>>>>>>         h323-conf-id = "749d2b5a-16ad-48e4-af58-24011949d1b5"
>>>>>>>>         Calling-Station-Id = "1001"
>>>>>>>>         NAS-Port = 0
>>>>>>>>         NAS-IP-Address = 127.0.0.1
>>>>>>>> # Executing section authorize from file
>>>>>>>> /usr/local/etc/raddb/sites-enabled/default
>>>>>>>> +- entering group authorize {...}
>>>>>>>> ++[preprocess] returns ok
>>>>>>>> [auth_log]      expand:
>>>>>>>> /usr/local/var/log/radius/radacct/%{Client-IP-Address}/auth-detail-%Y%m%d ->
>>>>>>>> /usr/local/var/log/radius/radacct/127.0.0.1/auth-detail-20110803
>>>>>>>> [auth_log]
>>>>>>>> /usr/local/var/log/radius/radacct/%{Client-IP-Address}/auth-detail-%Y%m%d
>>>>>>>> expands to /usr/local/var/log/radius/radacct/
>>>>>>>> 127.0.0.1/auth-detail-20110803
>>>>>>>> [auth_log]      expand: %t -> Wed Aug  3 12:06:33 2011
>>>>>>>> ++[auth_log] returns ok
>>>>>>>> ++[chap] returns noop
>>>>>>>> ++[mschap] returns noop
>>>>>>>> ++[digest] returns noop
>>>>>>>> [suffix] No '@' in User-Name = "1001", looking up realm NULL
>>>>>>>> [suffix] No such realm "NULL"
>>>>>>>> ++[suffix] returns noop
>>>>>>>> [eap] No EAP-Message, not doing EAP
>>>>>>>> ++[eap] returns noop
>>>>>>>> ++[unix] returns notfound
>>>>>>>> ++[files] returns noop
>>>>>>>> [sql]   expand: %{User-Name} -> 1001
>>>>>>>> [sql] sql_set_user escaped user --> '1001'
>>>>>>>> rlm_sql (sql): Reserving sql socket id: 4
>>>>>>>> [sql]   expand: SELECT id, username, attribute, value, op
>>>>>>>> FROM radcheck           WHERE username = '%{SQL-User-Name}'           ORDER
>>>>>>>> BY id -> SELECT id, username, attribute, value, op           FROM
>>>>>>>> radcheck           WHERE username = '1001'           ORDER BY id
>>>>>>>> [sql]   expand: SELECT groupname           FROM
>>>>>>>> radusergroup           WHERE username = '%{SQL-User-Name}'           ORDER
>>>>>>>> BY priority -> SELECT groupname           FROM radusergroup           WHERE
>>>>>>>> username = '1001'           ORDER BY priority
>>>>>>>> rlm_sql (sql): Released sql socket id: 4
>>>>>>>> [sql] User 1001 not found
>>>>>>>> ++[sql] returns notfound
>>>>>>>> ++[expiration] returns noop
>>>>>>>> ++[logintime] returns noop
>>>>>>>> [pap] WARNING! No "known good" password found for the user.
>>>>>>>> Authentication may fail because of this.
>>>>>>>> ++[pap] returns noop
>>>>>>>> ERROR: No authenticate method (Auth-Type) found for the request:
>>>>>>>> Rejecting the user
>>>>>>>> Failed to authenticate the user.
>>>>>>>>   WARNING: Unprintable characters in the password.
>>>>>>>> Double-check the shared secret on the server and the NAS!
>>>>>>>> Using Post-Auth-Type Reject
>>>>>>>> # Executing group from file
>>>>>>>> /usr/local/etc/raddb/sites-enabled/default
>>>>>>>> +- entering group REJECT {...}
>>>>>>>> [attr_filter.access_reject]     expand: %{User-Name} -> 1001
>>>>>>>>  attr_filter: Matched entry DEFAULT at line 11
>>>>>>>> ++[attr_filter.access_reject] returns updated
>>>>>>>> Delaying reject of request 8 for 1 seconds
>>>>>>>> Going to the next request
>>>>>>>> Waking up in 0.9 seconds.
>>>>>>>> Sending delayed reject for request 8
>>>>>>>> Sending Access-Reject of id 49 to 127.0.0.1 port 52684
>>>>>>>> Waking up in 4.9 seconds.
>>>>>>>> Cleaning up request 8 ID 49 with timestamp +7674
>>>>>>>> Ready to process requests.
>>>>>>>> WARNING! No "known good" password found for the user
>>>>>>>>
>>>>>>>> Regards,
>>>>>>>> Charles
>>>>>>>>
>>>>>>>>
>>>>>>>> 2011/8/3 Tihomir Culjaga <tculjaga at gmail.com>
>>>>>>>>
>>>>>>>>> did u use the dictionary i have attached ?
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On Tue, Aug 2, 2011 at 10:08 AM, fieldpeak <fieldpeak at gmail.com>wrote:
>>>>>>>>>
>>>>>>>>>> i tried change to 'h323-conf-id' to 'h323-call-origin' in
>>>>>>>>>> 02_unitest_rad-ANI-auth.xml, rad_auth.conf.xml, however, it still prompt
>>>>>>>>>> '[ERR] mod_rad_auth.c:428 Unknown attribute: key:h323-conf-id,
>>>>>>>>>> not found in dictionary', so where the mod_rad_auth read out the
>>>>>>>>>> 'h323-conf-id'? very very strange, which dictionary it was using...
>>>>>>>>>>
>>>>>>>>>> Regards,
>>>>>>>>>> Charles
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> 2011/8/2 fieldpeak <fieldpeak at gmail.com>
>>>>>>>>>>
>>>>>>>>>>> Hi Tihomir,
>>>>>>>>>>>
>>>>>>>>>>> Finally the answer coming, i see the hope, thanks for your reply,
>>>>>>>>>>> :)
>>>>>>>>>>>
>>>>>>>>>>> As your advise, i only use one attribute(h323-conf-id) in my
>>>>>>>>>>> dialplan, and only one attribute(h323-conf-id) in rad_auth.conf.xml, and
>>>>>>>>>>> using the attached dictionary (from ciso) which contains this attribute,
>>>>>>>>>>> however, it still prompt 'unknown attribute', so i suspected if it was
>>>>>>>>>>> reading /usr/local/etc/radiusclient/dictionary, so i copy the same
>>>>>>>>>>> dictionary to /usr/local/freeswitch/radius/, it did not any help at all...
>>>>>>>>>>> very strange...
>>>>>>>>>>>
>>>>>>>>>>> Log:
>>>>>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>>>> default_realm := .
>>>>>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>>>> radius_timeout := 3.
>>>>>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>>>> radius_retries := 2.
>>>>>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>>>> radius_deadtime := 0.
>>>>>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>>>> bindaddr := *.
>>>>>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:371 ... radius:
>>>>>>>>>>> User-Name: 38516060333
>>>>>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:380 ... radius:
>>>>>>>>>>> User-Password: 003282
>>>>>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:396 ... radius:
>>>>>>>>>>> Called-station-Id: 16094191500
>>>>>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:413 Handle
>>>>>>>>>>> attribute: h323-conf-id
>>>>>>>>>>> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:428 Unknown
>>>>>>>>>>> attribute: key:h323-conf-id, not found in dictionary
>>>>>>>>>>> 2011-08-02 15:37:26.578217 [DEBUG] mod_rad_auth.c:538 abort
>>>>>>>>>>> sending radius packet.
>>>>>>>>>>> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:546 An error
>>>>>>>>>>> occured during RADIUS Authentication(RC=-1)
>>>>>>>>>>> 2011-08-02 15:37:26.578217 [ERR] mod_rad_auth.c:702 An error
>>>>>>>>>>> occured during radius authorization.
>>>>>>>>>>>
>>>>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO
>>>>>>>>>>> AUTH_RESULT=)
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>   <extension name="unitest_rad-ANI-auth">
>>>>>>>>>>>     <condition field="destination_number" expression="^601$">
>>>>>>>>>>>       <!-- <action application="log" data="INFO  Before Auth "/>
>>>>>>>>>>> -->
>>>>>>>>>>>
>>>>>>>>>>>       <action inline="true" application="set" data="CALLID=
>>>>>>>>>>> h323-conf-id=${uuid}"/>
>>>>>>>>>>>
>>>>>>>>>>>       <action inline="true" application="set"
>>>>>>>>>>> data="USERNAME=1001"/>
>>>>>>>>>>>       <action inline="true" application="set"
>>>>>>>>>>> data="PASSWD=1111"/>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>       <action application="sleep" data="2000"/>
>>>>>>>>>>>       <action application="auth_function" data="in
>>>>>>>>>>> ${DIALED_NUMBER}, in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>
>>>>>>>>>>>
>>>>>>>>>>>     </condition>
>>>>>>>>>>>   </extension>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> <configuration name="rad_auth.conf" description="radius
>>>>>>>>>>> authentification module">
>>>>>>>>>>>   <settings>
>>>>>>>>>>>
>>>>>>>>>>>   </settings>
>>>>>>>>>>>
>>>>>>>>>>>   <client>
>>>>>>>>>>>     <param name="authserver" value="127.0.0.1:1812:gateway"/>
>>>>>>>>>>>     <param name="dictionary"
>>>>>>>>>>> value="/usr/local/etc/radiusclient/dictionary"/>
>>>>>>>>>>>     <param name="seqfile" value="/var/run/radius.seq"/>
>>>>>>>>>>>     <param name="mapfile"
>>>>>>>>>>> value="/usr/local/etc/radiusclient/port-id-map"/>
>>>>>>>>>>>     <param name="default_realm" value=""/>
>>>>>>>>>>>     <param name="radius_timeout" value="3"/>
>>>>>>>>>>>     <param name="radius_retries" value="2"/>
>>>>>>>>>>>     <param name="radius_deadtime" value="0"/>
>>>>>>>>>>>     <param name="bindaddr" value="*"/>
>>>>>>>>>>>   </client>
>>>>>>>>>>>
>>>>>>>>>>>   <vsas>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>     <param name="h323-conf-id" id="24" value="CALLID" pec="9"
>>>>>>>>>>> expr="1" direction="in"/>
>>>>>>>>>>>
>>>>>>>>>>>   </vsas>
>>>>>>>>>>>  </configuration>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> 2011/8/2 Tihomir Culjaga <tculjaga at gmail.com>
>>>>>>>>>>>
>>>>>>>>>>>> hi,
>>>>>>>>>>>>
>>>>>>>>>>>> dictionary.all is just the name of a file containing all
>>>>>>>>>>>> attributes i needed at that time.
>>>>>>>>>>>>
>>>>>>>>>>>> you can include other dictionaries by putting #INCLUDE
>>>>>>>>>>>> <pathname> at the end of the dictionary file you reference in
>>>>>>>>>>>> rad_auth.conf.xml.
>>>>>>>>>>>> if the INCLUDE doesn't work, just append dictionary.cisco to
>>>>>>>>>>>> your dictionary file... and make your own file.
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> check inline comments down below...
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> T.
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> On Sun, Jul 31, 2011 at 10:46 AM, fieldpeak <
>>>>>>>>>>>> fieldpeak at gmail.com> wrote:
>>>>>>>>>>>>
>>>>>>>>>>>>> Hello Gurus,
>>>>>>>>>>>>>
>>>>>>>>>>>>> i met a issue when using
>>>>>>>>>>>>> mod_rad_auth(http://wiki.freeswitch.org/wiki/Mod_rad_auth) to
>>>>>>>>>>>>> works
>>>>>>>>>>>>> with freeradius server+mysql for AAA, the details is below,
>>>>>>>>>>>>> Could
>>>>>>>>>>>>> anyone give any hints, Thanks in advance.
>>>>>>>>>>>>>
>>>>>>>>>>>>> i setup a dial plan "unitest_rad-ANI-auth" as wiki above,
>>>>>>>>>>>>> however,
>>>>>>>>>>>>> when i dialed 601 to trigger the dial plan, the console show
>>>>>>>>>>>>> errors,
>>>>>>>>>>>>> it looks "h323-conf-id" is not in the directory, then i tried
>>>>>>>>>>>>> to add
>>>>>>>>>>>>> this attribute to the dictionary, however, it does not help, in
>>>>>>>>>>>>> the
>>>>>>>>>>>>> wiki, it mentioned the rad_auth.conf.xml contains <param
>>>>>>>>>>>>> name="dictionary"
>>>>>>>>>>>>> value="/usr/local/etc/radiusclient/dictionary.all"/>, however i
>>>>>>>>>>>>> did
>>>>>>>>>>>>> not find the file "dictionary.all" at that directory, so i use
>>>>>>>>>>>>> dictionary. BTW, the freeradius server + mysql works well.
>>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> i just appended the information needed into dictionary.all
>>>>>>>>>>>> file... (vendor and attribute definition).
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> console errors:
>>>>>>>>>>>>>
>>>>>>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 auth_function(in ,
>>>>>>>>>>>>> in
>>>>>>>>>>>>> 38516060333, in 003282, out AUTH_RESULT)
>>>>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:301 allocate
>>>>>>>>>>>>> initial
>>>>>>>>>>>>> structure.
>>>>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:313
>>>>>>>>>>>>> initialzed configuration.
>>>>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>>>>>> authserver
>>>>>>>>>>>>> := 127.0.0.1:1812:gateway.
>>>>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>>>>>> dictionary
>>>>>>>>>>>>> := /usr/local/etc/radiusclient/dictionary.
>>>>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>>>>>> seqfile :=
>>>>>>>>>>>>> /var/run/radius.seq.
>>>>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>>>>>> mapfile :=
>>>>>>>>>>>>> /usr/local/etc/radiusclient/port-id-map.
>>>>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>>>>>> default_realm := .
>>>>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>>>>>> radius_timeout := 3.
>>>>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>>>>>> radius_retries := 2.
>>>>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>>>>>> radius_deadtime := 0.
>>>>>>>>>>>>> 2011-07-31 16:23:24.717088 [DEBUG] mod_rad_auth.c:318 set
>>>>>>>>>>>>> bindaddr := *.
>>>>>>>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:371 ...
>>>>>>>>>>>>> radius:
>>>>>>>>>>>>> User-Name: 38516060333
>>>>>>>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:380 ...
>>>>>>>>>>>>> radius:
>>>>>>>>>>>>> User-Password: 003282
>>>>>>>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:391 ...
>>>>>>>>>>>>> radius:
>>>>>>>>>>>>> Called-station-Id is empty, ignoring...
>>>>>>>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:413 Handle
>>>>>>>>>>>>> attribute: h323-conf-id
>>>>>>>>>>>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:428 Unknown
>>>>>>>>>>>>> attribute:
>>>>>>>>>>>>> key:h323-conf-id, not found in dictionary
>>>>>>>>>>>>> 2011-07-31 16:23:24.737004 [DEBUG] mod_rad_auth.c:538 abort
>>>>>>>>>>>>> sending
>>>>>>>>>>>>> radius packet.
>>>>>>>>>>>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:546 An error
>>>>>>>>>>>>> occured
>>>>>>>>>>>>> during RADIUS Authentication(RC=-1)
>>>>>>>>>>>>> 2011-07-31 16:23:24.737004 [ERR] mod_rad_auth.c:702 An error
>>>>>>>>>>>>> occured
>>>>>>>>>>>>> during radius authorization.
>>>>>>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO
>>>>>>>>>>>>>  AUTH_RESULT=)
>>>>>>>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202
>>>>>>>>>>>>>  AUTH_RESULT=
>>>>>>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO
>>>>>>>>>>>>>  billing_model=)
>>>>>>>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202
>>>>>>>>>>>>>  billing_model=
>>>>>>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO
>>>>>>>>>>>>>  credit_amount=)
>>>>>>>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202
>>>>>>>>>>>>>  credit_amount=
>>>>>>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO
>>>>>>>>>>>>>  currency=)
>>>>>>>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202  currency=
>>>>>>>>>>>>> EXECUTE sofia/internal/1001 at 124.193.106.104 log(INFO
>>>>>>>>>>>>>  preffered_lang=)
>>>>>>>>>>>>> 2011-07-31 16:23:24.737004 [INFO] mod_dptools.c:1202
>>>>>>>>>>>>>  preffered_lang=
>>>>>>>>>>>>>
>>>>>>>>>>>>> added below in the
>>>>>>>>>>>>> dictionary(/usr/local/etc/radiusclient/dictionary):
>>>>>>>>>>>>>
>>>>>>>>>>>>> ATTRIBUTE       h323-conf-id            1008    string
>>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> you need the vendor definition as well
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> dial plan:
>>>>>>>>>>>>> <extension name="unitest_rad-ANI-auth">
>>>>>>>>>>>>>    <condition field="destination_number" expression="^601$">
>>>>>>>>>>>>>      <action application="log" data="INFO  Before Auth "/>
>>>>>>>>>>>>>
>>>>>>>>>>>>>      <action inline="true" application="set"
>>>>>>>>>>>>> data="CALLID=h323-conf-id=${uuid}"/>
>>>>>>>>>>>>>      <action inline="true" application="set"
>>>>>>>>>>>>> data="SERVICENUM=h323-prompt-id=${destination_number}"/>
>>>>>>>>>>>>>      <action inline="true" application="set"
>>>>>>>>>>>>> data="TRANSACTIONID=h323-ivr-out=transactionID:1234"/>
>>>>>>>>>>>>>  <!--      <action inline="true" application="set"
>>>>>>>>>>>>> data="CALLINGNUMBER=${caller_id_number}"/> -->
>>>>>>>>>>>>>      <action inline="true" application="set"
>>>>>>>>>>>>> data="CALLINGNUMBER=38516060333"/>
>>>>>>>>>>>>>      <action inline="true" application="set"
>>>>>>>>>>>>> data="USERNAME=38516060333"/>
>>>>>>>>>>>>>  <!--      <action inline="true" application="set"
>>>>>>>>>>>>> data="USERNAME=209354"/> -->
>>>>>>>>>>>>>      <action inline="true" application="set"
>>>>>>>>>>>>> data="PASSWD=003282"/>
>>>>>>>>>>>>>  <!--      <action inline="true" application="set"
>>>>>>>>>>>>> data="DIALED_NUMBER=16094191500"/>  -->
>>>>>>>>>>>>>
>>>>>>>>>>>>>      <action application="sleep" data="2000"/>
>>>>>>>>>>>>>      <action application="auth_function" data="in
>>>>>>>>>>>>> ${DIALED_NUMBER},
>>>>>>>>>>>>> in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>>>>>>  AUTH_RESULT=${AUTH_RESULT}"/>
>>>>>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>>>>>>  billing_model=${billing_model}"/>
>>>>>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>>>>>>  credit_amount=${credit_amount}"/>
>>>>>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>>>>>>  currency=${currency}"/>
>>>>>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>>>>>>  preffered_lang=${preffered_lang}"/>
>>>>>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>>>>>>  credit_time=${credit_time}"/>
>>>>>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>>>>>> h323_ivr_duration=${h323_ivr_duration}"/>
>>>>>>>>>>>>>      <action application="log" data="INFO
>>>>>>>>>>>>>  return_code=${return_code}"/>
>>>>>>>>>>>>>      <!-- <action application="execute_extension" data="AUTH
>>>>>>>>>>>>> XML default"/> -->
>>>>>>>>>>>>>    </condition>
>>>>>>>>>>>>>  </extension>
>>>>>>>>>>>>>
>>>>>>>>>>>>>  radius_cdr.conf.xml:
>>>>>>>>>>>>>  <configuration name="radius_cdr.conf" description="RADIUS CDR
>>>>>>>>>>>>> Configuration">
>>>>>>>>>>>>>
>>>>>>>>>>>>>        <settings>
>>>>>>>>>>>>>
>>>>>>>>>>>>>                <!-- location of the radius dictionary files -->
>>>>>>>>>>>>>
>>>>>>>>>>>>>                <param name="dictionary"
>>>>>>>>>>>>> value="/usr/local/freeswitch/conf/radius/dictionary"/>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>> your dictionary file need to contain all the attributes you are
>>>>>>>>>>>> trying to use or to include other dictionaries (In this case
>>>>>>>>>>>> dictionary.cisco) from the dictionary file you are referencing here.
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>>                <!-- number of retries for each server -->
>>>>>>>>>>>>>
>>>>>>>>>>>>>                <param name="radius_retries" value="3"/>
>>>>>>>>>>>>>
>>>>>>>>>>>>>                <!-- number of seconds to wait between retries
>>>>>>>>>>>>> -->
>>>>>>>>>>>>>
>>>>>>>>>>>>>                <param name="radius_timeout" value="5"/>
>>>>>>>>>>>>>
>>>>>>>>>>>>>                <!-- accounting servers, up to 8 allowed -->
>>>>>>>>>>>>>
>>>>>>>>>>>>>                <!-- value is "host:port:secret", port is
>>>>>>>>>>>>> optional -->
>>>>>>>>>>>>>
>>>>>>>>>>>>>                <!-- use IP ADDRESSES, not hostnames -->
>>>>>>>>>>>>>
>>>>>>>>>>>>>                <param name="acct_server" value="127.0.0.1:1813
>>>>>>>>>>>>> :testing123"/>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>        </settings>
>>>>>>>>>>>>>
>>>>>>>>>>>>> </configuration>
>>>>>>>>>>>>>
>>>>>>>>>>>>>  the FS version:
>>>>>>>>>>>>>  FreeSWITCH Version 1.0.head (git-492bc6b 2011-07-23 12-53-04
>>>>>>>>>>>>> -0400)
>>>>>>>>>>>>>
>>>>>>>>>>>>>  Regards,
>>>>>>>>>>>>>  Charles
>>>>>>>>>>>>>
>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>>>>
>>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>>>
>>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>>>
>>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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,
Charles
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110814/770dd4b3/attachment-0001.html 

From mrene_lists at avgs.ca  Sun Aug 14 14:34:32 2011
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Sun, 14 Aug 2011 12:34:32 +0200
Subject: [Freeswitch-users] mod_rtmp & flex client
In-Reply-To: <CAA5vsdNFhNdXZv10+Cd6CKCQWsgTP6Eg9NsVwaRcvzsDxydioQ@mail.gmail.com>
References: <CAA5vsdNFhNdXZv10+Cd6CKCQWsgTP6Eg9NsVwaRcvzsDxydioQ@mail.gmail.com>
Message-ID: <2525F2EF-C5A6-4394-878F-61A097A9D7C8@avgs.ca>

Switch to loglevel DEBUG (press F8), it should have a lot more clues of what's going on.
 
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 2011-08-14, at 8:10 AM, Matthew Fong wrote:

> I am trying to get the mod_rtmp to work with the flex client, but I am receiving the following errors when I try to make a test call
> 
> 2011-08-14 06:04:32.744112 [NOTICE] rtmp_sig.c:121 Sent connect reply
> 2011-08-14 06:04:38.764112 [INFO] rtmp_sig.c:136 Replied to createStream (1)
> 2011-08-14 06:04:38.764112 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled control packet (type=0x3)
> 2011-08-14 06:04:38.764112 [INFO] rtmp_sig.c:136 Replied to createStream (2)
> 2011-08-14 06:04:38.764112 [NOTICE] switch_channel.c:904 New Channel rtmp/default/5000 [18578182-a702-49bc-9590-c53ec2d16b72]
> 2011-08-14 06:04:38.764112 [ERR] rtmp_sig.c:305 Couldn't create call.
> 2011-08-14 06:04:38.764112 [INFO] mod_dialplan_xml.c:336 Processing  <0000000000>->5000 in context public
> 2011-08-14 06:04:38.764112 [NOTICE] switch_core_state_machine.c:194 rtmp/default/5000 has executed the last dialplan instruction, hanging up.
> 2011-08-14 06:04:38.764112 [NOTICE] switch_core_state_machine.c:196 Hangup rtmp/default/5000 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-08-14 06:04:38.764112 [NOTICE] switch_core_session.c:1346 Session 1 (rtmp/default/5000) Ended
> 2011-08-14 06:04:38.764112 [NOTICE] switch_core_session.c:1348 Close Channel rtmp/default/5000 [CS_DESTROY]
> 2011-08-14 06:04:39.784111 [INFO] rtmp_sig.c:159 Sending audio
> 2011-08-14 06:04:39.784111 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled control packet (type=0x3)
> 2011-08-14 06:04:39.784111 [INFO] rtmp_sig.c:274 Got publish on stream 2.
> 2011-08-14 06:04:39.784111 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled control packet (type=0x3)
> 
> I am using a version checked-out from yesterday, Ubuntu 10.4 64-bit and flash debug version. Does anyone know what I am doing wrong? Or has anyone gotten it to work? Also I tried the FS conference call test using the hosted version on conference.freeswitch.org and it seems the first few seconds of audio are distorted. Is there a way to fix this? Thanks... mod_rtmp seems very promising.
> 
> --matt
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110814/6ce13dfc/attachment.html 

From jeff at jefflenk.com  Sun Aug 14 18:38:47 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Sun, 14 Aug 2011 07:38:47 -0700 (PDT)
Subject: [Freeswitch-users] incompatible destination because of crypto
	incoming calls
In-Reply-To: <942B67E68E554F0B9067531E61AE812D@omni1.local>
References: <69CD4836D15D45B683FFFBAB557B5060@omni1.local>
	<C4C8EBF3-F2FB-468E-8ABF-9A2F3B34705A@freeswitch.org>
	<942B67E68E554F0B9067531E61AE812D@omni1.local>
Message-ID: <1313332727236-6685041.post@n2.nabble.com>

Moc submitted a correction on the 9th for export that corrected this most
likely. The problem was only present for a few days.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/incompatible-destination-because-of-crypto-on-incoming-calls-tp6659067p6685041.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From avi at avimarcus.net  Sun Aug 14 21:00:53 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Sun, 14 Aug 2011 20:00:53 +0300
Subject: [Freeswitch-users] odbc basic_calls
In-Reply-To: <CALNZuw=uWC8kGww5Gjo_P7mxJ_QyrCDB1ipmamrG4QXVXb8TwQ@mail.gmail.com>
References: <CALNZuwk20r8=rmu9rOOd-aChOS6XOGVm7mxKNhd1-DnC+id7Hg@mail.gmail.com>
	<CAKbxfG-8E=5Do_mLf62tokgL5eN8=LfH954tUNRawcNCtmMTLA@mail.gmail.com>
	<CALNZuwkwgFmdmWxYMzkzGeW8u-TiRg2SbeJ1zFpOfxEPiyS2xg@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59EBABB8AD@cooper>
	<CALNZuw=uWC8kGww5Gjo_P7mxJ_QyrCDB1ipmamrG4QXVXb8TwQ@mail.gmail.com>
Message-ID: <CALNZuw=-PhE-OK38h8YPQ=00O+BPe3YE=C=XWVCXOUS+qX-W4Q@mail.gmail.com>

I dropped and recreated the channels table, which doesn't have callee_name
but sent_callee_name now. Then when I created the view manually, it worked,
but none of this got triggered to happen automatically.
-Avi


On Fri, Aug 12, 2011 at 3:05 PM, Avi Marcus <avi at avimarcus.net> wrote:

> Error in query: Unknown column 'a.sent_callee_name' in 'field list'
> So table channels.. no, I don't see that in there. I see callee_name in
> there though. Did it miss some channel table alterations?
>
> Huh, tables complete, db_data, interfaces and a few more all seem new. I've
> been running odbc in mysql for a year? but I've never seen these until now.
>
> -Avi Marcus
>
> p.s. move this to jira now that it's back up..?
>
>
> On Thu, Aug 11, 2011 at 5:02 PM, Peter Olsson <
> peter.olsson at visionutveckling.se> wrote:
>
>> Could you try to create it manually? The view's definition is in
>> switch_core_sqldb.c on line 1721.
>>
>> /Peter
>> ________________________________________
>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [
>> freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Avi Marcus [
>> avi at avimarcus.net]
>> Skickat: den 11 augusti 2011 15:48
>> Till: FreeSWITCH Users Help
>> ?mne: Re: [Freeswitch-users] odbc basic_calls
>>
>> I don't see anything in the startup log about it checking the tables:
>> http://pastebin.freeswitch.org/17013
>> This happens when I do "show calls". I know you changed something for show
>> calls to even show 1 legged IVRs since my last update, but not having looked
>> at the code, I don't see how that would be related.
>> I haven't heard of this basic_calls table before today.
>>
>> -Avi Marcus
>>
>> On Thu, Aug 11, 2011 at 4:33 PM, Anthony Minessale <
>> anthony.minessale at gmail.com<mailto:anthony.minessale at gmail.com>> wrote:
>> its a view that should be created? perhaps there is an error on
>> startup creating the view in mysuckwell?
>>
>>
>> On Thu, Aug 11, 2011 at 6:28 AM, Avi Marcus <avi at avimarcus.net<mailto:
>> avi at avimarcus.net>> wrote:
>> > I just upgraded FS since.. 7 weeks ago I think. Now: FreeSWITCH Version
>> > 1.0.head (git-9d98d49 2011-08-10 08-38-55 -0500)
>> > While testing the new build clean, when I hit F4 for show calls, I now
>> get:
>> > freeswitch at internal> 2011-08-11 14:24:04.574797 [ERR]
>> > switch_core_sqldb.c:825 ERR: [select * from basic_calls where
>> > hostname='sip2' order by call_created_epoch]
>> > [STATE: 42S02 CODE 1146 ERROR: [unixODBC][MySQL][ODBC 3.51
>> > Driver][mysqld-5.1.41-3ubuntu12.10-log]Table 'freeswitch.basic_calls'
>> > doesn't exist
>> > Aren't the core odbc tables supposed to be auto-created?
>> > -Avi
>> > _______________________________________________
>> > Join us at ClueCon 2011, Aug 9-11, Chicago
>> > http://www.cluecon.com 877-7-4ACLUE
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org<mailto:
>> FreeSWITCH-users at lists.freeswitch.org>
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com<mailto:
>> MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<mailto:
>> PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org<mailto:
>> sip%3A888 at conference.freeswitch.org>
>> googletalk:conf+888 at conference.freeswitch.org<mailto:
>> googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900<tel:%2B19193869900>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org<mailto:
>> FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> !DSPAM:4e43de1032765001342651!
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110814/b36dc749/attachment.html 

From hmkias at gmail.com  Sun Aug 14 22:38:56 2011
From: hmkias at gmail.com (hmkias at gmail.com)
Date: Sun, 14 Aug 2011 18:38:56 +0000
Subject: [Freeswitch-users] Custom SIP Profile
In-Reply-To: <4E437DF4.2050108@gmail.com>
References: <CAMXUpRy=ft+puH8T9YZabiffPMC=cZSJGV-adKEoa64f0=oBzQ@mail.gmail.com><4E437DF4.2050108@gmail.com>
Message-ID: <254116936-1313346959-cardhu_decombobulator_blackberry.rim.net-1631406973-@b13.c4.bise7.blackberry>

Thanks Nazim.
Sent from BSNL with my BlackBerry? smartphone

-----Original Message-----
From: Nazim Aghabayov <nazim.aghabayov at gmail.com>
Sender: freeswitch-users-bounces at lists.freeswitch.org
Date: Thu, 11 Aug 2011 12:00:04 
To: <freeswitch-users at lists.freeswitch.org>
Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Custom SIP Profile

Hi,
There is a nice wiki entry on sofia sip profiles at 
http://wiki.freeswitch.org/wiki/Sofia . It should answer most of your 
questions regarding sip profiles.

Best Regards,
Nazim

On 08/11/2011 04:37 AM, HM Kias wrote:
> Hi All,
>
> I m trying to create a SIP profile equivalent to the below asterisk config,
> please advise. Domain&  outboundproxy are very important in the
> registration.
>
> register =>
> XXXXXXXX at sia-nas01ca146.srg.com.bs:yyyyy:XXXXXXXX at nas-sbc-01.srg.com.bs:5060/XXXXXXXX
> [XXXXXXXX]
> host=sia-nas01ca146.srg.com.bs
> outboundproxy=nas-sbc-01.srg.com.bs
> type=friend
> username=XXXXXXXX
> fromuser=XXXXXXXX
> fromdomain=sia-nas01ca146.srg.com.bs
> secret=yyyyy
> dtmfmode=rfc2833
> disallow=all
> allow=ulaw&gsm
> context=users
> usereqphone=yes
> canreinvite=no
> insecure=port,invite
>
> Thanks in advance.
>
>
> Regards,
>
>

_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

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

From curriegrad2004 at gmail.com  Sun Aug 14 22:42:32 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sun, 14 Aug 2011 11:42:32 -0700
Subject: [Freeswitch-users] Custom SIP Profile
In-Reply-To: <254116936-1313346959-cardhu_decombobulator_blackberry.rim.net-1631406973-@b13.c4.bise7.blackberry>
References: <CAMXUpRy=ft+puH8T9YZabiffPMC=cZSJGV-adKEoa64f0=oBzQ@mail.gmail.com>
	<4E437DF4.2050108@gmail.com>
	<254116936-1313346959-cardhu_decombobulator_blackberry.rim.net-1631406973-@b13.c4.bise7.blackberry>
Message-ID: <CAMWiMy-BhEziPbtOhW1=AgFLt2pf5kMEpe-wZGe7pyrwKTiEmw@mail.gmail.com>

Quick answer: register this as a SIP gateway in the external profile

On Sun, Aug 14, 2011 at 11:38 AM,  <hmkias at gmail.com> wrote:
> Thanks Nazim.
> Sent from BSNL with my BlackBerry? smartphone
>
> -----Original Message-----
> From: Nazim Aghabayov <nazim.aghabayov at gmail.com>
> Sender: freeswitch-users-bounces at lists.freeswitch.org
> Date: Thu, 11 Aug 2011 12:00:04
> To: <freeswitch-users at lists.freeswitch.org>
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] Custom SIP Profile
>
> Hi,
> There is a nice wiki entry on sofia sip profiles at
> http://wiki.freeswitch.org/wiki/Sofia . It should answer most of your
> questions regarding sip profiles.
>
> Best Regards,
> Nazim
>
> On 08/11/2011 04:37 AM, HM Kias wrote:
>> Hi All,
>>
>> I m trying to create a SIP profile equivalent to the below asterisk config,
>> please advise. Domain& ?outboundproxy are very important in the
>> registration.
>>
>> register =>
>> XXXXXXXX at sia-nas01ca146.srg.com.bs:yyyyy:XXXXXXXX at nas-sbc-01.srg.com.bs:5060/XXXXXXXX
>> [XXXXXXXX]
>> host=sia-nas01ca146.srg.com.bs
>> outboundproxy=nas-sbc-01.srg.com.bs
>> type=friend
>> username=XXXXXXXX
>> fromuser=XXXXXXXX
>> fromdomain=sia-nas01ca146.srg.com.bs
>> secret=yyyyy
>> dtmfmode=rfc2833
>> disallow=all
>> allow=ulaw&gsm
>> context=users
>> usereqphone=yes
>> canreinvite=no
>> insecure=port,invite
>>
>> Thanks in advance.
>>
>>
>> Regards,
>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From mattdfong at gmail.com  Mon Aug 15 01:46:15 2011
From: mattdfong at gmail.com (Matthew Fong)
Date: Sun, 14 Aug 2011 14:46:15 -0700
Subject: [Freeswitch-users] mod_rtmp & flex client
In-Reply-To: <2525F2EF-C5A6-4394-878F-61A097A9D7C8@avgs.ca>
References: <CAA5vsdNFhNdXZv10+Cd6CKCQWsgTP6Eg9NsVwaRcvzsDxydioQ@mail.gmail.com>
	<2525F2EF-C5A6-4394-878F-61A097A9D7C8@avgs.ca>
Message-ID: <CAA5vsdMMJD9=wJjvvJCdWnqsg72197QeUKQWW+Mm7GjXuNYkuw@mail.gmail.com>

http://pastebin.freeswitch.org/17036

I looked thru the messages, but nothing apparent came out before

2011-08-14 21:40:43.284111 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled
control packet (type=0x3)

and

2011-08-14 21:40:43.284111 [ERR] rtmp_sig.c:305 Couldn't create call.

is there something I need to be considering on the .html/flex side? Thanks
for the help.

--matt



On Sun, Aug 14, 2011 at 3:34 AM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> Switch to loglevel DEBUG (press F8), it should have a lot more clues of
> what's going on.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 2011-08-14, at 8:10 AM, Matthew Fong wrote:
>
> I am trying to get the mod_rtmp to work with the flex client, but I am
> receiving the following errors when I try to make a test call
>
> 2011-08-14 06:04:32.744112 [NOTICE] rtmp_sig.c:121 Sent connect reply
> 2011-08-14 06:04:38.764112 [INFO] rtmp_sig.c:136 Replied to createStream
> (1)
> 2011-08-14 06:04:38.764112 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled
> control packet (type=0x3)
> 2011-08-14 06:04:38.764112 [INFO] rtmp_sig.c:136 Replied to createStream
> (2)
> 2011-08-14 06:04:38.764112 [NOTICE] switch_channel.c:904 New Channel
> rtmp/default/5000 [18578182-a702-49bc-9590-c53ec2d16b72]
> 2011-08-14 06:04:38.764112 [ERR] rtmp_sig.c:305 Couldn't create call.
> 2011-08-14 06:04:38.764112 [INFO] mod_dialplan_xml.c:336 Processing
>  <0000000000>->5000 in context public
> 2011-08-14 06:04:38.764112 [NOTICE] switch_core_state_machine.c:194
> rtmp/default/5000 has executed the last dialplan instruction, hanging up.
> 2011-08-14 06:04:38.764112 [NOTICE] switch_core_state_machine.c:196 Hangup
> rtmp/default/5000 [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-08-14 06:04:38.764112 [NOTICE] switch_core_session.c:1346 Session 1
> (rtmp/default/5000) Ended
> 2011-08-14 06:04:38.764112 [NOTICE] switch_core_session.c:1348 Close
> Channel rtmp/default/5000 [CS_DESTROY]
> 2011-08-14 06:04:39.784111 [INFO] rtmp_sig.c:159 Sending audio
> 2011-08-14 06:04:39.784111 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled
> control packet (type=0x3)
> 2011-08-14 06:04:39.784111 [INFO] rtmp_sig.c:274 Got publish on stream 2.
> 2011-08-14 06:04:39.784111 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled
> control packet (type=0x3)
>
> I am using a version checked-out from yesterday, Ubuntu 10.4 64-bit and
> flash debug version. Does anyone know what I am doing wrong? Or has anyone
> gotten it to work? Also I tried the FS conference call test using the hosted
> version on conference.freeswitch.org and it seems the first few seconds of
> audio are distorted. Is there a way to fix this? Thanks... mod_rtmp seems
> very promising.
>
> --matt
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110814/6adf35f6/attachment.html 

From mrene_lists at avgs.ca  Mon Aug 15 01:57:41 2011
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Sun, 14 Aug 2011 23:57:41 +0200
Subject: [Freeswitch-users] mod_rtmp & flex client
In-Reply-To: <CAA5vsdMMJD9=wJjvvJCdWnqsg72197QeUKQWW+Mm7GjXuNYkuw@mail.gmail.com>
References: <CAA5vsdNFhNdXZv10+Cd6CKCQWsgTP6Eg9NsVwaRcvzsDxydioQ@mail.gmail.com>
	<2525F2EF-C5A6-4394-878F-61A097A9D7C8@avgs.ca>
	<CAA5vsdMMJD9=wJjvvJCdWnqsg72197QeUKQWW+Mm7GjXuNYkuw@mail.gmail.com>
Message-ID: <11AA0E83-2E71-4953-97A5-34B159903DF4@avgs.ca>

The couldn't create call one should've been printed, the call was indeed launched, but if you look carefully, it has nowhere to go.

2011-08-14 21:40:43.284111 [NOTICE] switch_core_state_machine.c:194 rtmp/default/5000 has executed the last dialplan instruction, hanging up.

That means the dialplan didn't queue any actions for that channel.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 2011-08-14, at 11:46 PM, Matthew Fong wrote:

> http://pastebin.freeswitch.org/17036
> 
> I looked thru the messages, but nothing apparent came out before 
> 
> 2011-08-14 21:40:43.284111 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled control packet (type=0x3)
> 
> and
> 
> 2011-08-14 21:40:43.284111 [ERR] rtmp_sig.c:305 Couldn't create call.
> 
> is there something I need to be considering on the .html/flex side? Thanks for the help.
> 
> --matt
> 
> 
> 
> On Sun, Aug 14, 2011 at 3:34 AM, Mathieu Rene <mrene_lists at avgs.ca> wrote:
> Switch to loglevel DEBUG (press F8), it should have a lot more clues of what's going on.
>  
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
> 
> 
> 
> 
> On 2011-08-14, at 8:10 AM, Matthew Fong wrote:
> 
>> I am trying to get the mod_rtmp to work with the flex client, but I am receiving the following errors when I try to make a test call
>> 
>> 2011-08-14 06:04:32.744112 [NOTICE] rtmp_sig.c:121 Sent connect reply
>> 2011-08-14 06:04:38.764112 [INFO] rtmp_sig.c:136 Replied to createStream (1)
>> 2011-08-14 06:04:38.764112 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled control packet (type=0x3)
>> 2011-08-14 06:04:38.764112 [INFO] rtmp_sig.c:136 Replied to createStream (2)
>> 2011-08-14 06:04:38.764112 [NOTICE] switch_channel.c:904 New Channel rtmp/default/5000 [18578182-a702-49bc-9590-c53ec2d16b72]
>> 2011-08-14 06:04:38.764112 [ERR] rtmp_sig.c:305 Couldn't create call.
>> 2011-08-14 06:04:38.764112 [INFO] mod_dialplan_xml.c:336 Processing  <0000000000>->5000 in context public
>> 2011-08-14 06:04:38.764112 [NOTICE] switch_core_state_machine.c:194 rtmp/default/5000 has executed the last dialplan instruction, hanging up.
>> 2011-08-14 06:04:38.764112 [NOTICE] switch_core_state_machine.c:196 Hangup rtmp/default/5000 [CS_EXECUTE] [NORMAL_CLEARING]
>> 2011-08-14 06:04:38.764112 [NOTICE] switch_core_session.c:1346 Session 1 (rtmp/default/5000) Ended
>> 2011-08-14 06:04:38.764112 [NOTICE] switch_core_session.c:1348 Close Channel rtmp/default/5000 [CS_DESTROY]
>> 2011-08-14 06:04:39.784111 [INFO] rtmp_sig.c:159 Sending audio
>> 2011-08-14 06:04:39.784111 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled control packet (type=0x3)
>> 2011-08-14 06:04:39.784111 [INFO] rtmp_sig.c:274 Got publish on stream 2.
>> 2011-08-14 06:04:39.784111 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled control packet (type=0x3)
>> 
>> I am using a version checked-out from yesterday, Ubuntu 10.4 64-bit and flash debug version. Does anyone know what I am doing wrong? Or has anyone gotten it to work? Also I tried the FS conference call test using the hosted version on conference.freeswitch.org and it seems the first few seconds of audio are distorted. Is there a way to fix this? Thanks... mod_rtmp seems very promising.
>> 
>> --matt
>> 
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From reply at matthewfong.com  Mon Aug 15 02:18:04 2011
From: reply at matthewfong.com (Matthew Fong)
Date: Sun, 14 Aug 2011 15:18:04 -0700
Subject: [Freeswitch-users] mod_rtmp & flex client
In-Reply-To: <11AA0E83-2E71-4953-97A5-34B159903DF4@avgs.ca>
References: <CAA5vsdNFhNdXZv10+Cd6CKCQWsgTP6Eg9NsVwaRcvzsDxydioQ@mail.gmail.com>
	<2525F2EF-C5A6-4394-878F-61A097A9D7C8@avgs.ca>
	<CAA5vsdMMJD9=wJjvvJCdWnqsg72197QeUKQWW+Mm7GjXuNYkuw@mail.gmail.com>
	<11AA0E83-2E71-4953-97A5-34B159903DF4@avgs.ca>
Message-ID: <CAA5vsdP+WWU_0UrKUR47i8TR26O-vasSsE8S2Lnt+MpfVwGXfw@mail.gmail.com>

Hi Mathieu,

I know know why I missed that. Thanks!

--matt

On Sun, Aug 14, 2011 at 2:57 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> The couldn't create call one should've been printed, the call was indeed
> launched, but if you look carefully, it has nowhere to go.
>
> 2011-08-14 21:40:43.284111 [NOTICE] switch_core_state_machine.c:194
> rtmp/default/5000 has executed the last dialplan instruction, hanging up.
>
> That means the dialplan didn't queue any actions for that channel.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 2011-08-14, at 11:46 PM, Matthew Fong wrote:
>
> > http://pastebin.freeswitch.org/17036
> >
> > I looked thru the messages, but nothing apparent came out before
> >
> > 2011-08-14 21:40:43.284111 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled
> control packet (type=0x3)
> >
> > and
> >
> > 2011-08-14 21:40:43.284111 [ERR] rtmp_sig.c:305 Couldn't create call.
> >
> > is there something I need to be considering on the .html/flex side?
> Thanks for the help.
> >
> > --matt
> >
> >
> >
> > On Sun, Aug 14, 2011 at 3:34 AM, Mathieu Rene <mrene_lists at avgs.ca>
> wrote:
> > Switch to loglevel DEBUG (press F8), it should have a lot more clues of
> what's going on.
> >
> > Mathieu Rene
> > Avant-Garde Solutions Inc
> > Office: + 1 (514) 664-1044 x100
> > Cell: +1 (514) 664-1044 x200
> > mrene at avgs.ca
> >
> >
> >
> >
> > On 2011-08-14, at 8:10 AM, Matthew Fong wrote:
> >
> >> I am trying to get the mod_rtmp to work with the flex client, but I am
> receiving the following errors when I try to make a test call
> >>
> >> 2011-08-14 06:04:32.744112 [NOTICE] rtmp_sig.c:121 Sent connect reply
> >> 2011-08-14 06:04:38.764112 [INFO] rtmp_sig.c:136 Replied to createStream
> (1)
> >> 2011-08-14 06:04:38.764112 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled
> control packet (type=0x3)
> >> 2011-08-14 06:04:38.764112 [INFO] rtmp_sig.c:136 Replied to createStream
> (2)
> >> 2011-08-14 06:04:38.764112 [NOTICE] switch_channel.c:904 New Channel
> rtmp/default/5000 [18578182-a702-49bc-9590-c53ec2d16b72]
> >> 2011-08-14 06:04:38.764112 [ERR] rtmp_sig.c:305 Couldn't create call.
> >> 2011-08-14 06:04:38.764112 [INFO] mod_dialplan_xml.c:336 Processing
>  <0000000000>->5000 in context public
> >> 2011-08-14 06:04:38.764112 [NOTICE] switch_core_state_machine.c:194
> rtmp/default/5000 has executed the last dialplan instruction, hanging up.
> >> 2011-08-14 06:04:38.764112 [NOTICE] switch_core_state_machine.c:196
> Hangup rtmp/default/5000 [CS_EXECUTE] [NORMAL_CLEARING]
> >> 2011-08-14 06:04:38.764112 [NOTICE] switch_core_session.c:1346 Session 1
> (rtmp/default/5000) Ended
> >> 2011-08-14 06:04:38.764112 [NOTICE] switch_core_session.c:1348 Close
> Channel rtmp/default/5000 [CS_DESTROY]
> >> 2011-08-14 06:04:39.784111 [INFO] rtmp_sig.c:159 Sending audio
> >> 2011-08-14 06:04:39.784111 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled
> control packet (type=0x3)
> >> 2011-08-14 06:04:39.784111 [INFO] rtmp_sig.c:274 Got publish on stream
> 2.
> >> 2011-08-14 06:04:39.784111 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled
> control packet (type=0x3)
> >>
> >> I am using a version checked-out from yesterday, Ubuntu 10.4 64-bit and
> flash debug version. Does anyone know what I am doing wrong? Or has anyone
> gotten it to work? Also I tried the FS conference call test using the hosted
> version on conference.freeswitch.org and it seems the first few seconds of
> audio are distorted. Is there a way to fix this? Thanks... mod_rtmp seems
> very promising.
> >>
> >> --matt
> >>
> >> _______________________________________________
> >> Join us at ClueCon 2011, Aug 9-11, Chicago
> >> http://www.cluecon.com 877-7-4ACLUE
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110814/6e893f37/attachment-0001.html 

From freeswitch-list at puzzled.xs4all.nl  Mon Aug 15 04:10:22 2011
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Mon, 15 Aug 2011 02:10:22 +0200
Subject: [Freeswitch-users] Secure connection
In-Reply-To: <CAMWiMy-GFqHnfK=OB2SwKWF=tXNXT75K-8=ZvfMW7vE+XoGrTA@mail.gmail.com>
References: <4E467F1A.6030600@interfree.it>	<6B50778F-B8F2-49A0-AFB4-201B62AD488D@gmail.com>
	<CAMWiMy-GFqHnfK=OB2SwKWF=tXNXT75K-8=ZvfMW7vE+XoGrTA@mail.gmail.com>
Message-ID: <4E4863EE.2000605@puzzled.xs4all.nl>

On 08/13/2011 08:49 PM, curriegrad2004 wrote:
> OpenVPN, IPSec +L2TP and PPTP comes to mind here for VPN solutions

Afaik PPtP is not very secure (it's a Windows thing) and IPSec+L2TP is 
secure but quite a challenge to setup so I would suggest OpenVPN with 
client certificate authentication.

Regards,
Patrick


From avi at avimarcus.net  Mon Aug 15 04:25:30 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 15 Aug 2011 03:25:30 +0300
Subject: [Freeswitch-users] Secure connection
In-Reply-To: <4E4863EE.2000605@puzzled.xs4all.nl>
References: <4E467F1A.6030600@interfree.it>
	<6B50778F-B8F2-49A0-AFB4-201B62AD488D@gmail.com>
	<CAMWiMy-GFqHnfK=OB2SwKWF=tXNXT75K-8=ZvfMW7vE+XoGrTA@mail.gmail.com>
	<4E4863EE.2000605@puzzled.xs4all.nl>
Message-ID: <CALNZuw=pMPiVL56qTaGR_n7fsCdfE95DLGosbVzCNVwxK6wsCA@mail.gmail.com>

A simple ssh tunnel with port mapping should work..

-Avi


On Mon, Aug 15, 2011 at 3:10 AM, Patrick Lists <
freeswitch-list at puzzled.xs4all.nl> wrote:

> On 08/13/2011 08:49 PM, curriegrad2004 wrote:
> > OpenVPN, IPSec +L2TP and PPTP comes to mind here for VPN solutions
>
> Afaik PPtP is not very secure (it's a Windows thing) and IPSec+L2TP is
> secure but quite a challenge to setup so I would suggest OpenVPN with
> client certificate authentication.
>
> Regards,
> Patrick
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110815/f6877522/attachment.html 

From 12ukwn at gmail.com  Mon Aug 15 04:27:46 2011
From: 12ukwn at gmail.com (Jean-Yves F. Barbier)
Date: Mon, 15 Aug 2011 02:27:46 +0200
Subject: [Freeswitch-users] Secure connection
In-Reply-To: <4E4863EE.2000605@puzzled.xs4all.nl>
References: <4E467F1A.6030600@interfree.it>
	<6B50778F-B8F2-49A0-AFB4-201B62AD488D@gmail.com>
	<CAMWiMy-GFqHnfK=OB2SwKWF=tXNXT75K-8=ZvfMW7vE+XoGrTA@mail.gmail.com>
	<4E4863EE.2000605@puzzled.xs4all.nl>
Message-ID: <20110815022746.2154693b@anubis.defcon1>

On Mon, 15 Aug 2011 02:10:22 +0200, Patrick Lists
<freeswitch-list at puzzled.xs4all.nl> wrote:

> On 08/13/2011 08:49 PM, curriegrad2004 wrote:
> > OpenVPN, IPSec +L2TP and PPTP comes to mind here for VPN solutions
> 
> Afaik PPtP is not very secure (it's a Windows thing) and IPSec+L2TP is 
> secure but quite a challenge to setup so I would suggest OpenVPN with 
> client certificate authentication.

I don't see why people are talking about VPN: it is one more risk as both
machines are widely opened to each other.
Tunneling only the wanted service reduce this risk to almost nothing.

-- 
BOFH excuse #86:
Runt packets


From nbhatti at gmail.com  Mon Aug 15 13:22:00 2011
From: nbhatti at gmail.com (nbhatti)
Date: Mon, 15 Aug 2011 02:22:00 -0700 (PDT)
Subject: [Freeswitch-users] Open Source Billing for FreeSWITCH to get
	public soon ..
In-Reply-To: <CAMWiMy8=9C9mxdtPLkvoC=Kw+wWrV72N=872JVkz6TEpnyzRgA@mail.gmail.com>
References: <CACak_yx0UBoxVXiCBkWuRLJ0463QR+NRMMaiR_zhEb8n=_Fb9A@mail.gmail.com>
	<CAGpNWUnNNAQ5SwMdLvgkc5-71qkGEpwePA0LCv44OC1KTBn5eA@mail.gmail.com>
	<CAMWiMy8=9C9mxdtPLkvoC=Kw+wWrV72N=872JVkz6TEpnyzRgA@mail.gmail.com>
Message-ID: <1313400120174-6686831.post@n2.nabble.com>

Thank you everyone for your interest in vBilling. I received numerous emails
asking for the beta test. Just hang on tight and I will release more
information in a few days. Do keep an eye on
https://github.com/digitallinx/vBilling where it is going to get published. 
BTW: This is Goni on freenode if anyone likes to speak to me. :)

-B

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Open-Source-Billing-for-FreeSWITCH-to-get-public-soon-tp6684015p6686831.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From x.liu at hw.ac.uk  Mon Aug 15 14:05:23 2011
From: x.liu at hw.ac.uk (xl127)
Date: Mon, 15 Aug 2011 11:05:23 +0100
Subject: [Freeswitch-users] any IVR example in C/C++?
In-Reply-To: <CAA4nhyAnC0HfxJJPGpOdiBGaG__gBf6Bp9KVi6spjDQgjLTZ5w@mail.gmail.com>
References: <4E43EE36.60209@hw.ac.uk>
	<CAA4nhyAnC0HfxJJPGpOdiBGaG__gBf6Bp9KVi6spjDQgjLTZ5w@mail.gmail.com>
Message-ID: <4E48EF63.8040606@hw.ac.uk>

Thanks for the answer, Moises!

My final goal is to let FreeSwitch (FS) to communicate with my external 
applications in Java, C, or Python,
e.g. to send the ASR results from FS to my external applications.

I know I could do it by FS Event Socket but I am a bit worried about the 
performances by using the Event Socket.
So I thought it might be better to use embedded FS in my apps, set the 
speech input callback to point to
the method in my app.

To use embedded FS, it seems to be easier to find the C API e.g.

switch_core_init_and_modload( ..)

so I was thinking about the FS IVR in C/C++.
(I couldn't find the APIs for embedded FS in others of my chosen 
languages: Java, C and Python)

any suggestions for accomplishing the goal?

Thanks,
Xing


On 13/08/11 17:48, Moises Silva wrote:
> On Thu, Aug 11, 2011 at 10:59 AM, xl127<x.liu at hw.ac.uk>  wrote:
>> I am wondering how I could do this for a C/C++ application?
>> And in the scripts languages I can set a callback method, e.g.
>>    session.setInputCallback(myInputCallback)
>> but I didn't find how to do this in C/C++.
> The default question here is, why do you need C/C++ for an IVR?
> FreeSWITCH allows you to use simpler/safer languages to build IVR's.
>
> You can certainly do it, but the reason you don't find examples is
> probably because most people understand there is no need for C/C++
> there. Having said that, you can take a look at the IVR/say/play API's
> in switch_ivr_play_say.c to find out how to provide a callback to the
> different API's thru the switch_input_args_t structure.
>
> Moises Silva
> Senior Software Engineer, Software Development Manager
> Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
> L3R 9R6 Canada
> t. 1 905 474 1990 x128 | e. moy at sangoma.com
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Heriot-Watt University is a Scottish charity
registered under charity number SC000278.



From gmaruzz at gmail.com  Mon Aug 15 16:22:54 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Mon, 15 Aug 2011 14:22:54 +0200
Subject: [Freeswitch-users] any IVR example in C/C++?
In-Reply-To: <4E48EF63.8040606@hw.ac.uk>
References: <4E43EE36.60209@hw.ac.uk>
	<CAA4nhyAnC0HfxJJPGpOdiBGaG__gBf6Bp9KVi6spjDQgjLTZ5w@mail.gmail.com>
	<4E48EF63.8040606@hw.ac.uk>
Message-ID: <CALXCt0qfiOQWTbhHhBHsZkUH4FuwT0-hbp_N+p=Njz0tSiC9Mw@mail.gmail.com>

don't be worried about the performances of Event Socket. Nobody has
had that problem (with huuuuuuge loads).

Just test it in a load similar (or double) what you expect to be the
max, and you'll see no reason to worry.

-giovanni

On Mon, Aug 15, 2011 at 12:05 PM, xl127 <x.liu at hw.ac.uk> wrote:
> Thanks for the answer, Moises!
>
> My final goal is to let FreeSwitch (FS) to communicate with my external
> applications in Java, C, or Python,
> e.g. to send the ASR results from FS to my external applications.
>
> I know I could do it by FS Event Socket but I am a bit worried about the
> performances by using the Event Socket.
> So I thought it might be better to use embedded FS in my apps, set the
> speech input callback to point to
> the method in my app.
>
> To use embedded FS, it seems to be easier to find the C API e.g.
>
> switch_core_init_and_modload( ..)
>
> so I was thinking about the FS IVR in C/C++.
> (I couldn't find the APIs for embedded FS in others of my chosen
> languages: Java, C and Python)
>
> any suggestions for accomplishing the goal?
>
> Thanks,
> Xing
>
>
> On 13/08/11 17:48, Moises Silva wrote:
>> On Thu, Aug 11, 2011 at 10:59 AM, xl127<x.liu at hw.ac.uk> ?wrote:
>>> I am wondering how I could do this for a C/C++ application?
>>> And in the scripts languages I can set a callback method, e.g.
>>> ? ?session.setInputCallback(myInputCallback)
>>> but I didn't find how to do this in C/C++.
>> The default question here is, why do you need C/C++ for an IVR?
>> FreeSWITCH allows you to use simpler/safer languages to build IVR's.
>>
>> You can certainly do it, but the reason you don't find examples is
>> probably because most people understand there is no need for C/C++
>> there. Having said that, you can take a look at the IVR/say/play API's
>> in switch_ivr_play_say.c to find out how to provide a callback to the
>> different API's thru the switch_input_args_t structure.
>>
>> Moises Silva
>> Senior Software Engineer, Software Development Manager
>> Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
>> L3R 9R6 Canada
>> t. 1 905 474 1990 x128 | e. moy at sangoma.com
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Heriot-Watt University is a Scottish charity
> registered under charity number SC000278.
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 x.liu at hw.ac.uk  Mon Aug 15 17:15:30 2011
From: x.liu at hw.ac.uk (xl127)
Date: Mon, 15 Aug 2011 14:15:30 +0100
Subject: [Freeswitch-users] any IVR example in C/C++?
In-Reply-To: <CALXCt0qfiOQWTbhHhBHsZkUH4FuwT0-hbp_N+p=Njz0tSiC9Mw@mail.gmail.com>
References: <4E43EE36.60209@hw.ac.uk>	<CAA4nhyAnC0HfxJJPGpOdiBGaG__gBf6Bp9KVi6spjDQgjLTZ5w@mail.gmail.com>	<4E48EF63.8040606@hw.ac.uk>
	<CALXCt0qfiOQWTbhHhBHsZkUH4FuwT0-hbp_N+p=Njz0tSiC9Mw@mail.gmail.com>
Message-ID: <4E491BF2.2000200@hw.ac.uk>

okay, thanks! I will have a try on it!

Cheers,

Xing


On 15/08/11 13:22, Giovanni Maruzzelli wrote:
> don't be worried about the performances of Event Socket. Nobody has
> had that problem (with huuuuuuge loads).
>
> Just test it in a load similar (or double) what you expect to be the
> max, and you'll see no reason to worry.
>
> -giovanni
>
> On Mon, Aug 15, 2011 at 12:05 PM, xl127<x.liu at hw.ac.uk>  wrote:
>> Thanks for the answer, Moises!
>>
>> My final goal is to let FreeSwitch (FS) to communicate with my external
>> applications in Java, C, or Python,
>> e.g. to send the ASR results from FS to my external applications.
>>
>> I know I could do it by FS Event Socket but I am a bit worried about the
>> performances by using the Event Socket.
>> So I thought it might be better to use embedded FS in my apps, set the
>> speech input callback to point to
>> the method in my app.
>>
>> To use embedded FS, it seems to be easier to find the C API e.g.
>>
>> switch_core_init_and_modload( ..)
>>
>> so I was thinking about the FS IVR in C/C++.
>> (I couldn't find the APIs for embedded FS in others of my chosen
>> languages: Java, C and Python)
>>
>> any suggestions for accomplishing the goal?
>>
>> Thanks,
>> Xing
>>
>>
>> On 13/08/11 17:48, Moises Silva wrote:
>>> On Thu, Aug 11, 2011 at 10:59 AM, xl127<x.liu at hw.ac.uk>    wrote:
>>>> I am wondering how I could do this for a C/C++ application?
>>>> And in the scripts languages I can set a callback method, e.g.
>>>>     session.setInputCallback(myInputCallback)
>>>> but I didn't find how to do this in C/C++.
>>> The default question here is, why do you need C/C++ for an IVR?
>>> FreeSWITCH allows you to use simpler/safer languages to build IVR's.
>>>
>>> You can certainly do it, but the reason you don't find examples is
>>> probably because most people understand there is no need for C/C++
>>> there. Having said that, you can take a look at the IVR/say/play API's
>>> in switch_ivr_play_say.c to find out how to provide a callback to the
>>> different API's thru the switch_input_args_t structure.
>>>
>>> Moises Silva
>>> Senior Software Engineer, Software Development Manager
>>> Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
>>> L3R 9R6 Canada
>>> t. 1 905 474 1990 x128 | e. moy at sangoma.com
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> --
>> Heriot-Watt University is a Scottish charity
>> registered under charity number SC000278.
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>



-- 
Heriot-Watt University is a Scottish charity
registered under charity number SC000278.



From adrottenberg at gmail.com  Mon Aug 15 18:18:45 2011
From: adrottenberg at gmail.com (Duvid Rottenberg)
Date: Mon, 15 Aug 2011 10:18:45 -0400
Subject: [Freeswitch-users] Playback .wma/.wmv
In-Reply-To: <ADE331F6-F99A-4FEF-B85C-821CDFDA585A@freeswitch.org>
References: <CAC4JBEDuBqK1KfsWG4rY1dd8ASk9zGzAgP76YqxbpJX9Dg-YmA@mail.gmail.com>
	<1313081473061-6677105.post@n2.nabble.com>
	<CAC4JBECvNjZqC66HC9MqE9HVghHkTOn5xFSt_RLZ6gBk+TCOuw@mail.gmail.com>
	<ADE331F6-F99A-4FEF-B85C-821CDFDA585A@freeswitch.org>
Message-ID: <CAC4JBEChJgfjbTm3q31ObbzLqezY4RnQX8cAX+bbYFy2MngXRQ@mail.gmail.com>

Thanks for the input, I think that would meet my current needs I will look
into that. Also, I'm not an expert in licensing, but I did some googling and
it appears that the MPL used by freeswitch is not fully compatible with LGPL
used by ffmpeg and I would only be allowed to statically link to the ffmpeg
library and not include the code.


On Sat, Aug 13, 2011 at 11:25 PM, Brian West <brian at freeswitch.org> wrote:

> Use something that is license compatible which I think gstreamer is but I
> am not aware that gstreamer comes with the codec for WMA/WMV.  ffmpeg comes
> with the codec but personally I would extract the wma encoder and decoder
> and form a standalone lib if possible to use for mod_wma or something.
> Just my take.
>
> /b
>
> On Aug 11, 2011, at 12:15 PM, Duvid Rottenberg wrote:
>
> > Thanks for the response. I guess I will have to write that myself (need
> to
> > brush up my c programming skills), I am thinking of doing a mod using
> > gstreamer. Does anyone have any recommendations?
> >
> > On Thu, Aug 11, 2011 at 12:51 PM, Jeff Lenk <jeff at jefflenk.com> wrote:
> >
> >> Not at this time. AFAIK
> >>
> >> --
> >> View this message in context:
> >>
> http://freeswitch-users.2379917.n2.nabble.com/Playback-wma-wmv-tp6674880p6677105.html
> >> Sent from the freeswitch-users mailing list archive at Nabble.com.
> >>
> >> _______________________________________________
> >> Join us at ClueCon 2011, Aug 9-11, Chicago
> >> http://www.cluecon.com 877-7-4ACLUE
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110815/b191c792/attachment-0001.html 

From loyglenn at rock.com  Sun Aug 14 20:41:23 2011
From: loyglenn at rock.com (Loy Glenn)
Date: Sun, 14 Aug 2011 11:41:23 -0500
Subject: [Freeswitch-users] Record-Route Header Rejection
Message-ID: <20110814164123.1FDE91601E4@c-in3ws--03-05.sv2.lotuslive.com>

Accepted Header
Record-Route: <sip:xxx.xxx.xxx.xxx;lr=on;ftag=gK0653a65f>
 
Rejected Header
Record-Route: <sip:xxx.xxx.xxx.xxx;lr=on;ftag=SDpr0p702-3521913535-971555%uukeghgcjiudgieijkcuu0u>
 
So it must be the ftag...What is wrong with it? Or am I completely off the mark?

FreeSWITCH returns
SIP/2.0 400 Bad Record-Route Header
 
FreeSWITCH Version 1.0.head (git-6d1d4a9 2011-08-09 16-48-58 -0500)

-- 
You Rock! Your E-Mail Should Too! Signup Now at Rock.com and get 2GB of Storage!

http://connections.rock.com/user/displayUserRegisterPage.kickAction?as=116748&STATUS=MAIN


From govoiper at gmail.com  Mon Aug 15 09:58:16 2011
From: govoiper at gmail.com (Sam Govind)
Date: Mon, 15 Aug 2011 10:58:16 +0500
Subject: [Freeswitch-users] SIP proxy collect DTMF using FS
In-Reply-To: <0D92C5B0-CD84-47B1-A17C-A2B083B760E2@freeswitch.org>
References: <CAJUJwtjqMyvVX1eteYBWFf41Mz+fZjnRZA_0BR2DZs0R5zh_tA@mail.gmail.com>
	<0D92C5B0-CD84-47B1-A17C-A2B083B760E2@freeswitch.org>
Message-ID: <CAJUJwthVmvx3vvHwpO+ZOS0tDaLREp4H3OfX39hBJQ0cJOvp_Q@mail.gmail.com>

Thanks Brian for showing concern. I'm always open for ideas. What I'm trying
to achieve is collect DTMF from user and then have my SIP Proxy verify if a
particular caller is allowed to dial that destination(input as DTMF).

Obviously I could've done the same checks at FS, BUT I'm *required *to let
SIP Proxy verify instead. FreeSWITCH is only required to get input, release
the call and if Proxy allows the call only then call be routed to any other
FreeSWITCH (Pool of FS LoadBalanced).

This is supposed to simplify the operations of FS , decrease the load volume
on FS, and increase the call capacity.

If you've any better Ideas do share.

On Sun, Aug 14, 2011 at 8:39 AM, Brian West <brian at freeswitch.org> wrote:

> What are you trying to do exactly because it sounds like you have selected
> the most painful way to accomplish it?
>
> /b
>
> On Aug 9, 2011, at 2:11 AM, Sam Govind wrote:
>
> Hi guys,
>
> I'm looking to establish a scenario like this, any idea how to do it, if
> its
> possible.
>
> 1- SIP proxy send call to FS where DTMF will be collected. (I'm thinking of
> using PlayAndGetDigits)
> 2- DTMF collected be sent back to SIP proxy while FS ends the call
> 3- Call at SIP proxy end keeps running for some other processing.
>
> basically I just need FS to collect DTMF and send those back to SIP Proxy.
>
> Any ideas are welcome.
>
> Thanks.
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110815/94b843d0/attachment.html 

From petr at petris.info  Mon Aug 15 15:45:58 2011
From: petr at petris.info (Petr Nyklicek)
Date: Mon, 15 Aug 2011 13:45:58 +0200
Subject: [Freeswitch-users] freeswitch media_proxy and zrtp
Message-ID: <4E4906F6.9070609@petris.info>

Hi all,

I`ve freeswitch from git and zrtp 0.81.514. Zrtp with MITM is working. 
But I need proxy_media mode.

In sip/internal.xml :

     <param name="inbound-proxy-media" value="true"/>
     <param name="inbound-late-negotiation" value="true"/>

... but zrtp stream is still processes by FS

Thanks


Petr



From msc at freeswitch.org  Mon Aug 15 19:25:39 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 15 Aug 2011 10:25:39 -0500
Subject: [Freeswitch-users] Playback .wma/.wmv
In-Reply-To: <CAC4JBEChJgfjbTm3q31ObbzLqezY4RnQX8cAX+bbYFy2MngXRQ@mail.gmail.com>
References: <CAC4JBEDuBqK1KfsWG4rY1dd8ASk9zGzAgP76YqxbpJX9Dg-YmA@mail.gmail.com>
	<1313081473061-6677105.post@n2.nabble.com>
	<CAC4JBECvNjZqC66HC9MqE9HVghHkTOn5xFSt_RLZ6gBk+TCOuw@mail.gmail.com>
	<ADE331F6-F99A-4FEF-B85C-821CDFDA585A@freeswitch.org>
	<CAC4JBEChJgfjbTm3q31ObbzLqezY4RnQX8cAX+bbYFy2MngXRQ@mail.gmail.com>
Message-ID: <CAKzWOxWt40neGHvt0gtJgNM0Oxpkw+ZDqTFaSCFBB1xO1eVKFw@mail.gmail.com>

On Mon, Aug 15, 2011 at 9:18 AM, Duvid Rottenberg <adrottenberg at gmail.com>wrote:

> Thanks for the input, I think that would meet my current needs I will look
> into that. Also, I'm not an expert in licensing, but I did some googling and
> it appears that the MPL used by freeswitch is not fully compatible with LGPL
> used by ffmpeg and I would only be allowed to statically link to the ffmpeg
> library and not include the code.
>

Static linking sounds right if you need to include LGPL inside your
MPL-based module.
-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110815/bba488e2/attachment.html 

From msc at freeswitch.org  Mon Aug 15 19:44:58 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 15 Aug 2011 10:44:58 -0500
Subject: [Freeswitch-users] Running Post-Bridge scripts
In-Reply-To: <1ABE503ACB3B1D49A68430AE5DA62A552067B70B21@ESBS2K8.AMG.local>
References: <AcxYN0Jot7MOiw75RC2Acqe1LrKfUA==>
	<1ABE503ACB3B1D49A68430AE5DA62A552067B70B21@ESBS2K8.AMG.local>
Message-ID: <CAKzWOxU0sobpaC3p57SYeLPpZ-2tMAVDM7ZFstwwocoerqfz3g@mail.gmail.com>

I think Avi is right. In your case you want to use:
api_hangup_hook
and
session_in_hangup_hook

See this wiki page for tips on how to access the "session" information in a
hangup hook script in Lua:
http://wiki.freeswitch.org/wiki/Lua#Special_Case:_env_object

-MC

On Thu, Aug 11, 2011 at 10:06 AM, Alec Glassford <alec at efuse.co.uk> wrote:

> Hi,****
>
> ** **
>
> I need to run a LUA script after a Bridge to write call information to a
> MySQL DB.****
>
> ** **
>
> The problem is the script is only running intermittently, maybe 1 in 10
> calls.****
>
> ** **
>
> I have tried the exec_after_bridge_app command, I?ve tried
> uuid_bridge_continue_on_cancel, but it neither works.  I can provide output
> if necessary.****
>
> ** **
>
> I am bridging FreeTDM Sangoma A101 E1 ISDN30 calls to a SIP Provider, I do
> a DB lookup to find the correct Presentation number then bridge the call,
> below is from my Dialplan:****
>
> ** **
>
> <condition field="destination_number" expression="^(\d{11})$"> ****
>
>      <action application="lua" data="lookup02.lua"/> ****
>
>      <action application="set"
> data="effective_caller_id_number=${Presentation}"/>****
>
>      <action application="set"
> data="effective_caller_id_name=${Presentation}"/>****
>
>      <action application="log" data="INFO Dialling $1 presenting
> ${Presentation} from dblookup"/>****
>
>      <action application="set" data="uuid_bridge_continue_on_cancel=true"/>
> ****
>
>      <action application="set" data="exec_after_bridge_app=lua"/>****
>
>      <action application="set" data="exec_after_bridge_arg=endcall01.lua"/>
> ****
>
>      <action application="bridge" data="sofia/gateway/ProviderZZZ/$1" />**
> **
>
>      <action application="lua" data="endcall01.lua"/>****
>
>      <action application="hangup" />****
>
> </condition>****
>
> ** **
>
> Any help much appreciated****
>
> ** **
>
> ** **
>
> *Alec Glassford*****
>
> *efuse*****
>
> Tel:  0844 847 9707****
>
> Mob: 07540 417395****
>
> Fax: 0844 847 9708****
>
> www.efuse.co.uk****
>
> ** **
>
> This is an email from efuse, IT Solutions providers: www.efuse.co.uk****
>
>  ****
>
> P Save a tree...please don't print this e-mail* unless you really need to*
> ****
>
> Its contents are confidential and legally privileged and it is intended
> only for the use of the addressees named above.  If you are not an addressee
> you must not read it and must not use any information contained in it nor
> copy it nor inform any person other than Efuse Solutions or the addressees
> of its existence or contents.****
>
> If you have received this email and are not a named addressee please delete
> it and notify support at efuse.co.uk <helpdesk at efuse.co.uk>****
>
> Please note that Efuse Solutions nor the sender accepts any responsibility
> for viruses and that it is your responsibility to scan any attachments.  No
> contractual obligations may be established on behalf of Efuse Solutions by
> means of email communication.****
>
> ** **
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110815/abc312df/attachment.html 

From msc at freeswitch.org  Mon Aug 15 20:02:06 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 15 Aug 2011 11:02:06 -0500
Subject: [Freeswitch-users] DTMF issue when using execute_extension with
	play_and_get_digits
In-Reply-To: <CAA1Qw2sLHtu8ZHQn7GJFOs=f2k9H2pLa+mgu6Kt+FEtGcwWBOw@mail.gmail.com>
References: <CAA1Qw2sLHtu8ZHQn7GJFOs=f2k9H2pLa+mgu6Kt+FEtGcwWBOw@mail.gmail.com>
Message-ID: <CAKzWOxX9CEbvti8pQCRBgSipyW4y-Lx43RDQ0QTVoS2QwqbK9g@mail.gmail.com>

Were you able to resolve this yet?
-MC

On Tue, Aug 9, 2011 at 10:07 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:

> Hi Friends,
>      Facing an issue when using bind_meta_app and execute_extension(with
> play_and_get_digits) combined.
>
> Here is my dialplan,
> <condition field="destination_number" expression="^4567$" break="on-true">
> <action application="play_and_get_digits" data="1 5 3 7000 #
> '/voice_files/A.wav' '/voice_files/failed.wav' dtmf_received \d+"/>
> </condition>
> <condition field="destination_number" expression="^.*$" break="on-true">
> <action application="answer"/>
> <action application="bind_meta_app" data="5 b o execute_extension::4567 XML
> default"/>
> <action application="bridge" data="freetdm/1/a/9952248266"/>
> <action application="park"/>
> </condition>
>
> So, when callee enters *5,  I want the caller to enter a number. I get the
> extension executed as expected. The caller is able to hear the voice file
> played and when he enters the digits, it is not received. Digits are not
> even present in FS log.
>
> In the normal cases, there is no issues in getting DTMFs. I don't know,
> what am I doing wrong here. Kindly, help me to resolve this.
>
> --
> Regards,
> Nagalenoj H.
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110815/8d2e7146/attachment-0001.html 

From larclap at yahoo.com  Mon Aug 15 20:23:57 2011
From: larclap at yahoo.com (Lars Zeb)
Date: Mon, 15 Aug 2011 09:23:57 -0700
Subject: [Freeswitch-users] Help with dual IP gateways
Message-ID: <005501cc5b67$be0ace00$3a206a00$@yahoo.com>

Currently my LAN is connected to the internet via DSL. The FreeSWITCH box is
on this subnet. To save money, I am moving the data portion of my LAN to a
new ISP and I want to segregate the VOIP to another ISP. I am tired of
having a bad VOIP connection during lengthy downloads.

My VOIP and FreeSWITCH skills are minimal. I have used FreeSWITCH for over a
year in a home/business environment. The only reason it is working is with
the help of this list.

My knowledge of IP is similar. I do not know how to setup a LAN with two
gateways with all nodes seeing one another. I do want to be able to call out
via FreeSWITCH from a softphone on the data portion of the new LAN. 

A friend suggested I need a dual ported WAN firewall/router with load
balancing to enable all the nodes to be on the same subnet. Can anyone help
me with suggestions? Is there a consultant I can hire to help with this?

Thanks, Lars



From anthony.minessale at gmail.com  Mon Aug 15 20:30:53 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Aug 2011 11:30:53 -0500
Subject: [Freeswitch-users] FS performance using ESL
In-Reply-To: <CAJoKKdXR64qNfCYj0uyYFda4YLavwd_NFn3bH6Td+2eSGo7Y1w@mail.gmail.com>
References: <CAJoKKdU9VPzOS-C3jHCjG6Y0V-Jo_j7hqYwt6-Zh7ohb1pq3xQ@mail.gmail.com>
	<CAKbxfG-SWBzu0dBv_5B0v7710q7vLeeduT9L5V_cnceYHjrGJQ@mail.gmail.com>
	<CAJoKKdUcA=9CvSdRPhaoMopK+f4K2W1DO0=0b5mck_KEjOc9pQ@mail.gmail.com>
	<CAJoKKdXR64qNfCYj0uyYFda4YLavwd_NFn3bH6Td+2eSGo7Y1w@mail.gmail.com>
Message-ID: <CAKbxfG_SnLS2c6CnepKcy=ea7CY=JnM5Lyy_Rv8ahUYpqJdooA@mail.gmail.com>

You must have something setup strangely cos it would definitely reduce
your overall cps to use ESL but not down to 2 CPS.

Did you look over the server stats like top etc and look for any
misconfiguration?


On Thu, Aug 11, 2011 at 6:56 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:
> is there any other method than esl to controll calls on FS from an eternal
> application?
> will mod_curl or mod_xml_curl get better performance?
>
> T.
>
> On Fri, Aug 12, 2011 at 1:33 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:
>>
>> Hi Anthony, thanks for your response ...
>>
>>
>> this is what i have:
>>
>> ??????? esl_filter(&handle, "unique-id",
>> esl_event_get_header(handle.info_event, "caller-unique-id"));
>> ??????? esl_events(&handle, ESL_EVENT_TYPE_PLAIN, "CHANNEL_DATA
>> CHANNEL_EXECUTE_COMPLETE CHANNEL_HANGUP");
>>
>> what do you suggest i put there ?
>>
>>
>> is the inbound method less costly ?
>>
>>
>>
>>
>> I modified testserver.c just a bit...
>>
>> #include <sys/types.h>? /* include this before any other sys headers */
>> #include <sys/wait.h>?? /* header for waitpid() and various macros */
>> #include <signal.h>???? /* header for signal functions */
>> #include <stdio.h>????? /* header for fprintf() */
>> #include <unistd.h>???? /* header for fork() */
>> #include <stdlib.h>
>> #include <esl.h>
>>
>> void sig_chld(int);???? /* prototype for our SIGCHLD handler */
>>
>> static void mycallback(esl_socket_t server_sock, esl_socket_t client_sock,
>> struct sockaddr_in *addr)
>> {
>> ??????? esl_handle_t handle = {{0}};
>> ??????? int done = 0;
>> ??????? esl_status_t status;
>> ??????? time_t exp = 0;
>>
>> ??????? if (fork() != 0) {
>> ??????????????? close(client_sock);
>> ??????????????? return;
>> ??????? }
>>
>> ??????? esl_attach_handle(&handle, client_sock, addr);
>>
>> ??????? esl_log(ESL_LOG_INFO, "Connected! %d\n", handle.sock);
>>
>> ??????? esl_filter(&handle, "unique-id",
>> esl_event_get_header(handle.info_event, "caller-unique-id"));
>> ??????? esl_events(&handle, ESL_EVENT_TYPE_PLAIN, "CHANNEL_DATA
>> CHANNEL_EXECUTE_COMPLETE CHANNEL_HANGUP");
>>
>> ??????? esl_send_recv(&handle, "linger");
>>
>> ??????? esl_execute(&handle, "answer", NULL, NULL);
>> ??????? //esl_execute(&handle, "conference", "3000 at default", NULL);
>> ??????? esl_execute(&handle, "playback", "/home/tculjaga/myWavFile.wav",
>> NULL);
>> ??????? //esl_execute(&handle, "sleep", "1000", NULL);
>> ??????? //esl_execute(&handle, "hangup", NULL, NULL);
>>
>> ??????? while((status = esl_recv_timed(&handle, 1000)) != ESL_FAIL) {
>> ??????????????? if (done) {
>> ??????????????????????? if (time(NULL) >= exp) {
>> ??????????????????????????????? break;
>> ??????????????????????? }
>> ??????????????? } else if (status == ESL_SUCCESS) {
>> ??????????????????????? const char *type =
>> esl_event_get_header(handle.last_event, "content-type");
>> ??????????????????????? if (type && !strcasecmp(type,
>> "text/disconnect-notice")) {
>> ??????????????????????????????? const char *dispo =
>> esl_event_get_header(handle.last_event, "content-disposition");
>> ??????????????????????????????? esl_log(ESL_LOG_INFO, "Got a disconnection
>> notice dispostion: [%s]\n", dispo ? dispo : "");
>> ??????????????????????????????? if (!strcmp(dispo, "linger")) {
>> ??????????????????????????????????????? done = 1;
>> ??????????????????????????????????????? esl_log(ESL_LOG_INFO, "Waiting 5
>> seconds for any remaining events.\n");
>> ??????????????????????????????????????? exp = time(NULL) + 5;
>> ??????????????????????????????? }
>> ??????????????????????? }
>> ??????????????? }
>> ??????? }
>>
>> ??????? esl_log(ESL_LOG_INFO, "Disconnected! %d\n", handle.sock);
>> ??????? esl_disconnect(&handle);
>>
>> ??????? close(client_sock);
>>
>> ??????? _exit(0);
>> }
>>
>> /*
>> ?* The signal handler function -- only gets called when a SIGCHLD
>> ?* is received, ie when a child terminates
>> ?*/
>> void sig_chld(int signo)
>> {
>> ??? int status;
>>
>> ??? /* Wait for any child without blocking */
>> ??? if (waitpid(-1, &status, WNOHANG) < 0)
>> ??? {
>> ??????? /*
>> ???????? * calling standard I/O functions like fprintf() in a
>> ???????? * signal handler is not recommended, but probably OK
>> ???????? * in toy programs like this one.
>> ???????? */
>> ??????? fprintf(stderr, "waitpid failed\n");
>> ??????? return;
>> ??? }
>> }
>>
>> int main(void)
>> {
>> ??????? struct sigaction act;
>>
>> ??????? /* Assign sig_chld as our SIGCHLD handler */
>> ??????? act.sa_handler = sig_chld;
>>
>> ??????? /* We don't want to block any other signals in this example */
>> ??????? sigemptyset(&act.sa_mask);
>>
>> ??????? /*
>> ???????? * We're only interested in children that have terminated, not
>> ones
>> ???????? * which have been stopped (eg user pressing control-Z at
>> terminal)
>> ???????? */
>> ??????? act.sa_flags = SA_NOCLDSTOP;
>>
>> ??????? /*
>> ???????? * Make these values effective. If we were writing a real
>> ???????? * application, we would probably save the old value instead of
>> ???????? * passing NULL.
>> ???????? */
>> /*????? if (sigaction(SIGCHLD, &act, NULL) < 0)
>> ??????? {
>> ??????????????? fprintf(stderr, "sigaction failed\n");
>> ??????????????? return 1;
>> ??????? }
>> */
>> ??????? signal(SIGCHLD, SIG_IGN);
>>
>> ??????? esl_global_set_default_logger(0);
>> ??????? esl_listen("localhost", 8088, mycallback);
>>
>> ??????? return 0;
>> }
>>
>>
>>
>>
>> On Thu, Aug 11, 2011 at 9:59 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>>>
>>> try removing the filter and event subscriptions
>>> it's costly to consume all of the events especially at 75cps.
>>>
>>>
>>> On Thu, Aug 11, 2011 at 5:23 AM, Tihomir Culjaga <tculjaga at gmail.com>
>>> wrote:
>>> > hello,
>>> >
>>> > im wondering how much performance do we loose when using ESL instead of
>>> > running it via dialplan?
>>> >
>>> >
>>> > without ESL with a fine tuned FS and a short dialplan ( answer,
>>> > playback
>>> > like 20 seconds file, hangup ) im able to service 75 CPS. On the same
>>> > FS,
>>> > when i use ESL to answer the call, playback the same file and hangup,
>>> > im not
>>> > able to run more than 2 CPS... this is a huge impact and i really can't
>>> > believe it.
>>> >
>>> > I'm using event-socket outbound e.g.:
>>> >
>>> > <action application="socket" data="127.0.0.1:8088 async full"/>
>>> >
>>> > my extension looks like:
>>> >
>>> > <extension name="ESL_C">
>>> > ? <condition field="destination_number" expression="^(6666)$">
>>> > ??? <action application="socket" data="127.0.0.1:8088 async full"/>
>>> > ??? <action application="sleep" data="1000"/>
>>> > ??? <action application="hangup"/>
>>> > ? </condition>
>>> > </extension>
>>> >
>>> >
>>> > im using testserver from lib/esl/ and i just removed the conference
>>> > command
>>> > and added the playback one.... also i moved the esl_debug lvl to 0
>>> >
>>> >
>>> > anyhow, FS cannot run more than 2 CPS compared to 75 CPS when the
>>> > playback
>>> > is done from the dialplan.
>>> >
>>> >
>>> > Please, can someone give me a clue on what is going on?
>>> > Maybe im doing something wrong?
>>> > how to get maximum FS performance using ESL ?
>>> >
>>> >
>>> >
>>> > Regards,
>>> > Tihomir.
>>> >
>>> >
>>> > _______________________________________________
>>> > Join us at ClueCon 2011, Aug 9-11, Chicago
>>> > http://www.cluecon.com 877-7-4ACLUE
>>> >
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> 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
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From anthony.minessale at gmail.com  Mon Aug 15 20:43:39 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Aug 2011 11:43:39 -0500
Subject: [Freeswitch-users] FS performance using ESL
In-Reply-To: <CAKbxfG_SnLS2c6CnepKcy=ea7CY=JnM5Lyy_Rv8ahUYpqJdooA@mail.gmail.com>
References: <CAJoKKdU9VPzOS-C3jHCjG6Y0V-Jo_j7hqYwt6-Zh7ohb1pq3xQ@mail.gmail.com>
	<CAKbxfG-SWBzu0dBv_5B0v7710q7vLeeduT9L5V_cnceYHjrGJQ@mail.gmail.com>
	<CAJoKKdUcA=9CvSdRPhaoMopK+f4K2W1DO0=0b5mck_KEjOc9pQ@mail.gmail.com>
	<CAJoKKdXR64qNfCYj0uyYFda4YLavwd_NFn3bH6Td+2eSGo7Y1w@mail.gmail.com>
	<CAKbxfG_SnLS2c6CnepKcy=ea7CY=JnM5Lyy_Rv8ahUYpqJdooA@mail.gmail.com>
Message-ID: <CAKbxfG-yRM_rOr-VgogZMcgU55Dv9QK1_WkLM-M8rkoFQ6WcTg@mail.gmail.com>

my guess is you need to add the following line right after the
connected info line:

esl_log(ESL_LOG_INFO, "Connected! %d\n", handle.sock);

handle.event_lock = 1; // this keeps the hangup from being queued
right after the playback and ending the call



On Mon, Aug 15, 2011 at 11:30 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> You must have something setup strangely cos it would definitely reduce
> your overall cps to use ESL but not down to 2 CPS.
>
> Did you look over the server stats like top etc and look for any
> misconfiguration?
>
>
> On Thu, Aug 11, 2011 at 6:56 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:
>> is there any other method than esl to controll calls on FS from an eternal
>> application?
>> will mod_curl or mod_xml_curl get better performance?
>>
>> T.
>>
>> On Fri, Aug 12, 2011 at 1:33 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:
>>>
>>> Hi Anthony, thanks for your response ...
>>>
>>>
>>> this is what i have:
>>>
>>> ??????? esl_filter(&handle, "unique-id",
>>> esl_event_get_header(handle.info_event, "caller-unique-id"));
>>> ??????? esl_events(&handle, ESL_EVENT_TYPE_PLAIN, "CHANNEL_DATA
>>> CHANNEL_EXECUTE_COMPLETE CHANNEL_HANGUP");
>>>
>>> what do you suggest i put there ?
>>>
>>>
>>> is the inbound method less costly ?
>>>
>>>
>>>
>>>
>>> I modified testserver.c just a bit...
>>>
>>> #include <sys/types.h>? /* include this before any other sys headers */
>>> #include <sys/wait.h>?? /* header for waitpid() and various macros */
>>> #include <signal.h>???? /* header for signal functions */
>>> #include <stdio.h>????? /* header for fprintf() */
>>> #include <unistd.h>???? /* header for fork() */
>>> #include <stdlib.h>
>>> #include <esl.h>
>>>
>>> void sig_chld(int);???? /* prototype for our SIGCHLD handler */
>>>
>>> static void mycallback(esl_socket_t server_sock, esl_socket_t client_sock,
>>> struct sockaddr_in *addr)
>>> {
>>> ??????? esl_handle_t handle = {{0}};
>>> ??????? int done = 0;
>>> ??????? esl_status_t status;
>>> ??????? time_t exp = 0;
>>>
>>> ??????? if (fork() != 0) {
>>> ??????????????? close(client_sock);
>>> ??????????????? return;
>>> ??????? }
>>>
>>> ??????? esl_attach_handle(&handle, client_sock, addr);
>>>
>>> ??????? esl_log(ESL_LOG_INFO, "Connected! %d\n", handle.sock);
>>>
>>> ??????? esl_filter(&handle, "unique-id",
>>> esl_event_get_header(handle.info_event, "caller-unique-id"));
>>> ??????? esl_events(&handle, ESL_EVENT_TYPE_PLAIN, "CHANNEL_DATA
>>> CHANNEL_EXECUTE_COMPLETE CHANNEL_HANGUP");
>>>
>>> ??????? esl_send_recv(&handle, "linger");
>>>
>>> ??????? esl_execute(&handle, "answer", NULL, NULL);
>>> ??????? //esl_execute(&handle, "conference", "3000 at default", NULL);
>>> ??????? esl_execute(&handle, "playback", "/home/tculjaga/myWavFile.wav",
>>> NULL);
>>> ??????? //esl_execute(&handle, "sleep", "1000", NULL);
>>> ??????? //esl_execute(&handle, "hangup", NULL, NULL);
>>>
>>> ??????? while((status = esl_recv_timed(&handle, 1000)) != ESL_FAIL) {
>>> ??????????????? if (done) {
>>> ??????????????????????? if (time(NULL) >= exp) {
>>> ??????????????????????????????? break;
>>> ??????????????????????? }
>>> ??????????????? } else if (status == ESL_SUCCESS) {
>>> ??????????????????????? const char *type =
>>> esl_event_get_header(handle.last_event, "content-type");
>>> ??????????????????????? if (type && !strcasecmp(type,
>>> "text/disconnect-notice")) {
>>> ??????????????????????????????? const char *dispo =
>>> esl_event_get_header(handle.last_event, "content-disposition");
>>> ??????????????????????????????? esl_log(ESL_LOG_INFO, "Got a disconnection
>>> notice dispostion: [%s]\n", dispo ? dispo : "");
>>> ??????????????????????????????? if (!strcmp(dispo, "linger")) {
>>> ??????????????????????????????????????? done = 1;
>>> ??????????????????????????????????????? esl_log(ESL_LOG_INFO, "Waiting 5
>>> seconds for any remaining events.\n");
>>> ??????????????????????????????????????? exp = time(NULL) + 5;
>>> ??????????????????????????????? }
>>> ??????????????????????? }
>>> ??????????????? }
>>> ??????? }
>>>
>>> ??????? esl_log(ESL_LOG_INFO, "Disconnected! %d\n", handle.sock);
>>> ??????? esl_disconnect(&handle);
>>>
>>> ??????? close(client_sock);
>>>
>>> ??????? _exit(0);
>>> }
>>>
>>> /*
>>> ?* The signal handler function -- only gets called when a SIGCHLD
>>> ?* is received, ie when a child terminates
>>> ?*/
>>> void sig_chld(int signo)
>>> {
>>> ??? int status;
>>>
>>> ??? /* Wait for any child without blocking */
>>> ??? if (waitpid(-1, &status, WNOHANG) < 0)
>>> ??? {
>>> ??????? /*
>>> ???????? * calling standard I/O functions like fprintf() in a
>>> ???????? * signal handler is not recommended, but probably OK
>>> ???????? * in toy programs like this one.
>>> ???????? */
>>> ??????? fprintf(stderr, "waitpid failed\n");
>>> ??????? return;
>>> ??? }
>>> }
>>>
>>> int main(void)
>>> {
>>> ??????? struct sigaction act;
>>>
>>> ??????? /* Assign sig_chld as our SIGCHLD handler */
>>> ??????? act.sa_handler = sig_chld;
>>>
>>> ??????? /* We don't want to block any other signals in this example */
>>> ??????? sigemptyset(&act.sa_mask);
>>>
>>> ??????? /*
>>> ???????? * We're only interested in children that have terminated, not
>>> ones
>>> ???????? * which have been stopped (eg user pressing control-Z at
>>> terminal)
>>> ???????? */
>>> ??????? act.sa_flags = SA_NOCLDSTOP;
>>>
>>> ??????? /*
>>> ???????? * Make these values effective. If we were writing a real
>>> ???????? * application, we would probably save the old value instead of
>>> ???????? * passing NULL.
>>> ???????? */
>>> /*????? if (sigaction(SIGCHLD, &act, NULL) < 0)
>>> ??????? {
>>> ??????????????? fprintf(stderr, "sigaction failed\n");
>>> ??????????????? return 1;
>>> ??????? }
>>> */
>>> ??????? signal(SIGCHLD, SIG_IGN);
>>>
>>> ??????? esl_global_set_default_logger(0);
>>> ??????? esl_listen("localhost", 8088, mycallback);
>>>
>>> ??????? return 0;
>>> }
>>>
>>>
>>>
>>>
>>> On Thu, Aug 11, 2011 at 9:59 PM, Anthony Minessale
>>> <anthony.minessale at gmail.com> wrote:
>>>>
>>>> try removing the filter and event subscriptions
>>>> it's costly to consume all of the events especially at 75cps.
>>>>
>>>>
>>>> On Thu, Aug 11, 2011 at 5:23 AM, Tihomir Culjaga <tculjaga at gmail.com>
>>>> wrote:
>>>> > hello,
>>>> >
>>>> > im wondering how much performance do we loose when using ESL instead of
>>>> > running it via dialplan?
>>>> >
>>>> >
>>>> > without ESL with a fine tuned FS and a short dialplan ( answer,
>>>> > playback
>>>> > like 20 seconds file, hangup ) im able to service 75 CPS. On the same
>>>> > FS,
>>>> > when i use ESL to answer the call, playback the same file and hangup,
>>>> > im not
>>>> > able to run more than 2 CPS... this is a huge impact and i really can't
>>>> > believe it.
>>>> >
>>>> > I'm using event-socket outbound e.g.:
>>>> >
>>>> > <action application="socket" data="127.0.0.1:8088 async full"/>
>>>> >
>>>> > my extension looks like:
>>>> >
>>>> > <extension name="ESL_C">
>>>> > ? <condition field="destination_number" expression="^(6666)$">
>>>> > ??? <action application="socket" data="127.0.0.1:8088 async full"/>
>>>> > ??? <action application="sleep" data="1000"/>
>>>> > ??? <action application="hangup"/>
>>>> > ? </condition>
>>>> > </extension>
>>>> >
>>>> >
>>>> > im using testserver from lib/esl/ and i just removed the conference
>>>> > command
>>>> > and added the playback one.... also i moved the esl_debug lvl to 0
>>>> >
>>>> >
>>>> > anyhow, FS cannot run more than 2 CPS compared to 75 CPS when the
>>>> > playback
>>>> > is done from the dialplan.
>>>> >
>>>> >
>>>> > Please, can someone give me a clue on what is going on?
>>>> > Maybe im doing something wrong?
>>>> > how to get maximum FS performance using ESL ?
>>>> >
>>>> >
>>>> >
>>>> > Regards,
>>>> > Tihomir.
>>>> >
>>>> >
>>>> > _______________________________________________
>>>> > Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> > http://www.cluecon.com 877-7-4ACLUE
>>>> >
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >
>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>> >
>>>> >
>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> 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
>>>> googletalk:conf+888 at conference.freeswitch.org
>>>> pstn:+19193869900
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From msc at freeswitch.org  Mon Aug 15 21:11:17 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 15 Aug 2011 12:11:17 -0500
Subject: [Freeswitch-users] Conference calls through web-interface with
 moderation using FreeSWITCH?
In-Reply-To: <CAO+9iGdBYXQ2f8BFB+gSriP7bb_rKaFVVwaJzK2YmkfPWEAbdA@mail.gmail.com>
References: <CAO+9iGdBYXQ2f8BFB+gSriP7bb_rKaFVVwaJzK2YmkfPWEAbdA@mail.gmail.com>
Message-ID: <CAKzWOxUVyHoMjB+4Aw=H-cAvaHte8n5kHdhGZagnnw5TwVbgVw@mail.gmail.com>

Well, as the saying goes: "Some assembly required." There are hooks into FS
to let you build this. You will need to familiarize yourself with the event
socket as well as the conference API. If you prefer not to do the work
yourself and instead want professional assistance you can contact
consulting at freeswitch.org.

-MC

On Fri, Aug 12, 2011 at 10:55 AM, Alec Taylor <alec.taylor6 at gmail.com>wrote:

> Good Morning,
>
> I have been researching this for a while, basically I'd like to have a
> website with the following functionality:
> ? One-click call-in to show (after setting username, best-case
> scenario: sign-in through Drupal, use that name for conference-call)
> ? Web-interface only (Flash/Flex, Javascript/JQuery or Java), without
> any additional software/addons/plugins to install
> ? Moderation: host of conference call can quieten/mute or even kick
> people from the conference call if they're being rowdy
>
> So far I have setup an IceCAST server, broadcasting through edcast in
> an mp3 stream. Viewers of my website can now listen-in on the /radio/
> sub-page.
>
> How do I setup the aforementioned [3] features using FreeSWITCH? ? Do
> I need [Free, Open-Source] products other than FreeSWITCH to get this
> done, if so, which?
>
> Thanks for all suggestions,
>
> Alec Taylor
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110815/a46a498e/attachment.html 

From leonardo.bidinoto at voicetechnology.com.br  Mon Aug 15 22:21:01 2011
From: leonardo.bidinoto at voicetechnology.com.br (Leonardo P. Bidinoto)
Date: Mon, 15 Aug 2011 15:21:01 -0300
Subject: [Freeswitch-users] Mod event socket error
Message-ID: <CALV5c5NZXjzyy6e4kDv7+5X1gZHgwaXNRx_PY5r63Mu7Ts=6DQ@mail.gmail.com>

Hi Guys,

Im getting a lot of errors in my test machines, like these:

2011-08-15 10:12:39.433325 [CRIT] mod_event_socket.c:378 Lost 244 events!
2011-08-15 10:12:47.689947 [CRIT] mod_event_socket.c:378 Lost 32 events!
2011-08-15 10:12:48.770116 [CRIT] mod_event_socket.c:378 Lost 3 events!
2011-08-15 10:12:50.771600 [CRIT] mod_event_socket.c:378 Lost 54 events!
2011-08-15 10:12:50.791728 [CRIT] mod_event_socket.c:378 Lost 9 events!
2011-08-15 10:12:51.012380 [CRIT] mod_event_socket.c:378 Lost 39 events!
2011-08-15 10:12:51.012380 [CRIT] mod_event_socket.c:378 Lost 1 events!

What this kind of error means?

-- 
Leonardo Pires Bidinoto
Voice Technology
www.voicetechnology.com.br
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110815/f32c60fe/attachment.html 

From xing2kin at yahoo.com  Mon Aug 15 22:48:32 2011
From: xing2kin at yahoo.com (king2kin)
Date: Mon, 15 Aug 2011 11:48:32 -0700 (PDT)
Subject: [Freeswitch-users] Weird: Sofia-sip failed to be compiled on Visual
	Studio 2008 but it was OK three weeks ago
In-Reply-To: <CALV5c5NZXjzyy6e4kDv7+5X1gZHgwaXNRx_PY5r63Mu7Ts=6DQ@mail.gmail.com>
Message-ID: <1313434112.88678.YahooMailClassic@web39707.mail.mud.yahoo.com>

It's weird that the version of FreeSwitch (1.0 head ?) was successfully compiled on Visual Studio 2008? (standard edition) on windows 2003 server three weeks ago; however,??now it always fail to compile all the modules and libs related to sofia-sip.? it seems ok to compiler other projects in the solution of FreeSwitch.
?
Since VS2008 seems to download something from the Internet while FreeSwitch is being compiled, I am afraid that any latest codes?break the project. How do I fix this compiler error on sofia-sip?
?
for example, here is the log while compiling [libsofia_sip_ua_static]:
?
---------------------------------
?
1>------ Rebuild All started: Project: libsofia_sip_ua_static, Configuration: Debug Win32 ------
1>Deleting intermediate and output files for project 'libsofia_sip_ua_static', configuration 'Debug|Win32'
1>Performing Pre-Build Event...
1>multipart mismatch with Recursive multipart ()
1>../libsofia-sip-ua/sip/sip_bad_mask: unknown header ""
1>../libsofia-sip-ua/sip/sip_bad_mask: unknown header ""
1>../libsofia-sip-ua/sip/sip_bad_mask: unknown header ""
1>../libsofia-sip-ua/sip/sip_bad_mask: unknown header ""
1>../libsofia-sip-ua/sip/sip_bad_mask: unknown header ""
1>../libsofia-sip-ua/sip/sip_bad_mask: unknown header ""
1>../libsofia-sip-ua/sip/sip_bad_mask: unknown header ""
1>../libsofia-sip-ua/sip/sip_bad_mask: unknown header ""
1>../libsofia-sip-ua/sip/sip_bad_mask: unknown header ""
1>../libsofia-sip-ua/sip/sip_bad_mask: unknown header ""
1>../libsofia-sip-ua/sip/sip_bad_mask: unknown header ""
1>../libsofia-sip-ua/sip/sip_bad_mask: unknown header ""
1>../libsofia-sip-ua/sip/sip_bad_mask: unknown header ""
1>../libsofia-sip-ua/sip/sip_bad_mask: unknown header ""
1>NOTE:
1>NOTE: Remember to install pthreadVC2.dll to your path, too!
1>NOTE:
1>Compiling...
1>inet_pton.c
1>smoothsort.c
1>string0.c
1>su.c
1>su_addrinfo.c
1>su_alloc.c
1>su_alloc_lock.c
1>su_base_port.c
1>su_bm.c
1>su_default_log.c
1>su_errno.c
1>su_global_log.c
1>su_localinfo.c
1>su_log.c
1>su_md5.c
1>su_os_nw.c
1>su_port.c
1>su_pthread_port.c
1>su_root.c
1>su_socket_port.c
1>Generating Code...
1>Compiling...
1>su_sprintf.c
1>su_strdup.c
1>su_string.c
1>su_strlst.c
1>su_tag.c
1>su_tag_io.c
1>su_taglist.c
1>su_time.c
1>su_time0.c
1>su_timer.c
1>su_uniqueid.c
1>su_vector.c
1>su_wait.c
1>su_win32_port.c
1>base64.c
1>rc4.c
1>token64.c
1>url.c
1>url_tag.c
1>url_tag_ref.c
1>Generating Code...
1>Compiling...
1>features.c
1>bnf.c
1>msg.c
1>msg_auth.c
1>msg_basic.c
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : error C2065: 'msg_error_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : error C2065: 'msg_error_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : error C2065: 'msg_error_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(82) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(129) : error C2065: 'msg_unknown_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(129) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(129) : error C2065: 'msg_unknown_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(129) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(129) : error C2065: 'msg_unknown_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(129) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(129) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(129) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(129) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(129) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(129) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(209) : error C2065: 'msg_payload_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(209) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(209) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(209) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(209) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(209) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(209) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(209) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(209) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(322) : error C2065: 'msg_separator_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(322) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(322) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(322) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(322) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(322) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(322) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(322) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_basic.c(322) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>msg_date.c
1>msg_generic.c
1>msg_header_copy.c
1>msg_header_make.c
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_header_make.c(71) : error C2065: 'msg_payload_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_header_make.c(72) : error C2065: 'msg_separator_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_header_make.c(73) : error C2065: 'msg_error_hash' : undeclared identifier
1>msg_mclass.c
1>msg_mime.c
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(246) : error C2065: 'msg_multipart_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(246) : warning C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(246) : warning C4024: 'msg_header_alloc' : different types for formal and actual parameter 2
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(250) : warning C4013: 'msg_content_type_make' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(250) : warning C4047: '=' : 'msg_content_type_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(387) : error C2065: 'msg_multipart_mclass' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(387) : warning C4047: '=' : 'const msg_mclass_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(435) : error C2065: 'msg_multipart_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(435) : warning C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(435) : warning C4024: 'msg_header_alloc' : different types for formal and actual parameter 2
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(457) : error C2065: 'msg_payload_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(457) : warning C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(457) : warning C4024: 'msg_header_alloc' : different types for formal and actual parameter 2
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(641) : warning C4013: 'msg_payload_format' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(642) : warning C4047: '=' : 'msg_payload_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(661) : warning C4013: 'msg_separator_make' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(661) : warning C4047: '=' : 'msg_separator_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(836) : warning C4013: 'msg_is_multipart' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(838) : warning C4013: 'msg_payload_init' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1068) : error C2065: 'msg_accept_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1068) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1068) : error C2065: 'msg_accept_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1068) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1068) : error C2065: 'msg_accept_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1068) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1068) : error C2065: 'msg_accept_dup_xtra' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1068) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1068) : error C2065: 'msg_accept_dup_one' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1068) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1068) : error C2065: 'msg_accept_update' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1068) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1068) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'char [7]'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1068) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1068) : warning C4047: 'initializing' : 'msg_print_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1068) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1068) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'size_t'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1068) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1101) : warning C4013: 'msg_is_accept' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1284) : error C2065: 'msg_accept_charset_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1284) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1284) : error C2065: 'msg_accept_charset_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1284) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1284) : error C2065: 'msg_accept_charset_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1284) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1284) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1284) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1284) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [15]'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1284) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1284) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1284) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1293) : warning C4013: 'msg_is_accept_charset' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1335) : error C2065: 'msg_accept_encoding_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1335) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1335) : error C2065: 'msg_accept_encoding_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1335) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1335) : error C2065: 'msg_accept_encoding_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1335) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1335) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1335) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1335) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [16]'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1335) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1335) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1335) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1386) : error C2065: 'msg_accept_language_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1386) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1386) : error C2065: 'msg_accept_language_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1386) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1386) : error C2065: 'msg_accept_language_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1386) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1386) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1386) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1386) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [16]'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1386) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1386) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1386) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1395) : warning C4013: 'msg_is_accept_language' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1462) : error C2065: 'msg_content_disposition_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1462) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1462) : error C2065: 'msg_content_disposition_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1462) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1462) : error C2065: 'msg_content_disposition_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1462) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1462) : error C2065: 'msg_content_disposition_dup_xtra' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1462) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1462) : error C2065: 'msg_content_disposition_dup_one' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1462) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1462) : error C2065: 'msg_content_disposition_update' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1462) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1462) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'char [20]'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1462) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1462) : warning C4047: 'initializing' : 'msg_print_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1462) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1462) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'size_t'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1462) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1483) : warning C4013: 'msg_is_content_disposition' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1573) : error C2065: 'msg_content_encoding_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1573) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1573) : error C2065: 'msg_content_encoding_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1573) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1573) : error C2065: 'msg_content_encoding_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1573) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1573) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1573) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1573) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [17]'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1573) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1573) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [2]'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1573) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1583) : warning C4013: 'msg_is_content_encoding' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1628) : error C2065: 'msg_content_language_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1628) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1628) : error C2065: 'msg_content_language_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1628) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1628) : error C2065: 'msg_content_language_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1628) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1628) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1628) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1628) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [17]'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1628) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1628) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1628) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1638) : warning C4013: 'msg_is_content_language' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1678) : error C2065: 'msg_content_length_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1678) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1678) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1678) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1678) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1678) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1678) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1678) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [15]'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1678) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1678) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [2]'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1742) : error C2065: 'msg_content_md5_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1742) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1742) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1742) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1742) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1742) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1742) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1742) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [12]'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1742) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1742) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1779) : error C2065: 'msg_content_id_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1779) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1779) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1779) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1779) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1779) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1779) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1779) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [11]'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1779) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1779) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1823) : error C2065: 'msg_content_type_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1823) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1823) : error C2065: 'msg_content_type_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1823) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1823) : error C2065: 'msg_content_type_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1823) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1823) : error C2065: 'msg_content_type_dup_xtra' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1823) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1823) : error C2065: 'msg_content_type_dup_one' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1823) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1823) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'void *'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1823) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char [13]'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1823) : warning C4047: 'initializing' : 'msg_print_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1823) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [2]'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1823) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1823) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'size_t'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1823) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1850) : warning C4013: 'msg_is_content_type' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1921) : error C2065: 'msg_mime_version_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1921) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1921) : error C2065: 'msg_mime_version_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1921) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1921) : error C2065: 'msg_mime_version_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1921) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1921) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1921) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1921) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [13]'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1921) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1921) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1921) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1930) : warning C4013: 'msg_is_mime_version' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1962) : error C2065: 'msg_content_location_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1962) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1962) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1962) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1962) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1962) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1962) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1962) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [17]'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1962) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(1962) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2029) : error C2065: 'msg_content_transfer_encoding_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2029) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2029) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2029) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2029) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2029) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2029) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2029) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [26]'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2029) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2029) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2153) : error C2065: 'msg_warning_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2153) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2153) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2153) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2153) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2153) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2153) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2153) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [8]'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2153) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime.c(2153) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>msg_mime_table.c
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime_table.c(6) : error C2143: syntax error : missing '{' before '<cv-qualifer>'
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_mime_table.c(6) : error C2059: syntax error : '<Unknown>'
1>msg_parser.c
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_parser.c(127) : error C2065: 'msg_request_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_parser.c(132) : error C2065: 'msg_status_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_parser.c(1051) : error C2065: 'msg_unknown_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\msg\msg_parser.c(2151) : warning C4013: 'msg_is_payload' undefined; assuming extern returning int
1>msg_parser_util.c
1>msg_tag.c
1>memcspn.c
1>memmem.c
1>memspn.c
1>strcasestr.c
1>strtoull.c
1>Generating Code...
1>Compiling...
1>sip_basic.c
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(121) : error C2065: 'sip_request_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(121) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(121) : error C2065: 'sip_request_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(121) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(121) : error C2065: 'sip_request_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(121) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(121) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(121) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'msg_dup_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(121) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(121) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(121) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(296) : error C2065: 'sip_status_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(296) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(296) : error C2065: 'sip_status_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(296) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(296) : error C2065: 'sip_status_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(296) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(296) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(296) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'msg_dup_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(296) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(296) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(296) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(321) : warning C4013: 'sip_is_status' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(434) : error C2065: 'sip_payload_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(434) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(434) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(434) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(434) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(434) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(434) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(434) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(434) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(459) : warning C4013: 'sip_header_data' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(459) : warning C4047: 'initializing' : 'char *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(508) : error C2065: 'sip_separator_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(508) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(508) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(508) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(508) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(508) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(508) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(508) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(508) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(560) : error C2065: 'sip_unknown_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(560) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(560) : error C2065: 'sip_unknown_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(560) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(560) : error C2065: 'sip_unknown_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(560) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(560) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(560) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(560) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(560) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(560) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(564) : warning C4013: 'msg_unknown_d' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(569) : warning C4013: 'msg_unknown_e' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(599) : error C2065: 'sip_error_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(599) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(599) : error C2065: 'sip_error_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(599) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(599) : error C2065: 'sip_error_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(599) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(599) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(599) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(599) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(599) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(599) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(962) : warning C4047: 'initializing' : 'char *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1046) : error C2065: 'sip_call_id_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1046) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1046) : error C2065: 'sip_call_id_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1046) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1046) : error C2065: 'sip_call_id_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1046) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1046) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1046) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'msg_dup_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1046) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [8]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1046) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1046) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [2]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1046) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1199) : error C2065: 'sip_cseq_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1199) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1199) : error C2065: 'sip_cseq_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1199) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1199) : error C2065: 'sip_cseq_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1199) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1199) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1199) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'msg_dup_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1199) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [5]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1199) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1199) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1199) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1224) : warning C4013: 'sip_is_cseq' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1374) : error C2065: 'sip_contact_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1374) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1375) : error C2065: 'sip_contact_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1375) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1376) : error C2065: 'sip_contact_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1376) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1377) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1378) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'msg_dup_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1380) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [8]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1381) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1382) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [2]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1383) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1414) : warning C4013: 'sip_is_contact' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1529) : error C2065: 'sip_content_length_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1529) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1529) : error C2065: 'sip_content_length_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1529) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1529) : error C2065: 'sip_content_length_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1529) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1529) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1529) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1529) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [15]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1529) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1529) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [2]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1529) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1546) : warning C4013: 'sip_is_content_length' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1618) : error C2065: 'sip_date_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1618) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1618) : error C2065: 'sip_date_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1618) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1618) : error C2065: 'sip_date_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1618) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1618) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1618) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1618) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [5]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1618) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1618) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1618) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1699) : error C2065: 'sip_expires_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1699) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1699) : error C2065: 'sip_expires_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1699) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1699) : error C2065: 'sip_expires_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1699) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1699) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1699) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1699) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [8]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1699) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1699) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1699) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1786) : error C2065: 'sip_from_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1786) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1786) : error C2065: 'sip_from_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1786) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1786) : error C2065: 'sip_from_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1786) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1786) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const sip_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1786) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(sip_header_t *,const sip_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1786) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [5]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1786) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1786) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [2]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1786) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1798) : warning C4013: 'sip_is_from' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1900) : error C2065: 'sip_max_forwards_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1900) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1900) : error C2065: 'sip_max_forwards_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1900) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1900) : error C2065: 'sip_max_forwards_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1900) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1900) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1900) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1900) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [13]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1900) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1900) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1900) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1909) : warning C4013: 'sip_is_max_forwards' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1946) : error C2065: 'sip_min_expires_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1946) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1946) : error C2065: 'sip_min_expires_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1946) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1946) : error C2065: 'sip_min_expires_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1946) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1946) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1946) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1946) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [12]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1946) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1946) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1946) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(1955) : warning C4013: 'sip_is_min_expires' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2004) : error C2065: 'sip_retry_after_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2004) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2004) : error C2065: 'sip_retry_after_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2004) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2004) : error C2065: 'sip_retry_after_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2004) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2004) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2004) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'msg_dup_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2004) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [12]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2004) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2004) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2004) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2196) : warning C4047: '=' : 'char *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2252) : error C2065: 'sip_route_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2252) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2252) : error C2065: 'sip_route_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2252) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2252) : error C2065: 'sip_route_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2252) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2252) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const sip_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2252) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(sip_header_t *,const sip_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2252) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [6]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2252) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2252) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2252) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2264) : warning C4013: 'sip_is_route' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2326) : error C2065: 'sip_record_route_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2326) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2326) : error C2065: 'sip_record_route_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2326) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2326) : error C2065: 'sip_record_route_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2326) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2326) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const sip_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2326) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(sip_header_t *,const sip_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2326) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [13]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2326) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2326) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2326) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2338) : warning C4013: 'sip_is_record_route' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2401) : error C2065: 'sip_to_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_basic.c(2401) : fatal error C1003: error count exceeds 100; stopping compilation
1>sip_caller_prefs.c
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(100) : error C2065: 'sip_request_disposition_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(100) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(100) : error C2065: 'sip_request_disposition_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(100) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(100) : error C2065: 'sip_request_disposition_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(100) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(100) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(100) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'msg_dup_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(100) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [20]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(100) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(100) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [2]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(100) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(116) : warning C4013: 'sip_is_request_disposition' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(325) : error C2065: 'sip_accept_contact_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(325) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(325) : error C2065: 'sip_accept_contact_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(325) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(325) : error C2065: 'sip_accept_contact_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(325) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(325) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const sip_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(325) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(sip_header_t *,const sip_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(325) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [15]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(325) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(325) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [2]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(325) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(416) : error C2065: 'sip_reject_contact_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(416) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(416) : error C2065: 'sip_reject_contact_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(416) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(416) : error C2065: 'sip_reject_contact_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(416) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(416) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const sip_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(416) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(sip_header_t *,const sip_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(416) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [15]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(416) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(416) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [2]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_caller_prefs.c(416) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>sip_event.c
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(97) : error C2065: 'sip_event_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(97) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(97) : error C2065: 'sip_event_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(97) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(97) : error C2065: 'sip_event_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(97) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(97) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(97) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'msg_dup_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(97) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [6]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(97) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(97) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [2]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(97) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(120) : warning C4013: 'sip_is_event' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(210) : error C2065: 'sip_allow_events_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(210) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(210) : error C2065: 'sip_allow_events_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(210) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(210) : error C2065: 'sip_allow_events_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(210) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(210) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(210) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(210) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [13]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(210) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(210) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [2]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(210) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(219) : warning C4013: 'sip_is_allow_events' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(300) : error C2065: 'sip_subscription_state_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(300) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(300) : error C2065: 'sip_subscription_state_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(300) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(300) : error C2065: 'sip_subscription_state_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(300) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(300) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(300) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'msg_dup_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(300) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [19]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(300) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(300) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(300) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_event.c(331) : warning C4013: 'sip_is_subscription_state' undefined; assuming extern returning int
1>sip_extra.c
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_extra.c(44) : fatal error C1083: Cannot open include file: 'sofia-sip/sip_extra.h': No such file or directory
1>sip_feature.c
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(93) : error C2065: 'sip_allow_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(93) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(93) : error C2065: 'sip_allow_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(93) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(93) : error C2065: 'sip_allow_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(93) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(93) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(93) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(93) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [6]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(93) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(93) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(93) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(105) : warning C4013: 'sip_is_allow' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(182) : error C2065: 'sip_proxy_require_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(182) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(182) : error C2065: 'sip_proxy_require_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(182) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(182) : error C2065: 'sip_proxy_require_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(182) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(182) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(182) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(182) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [14]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(182) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(182) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(182) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(192) : warning C4013: 'sip_is_proxy_require' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(230) : error C2065: 'sip_require_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(230) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(230) : error C2065: 'sip_require_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(230) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(230) : error C2065: 'sip_require_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(230) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(230) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(230) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(230) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [8]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(230) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(230) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(230) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(240) : warning C4013: 'sip_is_require' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(279) : error C2065: 'sip_supported_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(279) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(279) : error C2065: 'sip_supported_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(279) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(279) : error C2065: 'sip_supported_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(279) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(279) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(279) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(279) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [10]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(279) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(279) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [2]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(279) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(289) : warning C4013: 'sip_is_supported' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(326) : error C2065: 'sip_unsupported_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(326) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(326) : error C2065: 'sip_unsupported_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(326) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(326) : error C2065: 'sip_unsupported_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(326) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(326) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(326) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(326) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [12]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(326) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(326) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(326) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(336) : warning C4013: 'sip_is_unsupported' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(450) : warning C4013: 'sip_unsupported_make' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(450) : warning C4047: '=' : 'sip_unsupported_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(529) : error C2065: 'sip_path_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(529) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(529) : error C2065: 'sip_path_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(529) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(529) : error C2065: 'sip_path_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(529) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(529) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const sip_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(529) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(sip_header_t *,const sip_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(529) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [5]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(529) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(529) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(529) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(538) : warning C4013: 'sip_is_path' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(585) : error C2065: 'sip_service_route_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(585) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(585) : error C2065: 'sip_service_route_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(585) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(585) : error C2065: 'sip_service_route_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(585) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(585) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const sip_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(585) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(sip_header_t *,const sip_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(585) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [14]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(585) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(585) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(585) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_feature.c(594) : warning C4013: 'sip_is_service_route' undefined; assuming extern returning int
1>sip_header.c
1>sip_mime.c
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : error C2065: 'sip_accept_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : error C2065: 'sip_accept_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : error C2065: 'sip_accept_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : error C2065: 'msg_accept_dup_xtra' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : error C2065: 'msg_accept_dup_one' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : error C2065: 'msg_accept_update' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'char [7]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : warning C4047: 'initializing' : 'msg_print_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'size_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(101) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(105) : warning C4013: 'msg_accept_d' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(110) : warning C4013: 'msg_accept_e' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : error C2065: 'sip_accept_encoding_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : error C2065: 'sip_accept_encoding_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : error C2065: 'sip_accept_encoding_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : error C2065: 'msg_accept_any_update' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : warning C4047: 'initializing' : 'msg_print_f (__cdecl *)' differs in levels of indirection from 'char [16]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(216) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'size_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(220) : warning C4013: 'msg_accept_encoding_d' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(234) : warning C4013: 'msg_accept_encoding_e' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : error C2065: 'sip_accept_language_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : error C2065: 'sip_accept_language_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : error C2065: 'sip_accept_language_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : error C2065: 'msg_accept_any_update' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : warning C4047: 'initializing' : 'msg_print_f (__cdecl *)' differs in levels of indirection from 'char [16]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(282) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'size_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(286) : warning C4013: 'msg_accept_language_d' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(299) : warning C4013: 'msg_accept_language_e' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : error C2065: 'sip_content_disposition_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : error C2065: 'sip_content_disposition_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : error C2065: 'sip_content_disposition_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'msg_dup_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : error C2065: 'msg_content_disposition_update' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : warning C4047: 'initializing' : 'msg_print_f (__cdecl *)' differs in levels of indirection from 'char [20]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(371) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'size_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(376) : warning C4013: 'msg_content_disposition_d' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(382) : warning C4013: 'msg_content_disposition_e' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(388) : warning C4013: 'msg_content_disposition_dup_xtra' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(397) : warning C4013: 'msg_content_disposition_dup_one' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(397) : warning C4047: 'return' : 'char *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(437) : error C2065: 'sip_content_encoding_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(437) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(437) : error C2065: 'sip_content_encoding_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(437) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(437) : error C2065: 'sip_content_encoding_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(437) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(437) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(437) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(437) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [17]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(437) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(437) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [2]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(437) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(487) : error C2065: 'sip_content_language_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(487) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(487) : error C2065: 'sip_content_language_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(487) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(487) : error C2065: 'sip_content_language_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(487) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(487) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(487) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(487) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [17]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(487) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(487) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(487) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(553) : error C2065: 'sip_content_type_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(553) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(553) : error C2065: 'sip_content_type_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(553) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(553) : error C2065: 'sip_content_type_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(553) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(553) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(553) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'msg_dup_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(553) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [13]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(553) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(553) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [2]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(553) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(557) : warning C4013: 'msg_content_type_d' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(562) : warning C4013: 'msg_content_type_e' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(568) : warning C4013: 'msg_content_type_dup_xtra' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(576) : warning C4013: 'msg_content_type_dup_one' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(578) : warning C4047: 'return' : 'char *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(614) : error C2065: 'sip_mime_version_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(614) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(614) : error C2065: 'sip_mime_version_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(614) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(614) : error C2065: 'sip_mime_version_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(614) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(614) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(614) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(614) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [13]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(614) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(614) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(614) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : error C2065: 'sip_warning_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : error C2065: 'sip_warning_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : error C2065: 'sip_warning_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : error C2065: 'msg_warning_dup_xtra' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : error C2065: 'msg_warning_dup_one' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'void *'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char [8]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : warning C4047: 'initializing' : 'msg_print_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(673) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(677) : warning C4013: 'msg_warning_d' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_mime.c(681) : warning C4013: 'msg_warning_e' undefined; assuming extern returning int
1>sip_parser.c
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser.c(598) : error C2220: warning treated as error - no 'object' file generated
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser.c(598) : warning C4013: 'sip_object' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser.c(598) : warning C4047: 'initializing' : 'sip_t *' differs in levels of indirection from 'int'
1>sip_parser_table.c
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(5) : error C2143: syntax error : missing '{' before 'const'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(5) : warning C4431: missing type specifier - int assumed. Note: C no longer supports default-int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(5) : error C2065: 'MC_SHORT_SIZE' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(5) : error C2057: expected constant expression
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(5) : error C2466: cannot allocate an array of constant size 0
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(7) : error C2065: 'sip_accept_contact_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(7) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(7) : warning C4013: 'offsetof' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(7) : error C2065: 'sip_t' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(7) : error C2065: 'sip_accept_contact' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(7) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(8) : error C2065: 'sip_mask_pref' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(8) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(9) : error C2065: 'sip_referred_by_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(9) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(9) : error C2065: 'sip_t' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(9) : error C2065: 'sip_referred_by' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(9) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(10) : error C2065: 'sip_content_type_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(10) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(10) : error C2065: 'sip_t' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(10) : error C2065: 'sip_content_type' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(10) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(11) : error C2065: 'sip_mask_ua' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(11) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(12) : error C2065: 'sip_request_disposition_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(12) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(12) : error C2065: 'sip_t' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(12) : error C2065: 'sip_request_disposition' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(12) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(13) : error C2065: 'sip_mask_pref' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(13) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(14) : error C2065: 'sip_content_encoding_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(14) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(14) : error C2065: 'sip_t' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(14) : error C2065: 'sip_content_encoding' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(14) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(15) : error C2065: 'sip_mask_ua' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(15) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(16) : error C2065: 'sip_from_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(16) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(16) : error C2065: 'sip_t' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(16) : error C2065: 'sip_from' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(16) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(17) : error C2065: 'sip_mask_request' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(17) : error C2065: 'sip_mask_response' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(17) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(18) : error C2065: 'NULL' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(18) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(19) : error C2065: 'NULL' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(19) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(20) : error C2065: 'sip_call_id_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(20) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(20) : error C2065: 'sip_t' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(20) : error C2065: 'sip_call_id' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(20) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(21) : error C2065: 'sip_mask_request' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(21) : error C2065: 'sip_mask_response' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(21) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(22) : error C2065: 'sip_reject_contact_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(22) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(22) : error C2065: 'sip_t' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(22) : error C2065: 'sip_reject_contact' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(22) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(23) : error C2065: 'sip_mask_pref' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(23) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(24) : error C2065: 'sip_supported_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(24) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(24) : error C2065: 'sip_t' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(24) : error C2065: 'sip_supported' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(24) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(25) : error C2065: 'sip_content_length_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(25) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(25) : error C2065: 'sip_t' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(25) : error C2065: 'sip_content_length' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(25) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(26) : error C2065: 'sip_mask_request' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(26) : error C2065: 'sip_mask_response' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(26) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(27) : error C2065: 'sip_contact_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(27) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(27) : error C2065: 'sip_t' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(27) : error C2065: 'sip_contact' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(27) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(28) : error C2065: 'sip_mask_ua' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(28) : error C2065: 'sip_mask_proxy' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(28) : error C2065: 'sip_mask_registrar' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(28) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(29) : error C2065: 'NULL' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(29) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(30) : error C2065: 'sip_event_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(30) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(30) : error C2065: 'sip_t' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(30) : error C2065: 'sip_event' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(30) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(31) : error C2065: 'sip_mask_events' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(31) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(32) : error C2065: 'NULL' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(32) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(33) : error C2065: 'NULL' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(33) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(34) : error C2065: 'sip_refer_to_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(34) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_parser_table.c(34) : fatal error C1003: error count exceeds 100; stopping compilation
1>sip_prack.c
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(95) : error C2065: 'sip_rack_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(95) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(95) : error C2065: 'sip_rack_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(95) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(95) : error C2065: 'sip_rack_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(95) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(95) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(95) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'msg_dup_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(95) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [5]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(95) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(95) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(95) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(122) : warning C4013: 'sip_is_rack' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(193) : error C2065: 'sip_rseq_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(193) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(193) : error C2065: 'sip_rseq_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(193) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(193) : error C2065: 'sip_rseq_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(193) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(193) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(193) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(193) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [5]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(193) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(193) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(193) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_prack.c(202) : warning C4013: 'sip_is_rseq' undefined; assuming extern returning int
1>sip_pref_util.c
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_pref_util.c(529) : error C2220: warning treated as error - no 'object' file generated
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_pref_util.c(529) : warning C4013: 'sip_contact_copy' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_pref_util.c(529) : warning C4047: '=' : 'sip_contact_t *' differs in levels of indirection from 'int'
1>sip_reason.c
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_reason.c(95) : error C2065: 'sip_reason_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_reason.c(95) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_reason.c(95) : error C2065: 'sip_reason_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_reason.c(95) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_reason.c(95) : error C2065: 'sip_reason_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_reason.c(95) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_reason.c(95) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_reason.c(95) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'msg_dup_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_reason.c(95) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [7]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_reason.c(95) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_reason.c(95) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_reason.c(95) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_reason.c(120) : warning C4013: 'sip_is_reason' undefined; assuming extern returning int
1>sip_refer.c
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_refer.c(43) : fatal error C1083: Cannot open include file: 'sofia-sip/sip_extra.h': No such file or directory
1>sip_security.c
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(131) : error C2065: 'sip_authorization_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(131) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(131) : error C2065: 'sip_authorization_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(131) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(131) : error C2065: 'sip_authorization_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(131) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(131) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(131) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(131) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [14]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(131) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(131) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(131) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(140) : warning C4013: 'sip_is_authorization' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(201) : error C2065: 'sip_proxy_authenticate_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(201) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(201) : error C2065: 'sip_proxy_authenticate_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(201) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(201) : error C2065: 'sip_proxy_authenticate_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(201) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(201) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(201) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(201) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [19]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(201) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(201) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(201) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(211) : warning C4013: 'sip_is_proxy_authenticate' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(257) : error C2065: 'sip_proxy_authorization_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(257) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(257) : error C2065: 'sip_proxy_authorization_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(257) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(257) : error C2065: 'sip_proxy_authorization_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(257) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(257) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(257) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(257) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [20]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(257) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(257) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(257) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(267) : warning C4013: 'sip_is_proxy_authorization' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(312) : error C2065: 'sip_www_authenticate_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(312) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(312) : error C2065: 'sip_www_authenticate_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(312) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(312) : error C2065: 'sip_www_authenticate_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(312) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(312) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(312) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(312) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [17]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(312) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(312) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(312) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(321) : warning C4013: 'sip_is_www_authenticate' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(370) : error C2065: 'sip_authentication_info_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(370) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(370) : error C2065: 'sip_authentication_info_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(370) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(370) : error C2065: 'sip_authentication_info_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(370) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(370) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(370) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(370) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [20]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(370) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(370) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(370) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(380) : warning C4013: 'sip_is_authentication_info' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(436) : error C2065: 'sip_proxy_authentication_info_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(436) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(436) : error C2065: 'sip_proxy_authentication_info_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(436) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(436) : error C2065: 'sip_proxy_authentication_info_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(436) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(436) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(436) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(436) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [26]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(436) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(436) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(436) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(447) : warning C4013: 'sip_is_proxy_authentication_info' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(606) : error C2065: 'sip_security_client_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(606) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(606) : error C2065: 'sip_security_client_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(606) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(606) : error C2065: 'sip_security_client_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(606) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(606) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const sip_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(606) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(sip_header_t *,const sip_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(606) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [16]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(606) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(606) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(606) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(655) : error C2065: 'sip_security_server_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(655) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(655) : error C2065: 'sip_security_server_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(655) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(655) : error C2065: 'sip_security_server_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(655) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(655) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const sip_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(655) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(sip_header_t *,const sip_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(655) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [16]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(655) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(655) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(655) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(705) : error C2065: 'sip_security_verify_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(705) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(705) : error C2065: 'sip_security_verify_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(705) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(705) : error C2065: 'sip_security_verify_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(705) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(705) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'isize_t (__cdecl *)(const sip_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(705) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(sip_header_t *,const sip_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(705) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [16]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(705) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(705) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(705) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(756) : error C2065: 'sip_privacy_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(756) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(756) : error C2065: 'sip_privacy_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(756) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(756) : error C2065: 'sip_privacy_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(756) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(756) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(756) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'msg_dup_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(756) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [8]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(756) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(756) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_security.c(756) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>sip_session.c
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(93) : error C2065: 'sip_session_expires_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(93) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(93) : error C2065: 'sip_session_expires_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(93) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(93) : error C2065: 'sip_session_expires_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(93) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(93) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(93) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'msg_dup_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(93) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [16]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(93) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(93) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [2]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(93) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(201) : error C2065: 'sip_min_se_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(201) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(201) : error C2065: 'sip_min_se_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(201) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(201) : error C2065: 'sip_min_se_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(201) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(201) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(201) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'msg_dup_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(201) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [7]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(201) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(201) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_session.c(201) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'uintptr_t'
1>sip_status.c
1>sip_tag.c
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag.c(2) : error C2220: warning treated as error - no 'object' file generated
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag.c(2) : warning C4206: nonstandard extension used : translation unit is empty
1>sip_tag_class.c
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(219) : error C2220: warning treated as error - no 'object' file generated
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(219) : warning C4013: 'sip_object' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(219) : warning C4047: '=' : 'sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(224) : error C2065: 'siptag_end' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(224) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(232) : warning C4013: 'SIPTAG_P' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(247) : error C2065: 'siptag_header' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(247) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(253) : warning C4013: 'SIPTAG_STR_P' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(259) : error C2065: 'siptag_header_str' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(259) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(364) : error C2065: 'sip_payload_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(454) : error C2065: 'siptag_payload' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(454) : warning C4047: '=' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(454) : error C2065: 'sip_payload_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(454) : warning C4047: '=' : 'msg_hclass_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(456) : error C2065: 'sip_tag_list' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(456) : error C2109: subscript requires array or pointer type
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(457) : error C2065: 'sip_tag_list' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(457) : error C2109: subscript requires array or pointer type
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(483) : error C2065: 'siptag_header_str' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(483) : warning C4047: '=' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(493) : error C2065: 'siptag_header_str' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_tag_class.c(493) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
1>sip_tag_ref.c
1>sip_time.c
1>Generating Code...
1>Compiling...
1>sip_util.c
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_util.c(130) : error C2220: warning treated as error - no 'object' file generated
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_util.c(130) : warning C4013: 'sip_contact_make' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_util.c(130) : warning C4047: '=' : 'sip_contact_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_util.c(196) : warning C4047: 'initializing' : 'sip_contact_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_util.c(575) : warning C4013: 'sip_route_init' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_util.c(615) : error C2065: 'sip_route_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_util.c(615) : warning C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_util.c(615) : warning C4024: 'sip_route_reverse_as' : different types for formal and actual parameter 2
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_util.c(668) : error C2065: 'sip_route_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_util.c(668) : warning C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\sip\sip_util.c(668) : warning C4024: 'sip_route_fixdup_as' : different types for formal and actual parameter 2
1>http_basic.c
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(161) : error C2065: 'http_request_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(161) : warning C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(161) : warning C4024: 'msg_header_alloc' : different types for formal and actual parameter 2
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(181) : error C2065: 'http_request_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(181) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(181) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,http_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(181) : warning C4113: 'isize_t (__cdecl *)(const http_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(181) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const http_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(181) : warning C4113: 'char *(__cdecl *)(http_header_t *,const http_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(181) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(http_header_t *,const http_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(181) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(181) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(271) : error C2065: 'http_status_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(271) : warning C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(271) : warning C4024: 'msg_header_alloc' : different types for formal and actual parameter 2
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(281) : error C2065: 'http_status_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(281) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(281) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,http_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(281) : warning C4113: 'isize_t (__cdecl *)(const http_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(281) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const http_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(281) : warning C4113: 'char *(__cdecl *)(http_header_t *,const http_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(281) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(http_header_t *,const http_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(281) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(281) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(313) : error C2065: 'http_accept_ranges_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(313) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(313) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(313) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(313) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(313) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(313) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(313) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [14]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(313) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(313) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(323) : error C2065: 'http_age_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(323) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(323) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(323) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(323) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(323) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(323) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(323) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [4]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(323) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(323) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(331) : error C2065: 'http_allow_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(331) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(331) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(331) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(331) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(331) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(331) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(331) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [6]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(331) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(331) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(345) : error C2065: 'http_authentication_info_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(345) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(345) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(345) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(345) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(345) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(345) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(345) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [20]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(345) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(345) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(357) : error C2065: 'http_authorization_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(357) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(357) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(357) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(357) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(357) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(357) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(357) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [14]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(357) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(357) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(366) : error C2065: 'http_cache_control_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(366) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(366) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(366) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(366) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(366) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(366) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(366) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [14]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(366) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(366) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(374) : error C2065: 'http_connection_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(374) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(374) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(374) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(374) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(374) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(374) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(374) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [11]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(374) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(374) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(500) : error C2065: 'http_content_range_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(500) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(500) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,http_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(500) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(500) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(500) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(500) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(500) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [14]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(500) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(500) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(584) : error C2065: 'http_date_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(584) : warning C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(584) : warning C4024: 'msg_header_alloc' : different types for formal and actual parameter 2
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(597) : error C2065: 'http_date_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(597) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(597) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,http_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(597) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(597) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(597) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(597) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(597) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [5]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(597) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(597) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(607) : error C2065: 'http_etag_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(607) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(607) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(607) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(607) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(607) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(607) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(607) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [5]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(607) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(607) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(616) : error C2065: 'http_expect_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(616) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(616) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(616) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(616) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(616) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(616) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(616) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [7]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(616) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(616) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(652) : error C2065: 'http_expires_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(652) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(652) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,http_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(652) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(652) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(652) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(652) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(652) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [8]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(652) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(652) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(666) : error C2065: 'http_from_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(666) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(666) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(666) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(666) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(666) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(666) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(666) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [5]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(666) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(666) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(735) : warning C4013: 'http_host_init' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(740) : warning C4013: 'http_host_dup' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(740) : warning C4047: 'return' : 'http_host_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(747) : error C2065: 'http_host_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(747) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(747) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,http_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(747) : warning C4113: 'isize_t (__cdecl *)(const http_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(747) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const http_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(747) : warning C4113: 'char *(__cdecl *)(http_header_t *,const http_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(747) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(http_header_t *,const http_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(747) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [5]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(747) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(747) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(755) : error C2065: 'http_if_match_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(755) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(755) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(755) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(755) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(755) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(755) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(755) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [9]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(755) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(755) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(794) : error C2065: 'http_if_modified_since_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(794) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(794) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,http_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(794) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(794) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(794) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(794) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(794) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [18]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(794) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(794) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(802) : error C2065: 'http_if_none_match_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(802) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(802) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(802) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(802) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(802) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(802) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(802) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [14]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(802) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(802) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(873) : error C2065: 'http_if_range_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(873) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(873) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,http_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(873) : warning C4113: 'isize_t (__cdecl *)(const http_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(873) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const http_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(873) : warning C4113: 'char *(__cdecl *)(http_header_t *,const http_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(873) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(http_header_t *,const http_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(873) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [9]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(873) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(873) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(913) : error C2065: 'http_if_unmodified_since_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(913) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(913) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,http_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(913) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(913) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(913) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(913) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(913) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [20]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(913) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(913) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(948) : error C2065: 'http_last_modified_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(948) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(948) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,http_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(948) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(948) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(948) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(948) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(948) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [14]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(948) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(948) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1023) : error C2065: 'http_location_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1023) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1023) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1023) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1023) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1023) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1023) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1023) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [9]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1023) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1023) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1031) : error C2065: 'http_max_forwards_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1031) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1031) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1031) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1031) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1031) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1031) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1031) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [13]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1031) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1031) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1039) : error C2065: 'http_pragma_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1039) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1039) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1039) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1039) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1039) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1039) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1039) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [7]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1039) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1039) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1048) : error C2065: 'http_proxy_authenticate_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1048) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1048) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1048) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1048) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1048) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1048) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1048) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [19]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1048) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1048) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1057) : error C2065: 'http_proxy_authorization_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1057) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1057) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1057) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1057) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1057) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1057) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1057) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [20]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1057) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1057) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1210) : error C2065: 'http_range_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1210) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1210) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1210) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1210) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1210) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1210) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1210) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [6]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1210) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1210) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1248) : error C2065: 'http_referer_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1248) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1248) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1248) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1248) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1248) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1248) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1248) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [8]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1248) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1248) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1314) : error C2065: 'http_retry_after_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1314) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1314) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,http_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1314) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1314) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1314) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1314) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1314) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [12]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1314) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1314) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1322) : error C2065: 'http_server_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1322) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1322) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1322) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1322) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1322) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1322) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1322) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [7]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1322) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1322) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1402) : warning C4013: 'http_is_te' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1440) : error C2065: 'http_te_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1440) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1440) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1440) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1440) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1440) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1440) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1440) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [3]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1440) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1440) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1448) : error C2065: 'http_trailer_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1448) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1448) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1448) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1448) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1448) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1448) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1448) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [8]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1448) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1448) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1457) : error C2065: 'http_transfer_encoding_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1457) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1457) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1457) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1457) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1457) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1457) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1457) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [18]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1457) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1457) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1465) : error C2065: 'http_upgrade_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1465) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1465) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1465) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1465) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1465) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1465) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1465) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [8]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1465) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1465) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1473) : error C2065: 'http_user_agent_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1473) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1473) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1473) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1473) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1473) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1473) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1473) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [11]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1473) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1473) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1481) : error C2065: 'http_vary_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1481) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1481) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1481) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1481) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1481) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1481) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1481) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [5]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1481) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1481) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1588) : error C2065: 'http_via_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1588) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1588) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,http_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1588) : warning C4113: 'isize_t (__cdecl *)(const http_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1588) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const http_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1588) : warning C4113: 'char *(__cdecl *)(http_header_t *,const http_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1588) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(http_header_t *,const http_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1588) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [4]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1588) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1588) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : error C2065: 'http_warning_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : error C2065: 'msg_warning_d' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : error C2065: 'msg_warning_e' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : error C2065: 'msg_warning_dup_xtra' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : error C2065: 'msg_warning_dup_one' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'void *'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : warning C4047: 'initializing' : 'msg_parse_f (__cdecl *)' differs in levels of indirection from 'char [8]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : warning C4047: 'initializing' : 'msg_print_f (__cdecl *)' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : warning C4047: 'initializing' : 'msg_dup_f (__cdecl *)' differs in levels of indirection from 'uintptr_t'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1599) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1608) : error C2065: 'http_www_authenticate_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1608) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1608) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1608) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1608) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1608) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1608) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1608) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [17]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1608) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_basic.c(1608) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>http_extra.c
1>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(57) : error C2065: 'http_proxy_connection_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(57) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(57) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(57) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(57) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(57) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(57) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(57) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [17]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(57) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(57) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(255) : error C2065: 'http_cookie_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(255) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(255) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(255) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(255) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(255) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(255) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(255) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [7]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(255) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(255) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(471) : error C2065: 'http_set_cookie_hash' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(471) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(471) : warning C4047: 'initializing' : 'int' differs in levels of indirection from 'issize_t (__cdecl *)(su_home_t *,msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(471) : warning C4113: 'isize_t (__cdecl *)(const msg_header_t *,isize_t)' differs in parameter lists from 'msg_print_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(471) : warning C4057: 'initializing' : 'msg_print_f (__cdecl *)' differs in indirection to slightly different base types from 'isize_t (__cdecl *)(const msg_header_t *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(471) : warning C4113: 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)' differs in parameter lists from 'msg_xtra_f (__cdecl *)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(471) : warning C4047: 'initializing' : 'msg_xtra_f (__cdecl *)' differs in levels of indirection from 'char *(__cdecl *)(msg_header_t *,const msg_header_t *,char *,isize_t)'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(471) : warning C4047: 'initializing' : 'msg_update_f (__cdecl *)' differs in levels of indirection from 'char [11]'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(471) : warning C4047: 'initializing' : 'const char *' differs in levels of indirection from 'unsigned int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_extra.c(471) : warning C4047: 'initializing' : 'short' differs in levels of indirection from 'char [1]'
1>http_header.c
1>..\..\sofia-sip\libsofia-sip-ua\http\http_header.c(223) : error C2220: warning treated as error - no 'object' file generated
1>..\..\sofia-sip\libsofia-sip-ua\http\http_header.c(223) : warning C4013: 'http_header_vformat' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\http\http_header.c(223) : warning C4047: '=' : 'http_header_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_header.c(274) : warning C4047: '=' : 'http_header_t *' differs in levels of indirection from 'int'
1>http_parser.c
1>http_parser_table.c
1>..\..\sofia-sip\libsofia-sip-ua\http\http_parser_table.c(6) : error C2143: syntax error : missing '{' before '<cv-qualifer>'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_parser_table.c(6) : error C2059: syntax error : '<Unknown>'
1>http_status.c
1>http_tag.c
1>..\..\sofia-sip\libsofia-sip-ua\http\http_tag.c(2) : error C2220: warning treated as error - no 'object' file generated
1>..\..\sofia-sip\libsofia-sip-ua\http\http_tag.c(2) : warning C4206: nonstandard extension used : translation unit is empty
1>http_tag_class.c
1>..\..\sofia-sip\libsofia-sip-ua\http\http_tag_class.c(177) : error C2220: warning treated as error - no 'object' file generated
1>..\..\sofia-sip\libsofia-sip-ua\http\http_tag_class.c(177) : warning C4013: 'HTTPTAG_P' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\http\http_tag_class.c(196) : error C2065: 'httptag_header' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_tag_class.c(196) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\http\http_tag_class.c(203) : warning C4013: 'HTTPTAG_STR_P' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\http\http_tag_class.c(209) : error C2065: 'httptag_header_str' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\http\http_tag_class.c(209) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
1>http_tag_ref.c
1>nth_client.c
1>..\..\sofia-sip\libsofia-sip-ua\nth\nth_client.c(652) : error C2220: warning treated as error - no 'object' file generated
1>..\..\sofia-sip\libsofia-sip-ua\nth\nth_client.c(652) : warning C4013: 'HTTPTAG_VERSION_REF' undefined; assuming extern returning int
1>nth_server.c
1>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(718) : error C2220: warning treated as error - no 'object' file generated
1>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(718) : warning C4013: 'HTTPTAG_SERVER_REF' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(719) : warning C4013: 'HTTPTAG_SERVER_STR_REF' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(751) : warning C4013: 'http_server_dup' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(751) : warning C4047: '=' : 'http_server_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(753) : warning C4013: 'http_server_make' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(753) : warning C4047: '=' : 'http_server_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(892) : warning C4013: 'http_location_init' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(966) : warning C4013: 'http_payload_format' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(970) : warning C4047: '=' : 'http_payload_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(975) : warning C4013: 'http_status_init' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(981) : warning C4013: 'HTTPTAG_STATUS' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(981) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(981) : warning C4024: 'http_add_tl' : different types for formal and actual parameter 3
1>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(982) : warning C4013: 'HTTPTAG_SERVER' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(983) : warning C4013: 'HTTPTAG_CONTENT_TYPE_STR' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(984) : warning C4013: 'HTTPTAG_SEPARATOR_STR' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(985) : warning C4013: 'HTTPTAG_CONNECTION_STR' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(985) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1077) : error C2065: 'http_www_authenticate_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1077) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1109) : warning C4013: 'HTTPTAG_HEADER' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1109) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1109) : warning C4024: 'nth_request_treply' : different types for formal and actual parameter 4
1>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1147) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1147) : warning C4024: 'nth_request_treply' : different types for formal and actual parameter 4
1>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1148) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_type_t'
1>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1148) : warning C4024: 'nth_request_treply' : different types for formal and actual parameter 5
1>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1263) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1263) : warning C4024: 'http_add_tl' : different types for formal and actual parameter 3
1>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1295) : warning C4013: 'http_date_init' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1295) : error C2223: left of '->d_time' must point to struct/union
1>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1311) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1311) : warning C4024: 'http_add_tl' : different types for formal and actual parameter 3
1>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1311) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_type_t'
1>..\..\sofia-sip\libsofia-sip-ua\nth\nth_server.c(1311) : warning C4024: 'http_add_tl' : different types for formal and actual parameter 4
1>nth_tag.c
1>nth_tag_ref.c
1>sres.c
1>sres_blocking.c
1>sres_cache.c
1>sres_sip.c
1>sresolv.c
1>nea.c
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(131) : error C2065: 'siptag_to' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(131) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(131) : warning C4024: 'tl_find' : different types for formal and actual parameter 2
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(131) : error C2065: 'siptag_to_str' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(131) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(131) : warning C4024: 'tl_find' : different types for formal and actual parameter 2
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(133) : error C2065: 'siptag_from' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(133) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(133) : warning C4024: 'tl_find' : different types for formal and actual parameter 2
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(133) : error C2065: 'siptag_from_str' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(133) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(133) : warning C4024: 'tl_find' : different types for formal and actual parameter 2
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(135) : error C2065: 'siptag_contact' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(135) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(135) : warning C4024: 'tl_find' : different types for formal and actual parameter 2
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(136) : error C2065: 'siptag_contact_str' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(136) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(136) : warning C4024: 'tl_find' : different types for formal and actual parameter 2
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(155) : warning C4013: 'SIPTAG_CONTACT' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(155) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(156) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_type_t'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(156) : warning C4024: 'tl_tlist' : different types for formal and actual parameter 3
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(160) : warning C4013: 'SIPTAG_EXPIRES_REF' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(160) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(160) : warning C4024: 'tl_gets' : different types for formal and actual parameter 2
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(161) : warning C4013: 'SIPTAG_EXPIRES_STR_REF' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(162) : warning C4013: 'SIPTAG_TO_REF' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(170) : warning C4013: 'sip_to_dup' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(170) : warning C4047: '=' : 'sip_to_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(173) : warning C4013: 'sip_expires_dup' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(173) : warning C4047: '=' : 'sip_expires_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(175) : warning C4013: 'sip_expires_make' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(175) : warning C4047: '=' : 'sip_expires_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(180) : warning C4013: 'SIPTAG_EXPIRES' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(180) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(180) : warning C4024: 'tl_tremove' : different types for formal and actual parameter 2
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(181) : warning C4013: 'SIPTAG_EXPIRES_STR' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(195) : warning C4013: 'SIPTAG_FROM' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(195) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(196) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_typedef_t'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(196) : warning C4024: 'nta_leg_tcreate' : different types for formal and actual parameter 5
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(205) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(205) : warning C4024: 'nta_outgoing_tcreate' : different types for formal and actual parameter 8
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(206) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_typedef_t'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(206) : warning C4024: 'nta_outgoing_tcreate' : different types for formal and actual parameter 9
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(242) : warning C4013: 'SIPTAG_CONTENT_TYPE_REF' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(242) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(242) : warning C4024: 'tl_gets' : different types for formal and actual parameter 2
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(243) : warning C4013: 'SIPTAG_CONTENT_TYPE_STR_REF' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(244) : warning C4013: 'SIPTAG_PAYLOAD_REF' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(254) : warning C4013: 'SIPTAG_CONTENT_TYPE' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(254) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(254) : warning C4024: 'tl_tremove' : different types for formal and actual parameter 2
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(255) : warning C4013: 'SIPTAG_CONTENT_TYPE_STR' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(256) : warning C4013: 'SIPTAG_PAYLOAD' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(257) : warning C4013: 'SIPTAG_PAYLOAD_STR' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(263) : warning C4047: '=' : 'sip_expires_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(273) : warning C4013: 'SIPTAG_TO' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(273) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(273) : warning C4024: 'nta_outgoing_tcreate' : different types for formal and actual parameter 8
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(275) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(276) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(307) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(307) : warning C4024: 'nta_outgoing_tcreate' : different types for formal and actual parameter 8
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(308) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_typedef_t'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(308) : warning C4024: 'nta_outgoing_tcreate' : different types for formal and actual parameter 9
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(346) : warning C4013: 'SIPTAG_ALLOW_STR' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(346) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(346) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(346) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_type_t'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(346) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 5
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(410) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(410) : warning C4024: 'nta_outgoing_tcreate' : different types for formal and actual parameter 8
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(411) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_typedef_t'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(411) : warning C4024: 'nta_outgoing_tcreate' : different types for formal and actual parameter 9
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(416) : warning C4013: 'sip_expires_format' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(416) : warning C4047: '=' : 'sip_expires_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(425) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(425) : warning C4024: 'nta_outgoing_tcreate' : different types for formal and actual parameter 8
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(426) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_typedef_t'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(426) : warning C4024: 'nta_outgoing_tcreate' : different types for formal and actual parameter 9
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(496) : warning C4013: 'sip_subscription_state_init' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(594) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(594) : warning C4024: 'nta_outgoing_tcreate' : different types for formal and actual parameter 8
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(595) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_typedef_t'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea.c(595) : warning C4024: 'nta_outgoing_tcreate' : different types for formal and actual parameter 9
1>Generating Code...
1>Compiling...
1>nea_debug.c
1>nea_event.c
1>nea_server.c
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(405) : error C2220: warning treated as error - no 'object' file generated
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(405) : warning C4013: 'SIPTAG_CONTACT_REF' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(405) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(405) : warning C4024: 'tl_gets' : different types for formal and actual parameter 2
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(406) : warning C4013: 'SIPTAG_CONTACT_STR_REF' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(407) : warning C4013: 'SIPTAG_ALLOW_EVENTS_REF' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(408) : warning C4013: 'SIPTAG_SERVER_STR_REF' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(409) : warning C4013: 'SIPTAG_REQUIRE_REF' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(410) : warning C4013: 'SIPTAG_REQUIRE_STR_REF' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(454) : warning C4013: 'sip_allow_events_dup' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(454) : warning C4047: '=' : 'sip_allow_events_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(456) : warning C4013: 'sip_allow_events_make' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(456) : warning C4047: '=' : 'sip_allow_events_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(458) : warning C4013: 'sip_allow_make' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(458) : warning C4047: '=' : 'sip_allow_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(467) : warning C4013: 'sip_contact_dup' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(467) : warning C4047: '=' : 'sip_contact_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(469) : warning C4013: 'sip_contact_make' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(469) : warning C4047: '=' : 'sip_contact_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(492) : warning C4013: 'sip_require_dup' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(492) : warning C4047: '=' : 'sip_require_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(494) : warning C4013: 'sip_require_make' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(494) : warning C4047: '=' : 'sip_require_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(730) : warning C4013: 'SIPTAG_CONTENT_TYPE_REF' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(730) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(730) : warning C4024: 'tl_gets' : different types for formal and actual parameter 2
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(731) : warning C4013: 'SIPTAG_CONTENT_TYPE_STR_REF' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(732) : warning C4013: 'SIPTAG_PAYLOAD_REF' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(733) : warning C4013: 'SIPTAG_PAYLOAD_STR_REF' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(789) : warning C4013: 'sip_payload_dup' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(790) : warning C4013: 'sip_payload_make' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(790) : warning C4047: '=' : 'sip_payload_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(792) : warning C4013: 'sip_content_type_dup' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(793) : warning C4013: 'sip_content_type_make' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(793) : warning C4047: '=' : 'sip_content_type_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(829) : warning C4047: '=' : 'sip_payload_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1212) : warning C4013: 'SIPTAG_CONTENT_TYPE_STR' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1212) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1212) : warning C4024: 'nea_event_tcreate' : different types for formal and actual parameter 6
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1213) : warning C4013: 'SIPTAG_ACCEPT_STR' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1274) : warning C4013: 'SIPTAG_ACCEPT_REF' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1275) : warning C4013: 'SIPTAG_ACCEPT_STR_REF' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1276) : warning C4013: 'SIPTAG_SUPPORTED_REF' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1277) : warning C4013: 'SIPTAG_SUPPORTED_STR_REF' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1284) : warning C4013: 'sip_event_format' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1287) : warning C4047: '=' : 'sip_event_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1298) : warning C4047: '=' : 'sip_require_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1300) : warning C4047: '=' : 'sip_require_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1311) : warning C4013: 'sip_accept_make' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1311) : warning C4047: '=' : 'const sip_accept_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1313) : warning C4047: '=' : 'const sip_accept_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1319) : warning C4013: 'sip_accept_dup' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1319) : warning C4047: '=' : 'const sip_accept_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1321) : warning C4047: '=' : 'const sip_accept_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1324) : warning C4013: 'sip_supported_dup' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1324) : warning C4047: '=' : 'sip_supported_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1326) : warning C4013: 'sip_supported_make' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1326) : warning C4047: '=' : 'sip_supported_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1507) : warning C4013: 'sip_from_dup' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1507) : warning C4047: '=' : 'sip_from_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1512) : warning C4047: '=' : 'sip_contact_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1520) : warning C4013: 'SIPTAG_FROM' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1520) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1520) : warning C4024: 'nta_leg_tcreate' : different types for formal and actual parameter 4
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1521) : warning C4013: 'SIPTAG_TO' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1522) : warning C4013: 'SIPTAG_CALL_ID' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1580) : warning C4013: 'SIPTAG_ALLOW_STR' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1580) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1580) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1581) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_type_t'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1581) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 5
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1623) : warning C4013: 'SIP_EXPIRES_INIT' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1623) : warning C4047: 'initializing' : 'msg_header_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1635) : warning C4013: 'SIPTAG_SERVER_STR' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1635) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1635) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1636) : warning C4013: 'SIPTAG_ALLOW_EVENTS' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1637) : warning C4013: 'SIPTAG_ALLOW' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1647) : warning C4013: 'sip_min_expires_init' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1652) : warning C4013: 'SIPTAG_ACCEPT' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1652) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1652) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1653) : warning C4013: 'SIPTAG_MIN_EXPIRES' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1670) : warning C4013: 'SIPTAG_REQUIRE' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1670) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1670) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1671) : warning C4013: 'SIPTAG_UNSUPPORTED' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1692) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1692) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1693) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_type_t'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1693) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 5
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1738) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1738) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1750) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1750) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1752) : warning C4013: 'SIPTAG_SUPPORTED' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1767) : warning C4013: 'sip_accept_init' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1781) : warning C4047: '=' : 'sip_accept_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1857) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1857) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1883) : warning C4047: '=' : 'sip_contact_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1888) : warning C4047: '=' : 'sip_content_type_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1890) : warning C4047: '=' : 'sip_payload_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1905) : warning C4013: 'sip_event_dup' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1905) : warning C4047: '=' : 'sip_event_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1913) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1913) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1918) : warning C4013: 'SIPTAG_EXPIRES' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1919) : warning C4013: 'SIPTAG_CONTACT' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1946) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1946) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1947) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_type_t'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1947) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 5
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1950) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1950) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1960) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(1960) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(2042) : warning C4013: 'sip_subscription_state_init' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(2096) : warning C4013: 'SIPTAG_SUBSCRIPTION_STATE' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(2096) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(2096) : warning C4024: 'nta_outgoing_tcreate' : different types for formal and actual parameter 8
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(2099) : warning C4013: 'SIPTAG_USER_AGENT_STR' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(2101) : warning C4013: 'SIPTAG_EVENT' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(2103) : warning C4013: 'SIPTAG_CONTENT_TYPE' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(2103) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(2105) : warning C4013: 'SIPTAG_PAYLOAD' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nea\nea_server.c(2105) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
1>nea_tag.c
1>nea_tag_ref.c
1>auth_client.c
1>auth_common.c
1>auth_digest.c
1>auth_module.c
1>auth_module_http.c
1>..\..\sofia-sip\libsofia-sip-ua\iptsec\auth_module_http.c(47) : error C2065: 'http_www_authenticate_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\iptsec\auth_module_http.c(47) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\iptsec\auth_module_http.c(50) : error C2065: 'http_proxy_authenticate_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\iptsec\auth_module_http.c(50) : error C2099: initializer is not a constant
1>auth_module_sip.c
1>..\..\sofia-sip\libsofia-sip-ua\iptsec\auth_module_sip.c(49) : error C2065: 'sip_www_authenticate_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\iptsec\auth_module_sip.c(49) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\iptsec\auth_module_sip.c(51) : error C2065: 'sip_authentication_info_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\iptsec\auth_module_sip.c(51) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\iptsec\auth_module_sip.c(54) : error C2065: 'sip_proxy_authenticate_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\iptsec\auth_module_sip.c(54) : error C2099: initializer is not a constant
1>auth_plugin.c
1>auth_plugin_delayed.c
1>auth_tag.c
1>auth_tag_ref.c
1>iptsec_debug.c
1>stun.c
1>stun_common.c
1>stun_dns.c
1>stun_mini.c
1>Generating Code...
1>Compiling...
1>stun_tag.c
1>stun_tag_ref.c
1>nua.c
1>nua_client.c
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(496) : error C2220: warning treated as error - no 'object' file generated
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(496) : warning C4013: 'sip_object' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(496) : warning C4047: '=' : 'sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(502) : error C2065: 'siptag_contact' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(502) : warning C4047: '==' : 'const tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(503) : error C2065: 'siptag_contact_str' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(503) : warning C4047: '==' : 'const tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(529) : error C2065: 'siptag_contact' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(529) : warning C4047: '==' : 'const tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(530) : error C2065: 'siptag_contact_str' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(530) : warning C4047: '==' : 'const tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(549) : warning C4013: 'sip_add_tagis' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(569) : warning C4013: 'sip_to_tag' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(573) : warning C4013: 'sip_add_dup' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(577) : warning C4013: 'sip_add_dup_as' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(577) : error C2065: 'sip_to_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(615) : warning C4047: 'initializing' : 'sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(625) : error C2065: 'siptag_from' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(625) : warning C4047: '==' : 'const tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(743) : warning C4013: 'SIPTAG_CALL_ID' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(743) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(743) : warning C4024: 'nta_leg_tcreate' : different types for formal and actual parameter 4
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(744) : warning C4013: 'SIPTAG_FROM' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(745) : warning C4013: 'SIPTAG_TO' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(746) : warning C4013: 'SIPTAG_CSEQ' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(754) : warning C4013: 'sip_from_tag' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(768) : warning C4047: '=' : 'sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(781) : warning C4013: 'sip_route_dup' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(781) : warning C4047: '=' : 'sip_route_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(813) : warning C4013: 'sip_to_dup' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(813) : warning C4047: '=' : 'const sip_to_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(815) : warning C4013: 'sip_from_dup' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(815) : warning C4047: '=' : 'const sip_from_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(830) : warning C4013: 'sip_has_feature' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(832) : warning C4013: 'sip_add_make' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(832) : error C2065: 'sip_supported_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(835) : error C2065: 'sip_organization_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(838) : error C2065: 'sip_user_agent_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(869) : warning C4013: 'sip_contact_dup' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(869) : warning C4047: 'initializing' : 'sip_contact_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(1164) : warning C4013: 'sip_expires_init' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(1181) : error C2065: 'sip_authorization_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(1181) : warning C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(1181) : warning C4024: 'auc_challenge' : different types for formal and actual parameter 4
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(1186) : error C2065: 'sip_proxy_authorization_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(1186) : warning C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(1186) : warning C4024: 'auc_challenge' : different types for formal and actual parameter 4
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(1477) : error C2065: 'sip_authorization_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(1477) : warning C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(1477) : warning C4024: 'auc_info' : different types for formal and actual parameter 3
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(1481) : error C2065: 'sip_proxy_authorization_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(1481) : warning C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_client.c(1481) : warning C4024: 'auc_info' : different types for formal and actual parameter 3
1>nua_common.c
1>nua_dialog.c
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_types.h(41) : error C2061: syntax error : identifier 'nua_owner_t'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_types.h(41) : error C2059: syntax error : ';'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(47) : error C2016: C requires that a struct or union has at least one member
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(47) : error C2061: syntax error : identifier 'nua_owner_t'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(58) : error C2143: syntax error : missing '{' before ':'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(58) : error C2059: syntax error : ':'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(60) : error C2143: syntax error : missing '{' before ':'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(60) : error C2059: syntax error : ':'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(61) : error C2143: syntax error : missing '{' before ':'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(61) : error C2059: syntax error : ':'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(63) : error C2143: syntax error : missing '{' before ':'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(63) : error C2059: syntax error : ':'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(64) : error C2143: syntax error : missing '{' before ':'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(64) : error C2059: syntax error : ':'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(65) : error C2143: syntax error : missing '{' before ':'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(65) : error C2059: syntax error : ':'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(67) : error C2143: syntax error : missing '{' before ':'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(67) : error C2059: syntax error : ':'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(68) : error C2143: syntax error : missing '{' before ':'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(68) : error C2059: syntax error : ':'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(70) : error C2059: syntax error : ':'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(92) : error C2059: syntax error : '}'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(97) : error C2143: syntax error : missing ')' before '*'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(97) : error C2143: syntax error : missing ';' before '*'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(97) : error C2059: syntax error : '*'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(99) : error C2059: syntax error : ')'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(100) : error C2143: syntax error : missing ')' before '*'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(100) : error C2143: syntax error : missing '{' before '*'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(100) : error C2059: syntax error : ','
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(104) : error C2059: syntax error : ')'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(113) : error C2143: syntax error : missing ')' before '*'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(113) : error C2143: syntax error : missing '{' before '*'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(113) : error C2059: syntax error : ','
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(114) : error C2059: syntax error : ')'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(115) : error C2143: syntax error : missing ')' before '*'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(115) : error C2143: syntax error : missing '{' before '*'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(115) : error C2059: syntax error : ','
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(116) : error C2059: syntax error : ')'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(117) : error C2059: syntax error : '}'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(123) : error C2061: syntax error : identifier 'nua_usage_class'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(129) : error C2143: syntax error : missing '{' before ':'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(129) : error C2059: syntax error : ':'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(130) : error C2143: syntax error : missing '{' before ':'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(130) : error C2059: syntax error : ':'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(131) : error C2059: syntax error : ':'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(140) : error C2059: syntax error : '}'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(142) : error C2143: syntax error : missing ')' before '*'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(142) : error C2143: syntax error : missing '{' before '*'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(142) : error C2059: syntax error : ','
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(143) : error C2059: syntax error : ')'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(144) : error C2143: syntax error : missing ')' before '*'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(144) : error C2143: syntax error : missing '{' before '*'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(144) : error C2059: syntax error : ','
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(145) : error C2059: syntax error : ')'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(146) : error C2143: syntax error : missing ')' before '*'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(146) : error C2143: syntax error : missing '{' before '*'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(146) : error C2059: syntax error : ','
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(147) : error C2059: syntax error : ')'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(148) : error C2143: syntax error : missing ')' before '*'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(148) : error C2143: syntax error : missing '{' before '*'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(148) : warning C4431: missing type specifier - int assumed. Note: C no longer supports default-int
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(149) : warning C4431: missing type specifier - int assumed. Note: C no longer supports default-int
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(149) : error C2371: 'nua_dialog_state_t' : redefinition; different basic types
1>??????? c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_types.h(43) : see declaration of 'nua_dialog_state_t'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(149) : error C2143: syntax error : missing ';' before '*'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(149) : warning C4218: nonstandard extension used : must specify at least a storage class or a type
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(149) : error C2059: syntax error : ')'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(149) : warning C4431: missing type specifier - int assumed. Note: C no longer supports default-int
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(149) : warning C4218: nonstandard extension used : must specify at least a storage class or a type
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(150) : error C2143: syntax error : missing ')' before '*'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(150) : error C2143: syntax error : missing '{' before '*'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(150) : warning C4431: missing type specifier - int assumed. Note: C no longer supports default-int
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(151) : warning C4431: missing type specifier - int assumed. Note: C no longer supports default-int
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(151) : error C2371: 'nua_dialog_state_t' : redefinition; different basic types
1>??????? c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_types.h(43) : see declaration of 'nua_dialog_state_t'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(151) : error C2143: syntax error : missing ';' before '*'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(151) : warning C4218: nonstandard extension used : must specify at least a storage class or a type
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(151) : warning C4431: missing type specifier - int assumed. Note: C no longer supports default-int
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(152) : warning C4431: missing type specifier - int assumed. Note: C no longer supports default-int
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(152) : error C2371: 'nua_dialog_usage_t' : redefinition; different basic types
1>??????? c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_types.h(44) : see declaration of 'nua_dialog_usage_t'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(152) : error C2143: syntax error : missing ';' before '*'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(152) : warning C4218: nonstandard extension used : must specify at least a storage class or a type
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(152) : error C2059: syntax error : ')'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(152) : warning C4431: missing type specifier - int assumed. Note: C no longer supports default-int
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(152) : warning C4218: nonstandard extension used : must specify at least a storage class or a type
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(156) : error C2037: left of 'ds_reporting' specifies undefined struct/union 'nua_dialog_state'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(156) : warning C4033: 'nua_dialog_is_reporting' must return a value
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(161) : error C2143: syntax error : missing ')' before '*'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(161) : error C2143: syntax error : missing '{' before '*'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(161) : error C2059: syntax error : ','
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(164) : error C2059: syntax error : ')'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(167) : error C2143: syntax error : missing ')' before 'const'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(167) : error C2081: 'nua_usage_class' : name in formal parameter list illegal
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(167) : error C2143: syntax error : missing '{' before 'const'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(167) : warning C4431: missing type specifier - int assumed. Note: C no longer supports default-int
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(168) : error C2373: 'sip_event_t' : redefinition; different type modifiers
1>??????? c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\sip\sofia-sip/sip.h(212) : see declaration of 'sip_event_t'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(168) : error C2143: syntax error : missing ';' before 'const'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(168) : error C2059: syntax error : ')'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(170) : error C2143: syntax error : missing ')' before '*'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(170) : error C2143: syntax error : missing '{' before '*'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(170) : error C2059: syntax error : ','
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(174) : error C2059: syntax error : ')'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(186) : error C2143: syntax error : missing ')' before '*'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(186) : error C2143: syntax error : missing '{' before '*'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(186) : warning C4431: missing type specifier - int assumed. Note: C no longer supports default-int
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(187) : warning C4431: missing type specifier - int assumed. Note: C no longer supports default-int
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(187) : error C2371: 'nua_dialog_state_t' : redefinition; different basic types
1>??????? c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_types.h(43) : see declaration of 'nua_dialog_state_t'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(187) : error C2143: syntax error : missing ';' before '*'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(187) : warning C4218: nonstandard extension used : must specify at least a storage class or a type
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(187) : error C2059: syntax error : ')'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(187) : warning C4431: missing type specifier - int assumed. Note: C no longer supports default-int
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(187) : warning C4218: nonstandard extension used : must specify at least a storage class or a type
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(189) : error C2143: syntax error : missing ')' before '*'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(189) : error C2143: syntax error : missing '{' before '*'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(189) : warning C4431: missing type specifier - int assumed. Note: C no longer supports default-int
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(189) : warning C4431: missing type specifier - int assumed. Note: C no longer supports default-int
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(189) : error C2371: 'nua_dialog_state_t' : redefinition; different basic types
1>??????? c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_types.h(43) : see declaration of 'nua_dialog_state_t'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(189) : error C2143: syntax error : missing ';' before '*'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(189) : warning C4218: nonstandard extension used : must specify at least a storage class or a type
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(189) : error C2059: syntax error : ')'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(189) : warning C4431: missing type specifier - int assumed. Note: C no longer supports default-int
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(189) : warning C4218: nonstandard extension used : must specify at least a storage class or a type
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(191) : error C2143: syntax error : missing ')' before '*'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(191) : error C2143: syntax error : missing '{' before '*'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(191) : warning C4431: missing type specifier - int assumed. Note: C no longer supports default-int
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(192) : warning C4431: missing type specifier - int assumed. Note: C no longer supports default-int
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(192) : error C2371: 'nua_dialog_state_t' : redefinition; different basic types
1>??????? c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_types.h(43) : see declaration of 'nua_dialog_state_t'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(192) : error C2143: syntax error : missing ';' before '*'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(192) : warning C4218: nonstandard extension used : must specify at least a storage class or a type
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(192) : error C2059: syntax error : ')'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(192) : warning C4431: missing type specifier - int assumed. Note: C no longer supports default-int
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(192) : warning C4218: nonstandard extension used : must specify at least a storage class or a type
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(204) : error C2143: syntax error : missing ')' before '*'
1>c:\c4dev\freeswitch\libs\sofia-sip\libsofia-sip-ua\nua\nua_dialog.h(204) : fatal error C1003: error count exceeds 100; stopping compilation
1>nua_event_server.c
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(86) : error C2220: warning treated as error - no 'object' file generated
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(86) : warning C4013: 'SIPTAG_EVENT_REF' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(87) : warning C4013: 'SIPTAG_EVENT_STR_REF' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(88) : warning C4013: 'SIPTAG_CONTENT_TYPE_STR_REF' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(89) : warning C4013: 'SIPTAG_PAYLOAD_REF' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(90) : warning C4013: 'SIPTAG_PAYLOAD_STR_REF' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(108) : warning C4013: 'sip_event_make' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(108) : warning C4047: '=' : 'const sip_event_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(122) : warning C4013: 'SIPTAG_EVENT' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(122) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(122) : warning C4024: 'nua_stack_tevent' : different types for formal and actual parameter 7
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(123) : warning C4013: 'SIPTAG_CONTENT_TYPE' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(158) : warning C4013: 'SIPTAG_ACCEPT_REF' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(158) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(158) : warning C4024: 'tl_gets' : different types for formal and actual parameter 2
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(159) : warning C4013: 'SIPTAG_ACCEPT_STR_REF' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(160) : warning C4013: 'SIPTAG_CONTENT_TYPE_REF' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(169) : warning C4013: 'sip_header_as_string' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(169) : warning C4047: '=' : 'char *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(334) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_event_server.c(334) : warning C4024: 'tl_gets' : different types for formal and actual parameter 2
1>nua_extension.c
1>nua_message.c
1>nua_notifier.c
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_notifier.c(47) : fatal error C1083: Cannot open include file: 'sofia-sip/sip_extra.h': No such file or directory
1>nua_options.c
1>nua_params.c
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(166) : error C2220: warning treated as error - no 'object' file generated
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(166) : warning C4013: 'sip_allow_make' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(166) : warning C4047: '=' : 'sip_allow_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(167) : warning C4013: 'sip_supported_make' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(167) : warning C4047: '=' : 'sip_supported_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(175) : warning C4047: '=' : 'sip_allow_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(196) : warning C4013: 'SIPTAG_FROM_REF' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(196) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(196) : warning C4024: 'tl_gets' : different types for formal and actual parameter 2
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(197) : warning C4013: 'SIPTAG_FROM_STR_REF' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(203) : warning C4013: 'sip_from_init' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(208) : warning C4013: 'sip_from_dup' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(208) : warning C4047: '=' : 'sip_from_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(212) : warning C4013: 'sip_from_make' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(212) : warning C4047: '=' : 'sip_from_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(225) : warning C4047: '=' : 'sip_from_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(833) : error C2065: 'siptag_supported' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(833) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(834) : error C2065: 'siptag_supported_str' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(834) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(839) : error C2065: 'sip_supported_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(839) : warning C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(839) : warning C4024: 'nhp_merge_lists' : different types for formal and actual parameter 2
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(841) : error C2065: 'siptag_supported' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(841) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(853) : error C2065: 'siptag_allow_str' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(853) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(854) : error C2065: 'siptag_allow' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(854) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(859) : error C2065: 'sip_allow_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(859) : warning C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(859) : warning C4024: 'nhp_merge_lists' : different types for formal and actual parameter 2
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(863) : error C2065: 'siptag_allow' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(863) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(875) : error C2065: 'siptag_allow_events_str' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(875) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(876) : error C2065: 'siptag_allow_events' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(876) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(881) : error C2065: 'sip_allow_events_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(881) : warning C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(881) : warning C4024: 'nhp_merge_lists' : different types for formal and actual parameter 2
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(885) : error C2065: 'siptag_allow_events' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(885) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(903) : error C2065: 'sip_allow_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(903) : warning C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(903) : warning C4024: 'nhp_merge_lists' : different types for formal and actual parameter 2
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(923) : warning C4013: 'sip_route_make' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(923) : warning C4013: 'sip_route_dup' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(923) : warning C4047: '=' : 'sip_route_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(927) : error C2065: 'siptag_user_agent' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(927) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(928) : warning C4013: 'sip_header_as_string' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(928) : warning C4047: '=' : 'char *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(931) : error C2065: 'siptag_user_agent_str' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(931) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(951) : error C2065: 'siptag_organization' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(951) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(952) : warning C4047: '=' : 'char *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(955) : error C2065: 'siptag_organization_str' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(955) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1201) : error C2065: 'siptag_from' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1201) : warning C4047: '==' : 'const tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1205) : error C2065: 'siptag_from_str' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1205) : warning C4047: '==' : 'const tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1209) : error C2065: 'siptag_to' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1209) : warning C4047: '==' : 'const tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1213) : error C2065: 'siptag_to_str' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1213) : warning C4047: '==' : 'const tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1235) : warning C4047: '=' : 'const sip_from_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1242) : warning C4013: 'sip_to_make' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1242) : warning C4047: '=' : 'const sip_to_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1244) : warning C4013: 'sip_to_create' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1244) : warning C4047: '=' : 'const sip_to_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1256) : warning C4013: 'SIPTAG_FROM' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1256) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1257) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'const tag_type_s *'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1257) : warning C4024: 'tl_filtered_tlist' : different types for formal and actual parameter 4
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1258) : warning C4013: 'SIPTAG_TO' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1258) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1269) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1269) : warning C4024: 'tl_gets' : different types for formal and actual parameter 2
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1270) : warning C4013: 'SIPTAG_TO_REF' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1319) : error C2065: 'siptag_from' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1319) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1319) : error C2065: 'siptag_to' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1319) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1327) : error C2065: 'siptag_from_str' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1327) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1329) : error C2065: 'siptag_to_str' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1329) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1333) : error C2065: 'siptag_cseq' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1333) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1333) : error C2065: 'siptag_cseq_str' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1333) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1335) : error C2065: 'siptag_rseq' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1335) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1335) : error C2065: 'siptag_rseq_str' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1335) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1337) : error C2065: 'siptag_rack' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1337) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1337) : error C2065: 'siptag_rack_str' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1337) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1339) : error C2065: 'siptag_timestamp' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1339) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1339) : error C2065: 'siptag_timestamp_str' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1339) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1341) : error C2065: 'siptag_content_length' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1341) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1341) : error C2065: 'siptag_content_length_str' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1341) : warning C4047: '==' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1575) : warning C4013: 'siptag_contact_vr' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1621) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1625) : warning C4013: 'SIPTAG_FROM_STR' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1625) : warning C4047: ':' : 'int' differs in levels of indirection from 'void *'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1625) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1625) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'const tag_type_s *'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1625) : warning C4024: 'tl_filtered_tlist' : different types for formal and actual parameter 4
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1662) : warning C4013: 'SIPTAG_SUPPORTED' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1662) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1663) : warning C4013: 'SIPTAG_SUPPORTED_STR' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1663) : warning C4047: ':' : 'int' differs in levels of indirection from 'void *'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1663) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1664) : warning C4013: 'SIPTAG_ALLOW' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1664) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1665) : warning C4013: 'SIPTAG_ALLOW_STR' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1665) : warning C4047: ':' : 'int' differs in levels of indirection from 'void *'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1665) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1666) : warning C4047: ':' : 'int' differs in levels of indirection from 'void *'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1667) : warning C4013: 'SIPTAG_ALLOW_EVENTS' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1667) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1668) : warning C4013: 'SIPTAG_ALLOW_EVENTS_STR' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1668) : warning C4047: ':' : 'int' differs in levels of indirection from 'void *'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1668) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1669) : warning C4013: 'SIPTAG_USER_AGENT' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1669) : warning C4013: 'sip_user_agent_make' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1669) : warning C4047: ':' : 'int' differs in levels of indirection from 'void *'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1669) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1670) : warning C4013: 'SIPTAG_USER_AGENT_STR' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1670) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1673) : warning C4013: 'SIPTAG_ORGANIZATION' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1673) : warning C4013: 'sip_organization_make' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1673) : warning C4047: ':' : 'int' differs in levels of indirection from 'void *'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1673) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1674) : warning C4013: 'SIPTAG_ORGANIZATION_STR' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1674) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1676) : warning C4013: 'siptag_route_v' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1677) : warning C4047: ':' : 'int' differs in levels of indirection from 'void *'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_params.c(1705) : warning C4013: 'siptag_contact_v' undefined; assuming extern returning int
1>nua_publish.c
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(292) : error C2220: warning treated as error - no 'object' file generated
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(292) : warning C4013: 'msg_copy' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(292) : warning C4047: '=' : 'msg_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(314) : warning C4013: 'sip_etag_dup' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(314) : warning C4047: '=' : 'sip_etag_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(317) : warning C4013: 'sip_header_remove' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(355) : warning C4013: 'SIPTAG_IF_MATCH' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(355) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(355) : warning C4024: 'nua_base_client_trequest' : different types for formal and actual parameter 4
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(356) : warning C4013: 'SIPTAG_PAYLOAD' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(356) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(356) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'const tag_type_s *'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(356) : warning C4024: 'nua_base_client_trequest' : different types for formal and actual parameter 5
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(357) : warning C4013: 'SIPTAG_CONTENT_TYPE' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(357) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(358) : warning C4013: 'SIPTAG_EXPIRES_STR' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(358) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(408) : warning C4047: '=' : 'sip_etag_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_publish.c(534) : warning C4013: 'msg_header_find_param' undefined; assuming extern returning int
1>nua_register.c
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(756) : error C2220: warning treated as error - no 'object' file generated
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(756) : warning C4013: 'sip_add_dup' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(770) : warning C4013: 'sip_header_remove' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(793) : warning C4013: 'SIPTAG_EXPIRES_STR' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(793) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(798) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_typedef_t'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(798) : warning C4024: 'nua_base_client_trequest' : different types for formal and actual parameter 5
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(812) : warning C4013: 'sip_object' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(812) : warning C4047: 'initializing' : 'sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(855) : warning C4013: 'sip_now' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(860) : warning C4047: 'initializing' : 'sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(905) : warning C4013: 'sip_contact_expires' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(939) : warning C4013: 'sip_route_dup' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(939) : warning C4047: '=' : 'sip_route_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(952) : warning C4013: 'sip_path_dup' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(952) : warning C4047: '=' : 'sip_path_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1319) : warning C4013: 'sip_via_init' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1319) : error C2223: left of '->v_next' must point to struct/union
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1320) : error C2065: 'sip_transport_udp' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1320) : warning C4047: '=' : 'const char *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1323) : error C2065: 'sip_transport_tcp' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1323) : warning C4047: '=' : 'const char *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1360) : warning C4013: 'sip_via_copy' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1360) : warning C4047: '=' : 'sip_via_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1374) : error C2065: 'sip_transport_tcp' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1374) : warning C4047: '==' : 'const char *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1375) : error C2065: 'sip_transport_udp' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1375) : warning C4047: '=' : 'const char *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1376) : error C2065: 'sip_transport_udp' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1376) : warning C4047: '==' : 'const char *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1377) : error C2065: 'sip_transport_tcp' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1377) : warning C4047: '=' : 'const char *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1422) : warning C4013: 'sip_via_dup' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1422) : warning C4047: '=' : 'sip_via_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1647) : warning C4047: '=' : 'sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1674) : warning C4047: '=' : 'sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1743) : warning C4013: 'sip_contact_format' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1755) : warning C4047: '=' : 'const sip_contact_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1806) : warning C4013: 'sip_from_dup' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1806) : warning C4047: '=' : 'sip_from_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1834) : warning C4013: 'sip_contact_dup' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(1834) : warning C4047: '=' : 'sip_contact_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(2006) : warning C4013: 'sip_via_port' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(2006) : warning C4047: '=' : 'const char *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(2122) : warning C4013: 'sip_contact_make' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_register.c(2122) : warning C4047: '=' : 'sip_contact_t *' differs in levels of indirection from 'int'
1>nua_registrar.c
1>nua_server.c
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(118) : error C2220: warning treated as error - no 'object' file generated
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(118) : warning C4013: 'SIPTAG_SUPPORTED' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(118) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(118) : warning C4024: 'nta_check_method' : different types for formal and actual parameter 4
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(119) : warning C4013: 'SIPTAG_USER_AGENT_STR' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(132) : warning C4013: 'SIPTAG_ALLOW' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(132) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(132) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(140) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(140) : warning C4024: 'nta_check_required' : different types for formal and actual parameter 4
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(178) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(178) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(207) : warning C4013: 'sip_object' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(207) : warning C4047: '=' : 'sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(236) : warning C4013: 'sip_is_allowed' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(266) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(266) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(267) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_type_t'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(267) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 5
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(497) : warning C4013: 'SIPTAG_END' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(497) : warning C4047: 'initializing' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(519) : warning C4047: '=' : 'sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(530) : warning C4013: 'sip_add_dup' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(533) : warning C4013: 'sip_add_make' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(533) : error C2065: 'sip_user_agent_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(537) : error C2065: 'sip_organization_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(570) : warning C4013: 'sip_contact_dup' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_server.c(570) : warning C4047: '=' : 'sip_contact_t *' differs in levels of indirection from 'int'
1>nua_session.c
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(797) : error C2220: warning treated as error - no 'object' file generated
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(797) : warning C4013: 'sip_has_feature' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(819) : warning C4013: 'sip_accept_contact_init' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(826) : warning C4013: 'sip_add_dup' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1076) : warning C4013: 'SIP_IS_ALLOWED' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1079) : warning C4013: 'sip_rack_init' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1086) : warning C4013: 'SIPTAG_RACK' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1086) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1086) : warning C4024: 'nua_client_tcreate' : different types for formal and actual parameter 4
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1087) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_type_t'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1087) : warning C4024: 'nua_client_tcreate' : different types for formal and actual parameter 5
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1261) : warning C4013: 'sip_object' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1261) : warning C4047: '=' : 'sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1266) : warning C4013: 'sip_authorization' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1266) : warning C4047: '=' : 'sip_authorization_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1267) : warning C4013: 'sip_proxy_authorization' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1267) : warning C4047: '=' : 'sip_proxy_authorization_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1272) : warning C4047: '=' : 'sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1276) : warning C4013: 'sip_cseq_create' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1276) : warning C4047: '=' : 'sip_cseq_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1280) : warning C4013: 'sip_add_tl' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1288) : warning C4013: 'sip_header_insert' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1295) : warning C4013: 'sip_header_remove' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(1362) : warning C4013: 'SIPTAG_END' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2164) : warning C4013: 'msg_content_type_dup' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2164) : warning C4047: '=' : 'sip_content_type_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2189) : warning C4013: 'sip_add_make' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2189) : error C2065: 'sip_accept_encoding_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2195) : error C2065: 'sip_accept_encoding_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2383) : warning C4013: 'sip_warning_format' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2384) : warning C4047: '=' : 'sip_warning_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2760) : warning C4013: 'SIPTAG_REASON_STR' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2760) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2760) : warning C4024: 'nua_server_trespond' : different types for formal and actual parameter 2
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2761) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_type_t'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2761) : warning C4024: 'nua_server_trespond' : different types for formal and actual parameter 3
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2857) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2857) : warning C4024: 'nua_server_trespond' : different types for formal and actual parameter 2
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2858) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_type_t'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(2858) : warning C4024: 'nua_server_trespond' : different types for formal and actual parameter 3
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(3017) : warning C4013: 'siptag_event_vr' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(3034) : warning C4013: 'siptag_event_v' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(3038) : warning C4013: 'sip_event_dup' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(3038) : warning C4047: '=' : 'sip_event_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(3098) : warning C4013: 'SIPTAG_EVENT' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(3098) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(3098) : warning C4024: 'nua_stack_post_signal' : different types for formal and actual parameter 3
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(3099) : warning C4013: 'SIPTAG_SUBSCRIPTION_STATE_STR' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(3100) : warning C4013: 'SIPTAG_CONTENT_TYPE_STR' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(3101) : warning C4013: 'SIPTAG_PAYLOAD_STR' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(3802) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(3802) : warning C4024: 'nua_base_client_trequest' : different types for formal and actual parameter 4
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(3803) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_typedef_t'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(3803) : warning C4024: 'nua_base_client_trequest' : different types for formal and actual parameter 5
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4264) : warning C4013: 'sip_retry_after_init' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4355) : warning C4013: 'sip_min_se_init' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4355) : error C2223: left of '->min_delta' must point to struct/union
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4461) : error C2223: left of '->min_delta' must point to struct/union
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4463) : warning C4013: 'sip_session_expires_init' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4463) : error C2223: left of '->x_delta' must point to struct/union
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4474) : warning C4013: 'SIPTAG_SESSION_EXPIRES' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4474) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4478) : warning C4013: 'SIPTAG_MIN_SE' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4478) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4480) : warning C4013: 'SIPTAG_REQUIRE_STR' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4480) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4674) : warning C4013: 'sip_payload_create' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4674) : warning C4047: '=' : 'sip_payload_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4675) : warning C4013: 'sip_content_type_make' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4675) : warning C4047: '=' : 'sip_content_type_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4677) : warning C4013: 'sip_content_disposition_make' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4677) : warning C4047: '=' : 'sip_content_disposition_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_session.c(4752) : warning C4013: 'SIPTAG_ACCEPT' undefined; assuming extern returning int
1>nua_stack.c
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(152) : error C2220: warning treated as error - no 'object' file generated
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(152) : warning C4013: 'sip_from_init' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(174) : warning C4013: 'sip_accept_make' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(174) : warning C4047: '=' : 'sip_accept_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(396) : warning C4013: 'sip_object' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(396) : warning C4047: ':' : 'int' differs in levels of indirection from 'void *'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(396) : warning C4047: 'function' : 'const sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(396) : warning C4024: 'function through pointer' : different types for formal and actual parameter 8
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(993) : error C2223: left of '->a_display' must point to struct/union
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(996) : warning C4013: 'sip_to_init' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(996) : error C2223: left of '->a_display' must point to struct/union
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(1001) : warning C4013: 'SIPTAG_TO' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(1002) : warning C4013: 'SIPTAG_FROM' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(1014) : warning C4013: 'SIPTAG_CALL_ID' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(1014) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_stack.c(1014) : warning C4024: 'nta_leg_tcreate' : different types for formal and actual parameter 4
1>nua_subnotref.c
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_subnotref.c(50) : fatal error C1083: Cannot open include file: 'sofia-sip/sip_extra.h': No such file or directory
1>nua_tag.c
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_tag.c(2389) : error C2065: 'sip_event_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\nua_tag.c(2389) : error C2099: initializer is not a constant
1>Generating Code...
1>Compiling...
1>nua_tag_ref.c
1>outbound.c
1>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(405) : error C2220: warning treated as error - no 'object' file generated
1>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(405) : warning C4013: 'sip_contact_dup' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(405) : warning C4047: '=' : 'sip_contact_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(523) : warning C4013: 'sip_via_port' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(523) : warning C4047: '=' : 'const char *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(623) : warning C4013: 'sip_contact_format' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(623) : warning C4047: ':' : 'int' differs in levels of indirection from 'void *'
1>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(623) : warning C4047: '=' : 'sip_contact_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(667) : error C2065: 'sip_transport_udp' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(667) : warning C4047: '==' : 'const char *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(709) : warning C4013: 'sip_object' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(709) : warning C4047: 'initializing' : 'const sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(739) : warning C4047: 'initializing' : 'sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(754) : warning C4013: 'sip_accept_contact_make' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(754) : warning C4047: '=' : 'sip_accept_contact_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(773) : warning C4013: 'sip_add_tl' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(774) : warning C4013: 'SIPTAG_TO' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(775) : warning C4013: 'SIPTAG_FROM' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(780) : warning C4013: 'SIPTAG_ROUTE' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(782) : warning C4013: 'SIPTAG_MAX_FORWARDS_STR' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(782) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(783) : warning C4013: 'SIPTAG_SUBJECT_STR' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(783) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(784) : warning C4013: 'SIPTAG_CALL_ID_STR' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(785) : warning C4013: 'SIPTAG_ACCEPT_STR' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(817) : warning C4047: '=' : 'sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(878) : warning C4047: 'initializing' : 'sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(892) : error C2065: 'sip_authorization_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(892) : warning C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(892) : warning C4024: 'auc_challenge' : different types for formal and actual parameter 4
1>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(897) : error C2065: 'sip_proxy_authorization_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(897) : warning C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(897) : warning C4024: 'auc_challenge' : different types for formal and actual parameter 4
1>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(1000) : warning C4047: '=' : 'sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(1022) : warning C4013: 'SIPTAG_MAX_FORWARDS' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(1061) : warning C4013: 'SIPTAG_CONTENT_TYPE_STR' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(1061) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(1061) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
1>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(1062) : warning C4013: 'SIPTAG_PAYLOAD_STR' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(1101) : warning C4013: 'sip_via_dup' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(1101) : warning C4047: '=' : 'sip_via_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(1187) : warning C4047: '=' : 'sip_contact_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nua\outbound.c(1268) : warning C4013: 'sip_has_feature' undefined; assuming extern returning int
1>nta.c
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(912) : error C2220: warning treated as error - no 'object' file generated
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(912) : warning C4013: 'sip_max_forwards_init' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(1497) : warning C4013: 'siptag_contact_vr' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(1567) : warning C4013: 'sip_contact_dup' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(1567) : warning C4047: '=' : 'sip_contact_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(1779) : warning C4013: 'siptag_contact_v' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(2380) : warning C4013: 'sip_via_format' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(2385) : warning C4047: '=' : 'sip_via_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(2396) : warning C4013: 'sip_via_dup' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(2396) : warning C4047: '=' : 'sip_via_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(2398) : warning C4047: '=' : 'sip_via_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(2535) : warning C4013: 'sip_object' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(2535) : warning C4047: 'initializing' : 'sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(2549) : warning C4013: 'sip_via_copy' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(2549) : warning C4047: '=' : 'sip_via_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(2746) : warning C4047: 'initializing' : 'sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(2840) : error C2065: 'sip_error_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(2840) : warning C4047: '==' : 'msg_hclass_t *const ' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3034) : warning C4047: '=' : 'sip_via_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3392) : warning C4047: 'initializing' : 'sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3566) : warning C4047: '=' : 'sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3650) : warning C4047: 'initializing' : 'sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3651) : warning C4047: 'initializing' : 'const sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3661) : warning C4047: '=' : 'sip_via_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3663) : warning C4013: 'sip_from_dup' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3663) : warning C4047: '=' : 'sip_from_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3665) : warning C4013: 'sip_to_dup' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3665) : warning C4047: '=' : 'sip_to_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3668) : warning C4013: 'sip_call_id_dup' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3668) : warning C4047: '=' : 'sip_call_id_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3670) : warning C4013: 'sip_cseq_dup' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3670) : warning C4047: '=' : 'sip_cseq_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3708) : warning C4047: 'initializing' : 'sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3710) : warning C4047: 'initializing' : 'sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3724) : warning C4013: 'SIPTAG_TO' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3724) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3724) : warning C4024: 'sip_add_tl' : different types for formal and actual parameter 3
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3725) : warning C4013: 'SIPTAG_FROM' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3726) : warning C4013: 'SIPTAG_CALL_ID' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3743) : warning C4013: 'sip_route_init' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3743) : error C2223: left of '->r_url' must point to struct/union
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3744) : warning C4013: 'sip_route_dup' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3744) : warning C4047: '=' : 'sip_route_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3753) : warning C4047: '=' : 'sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3828) : warning C4047: 'initializing' : 'sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3842) : warning C4047: '=' : 'sip_route_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3848) : warning C4047: '=' : 'sip_route_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3854) : warning C4047: '=' : 'sip_route_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3924) : warning C4047: '=' : 'sip_from_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3941) : warning C4047: '=' : 'sip_to_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(3969) : warning C4047: '=' : 'sip_call_id_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4107) : warning C4013: 'SIPTAG_CALL_ID_REF' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4108) : warning C4013: 'SIPTAG_CALL_ID_STR_REF' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4109) : warning C4013: 'SIPTAG_FROM_REF' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4110) : warning C4013: 'SIPTAG_FROM_STR_REF' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4111) : warning C4013: 'SIPTAG_TO_REF' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4112) : warning C4013: 'SIPTAG_TO_STR_REF' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4113) : warning C4013: 'SIPTAG_ROUTE_REF' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4116) : warning C4013: 'SIPTAG_CSEQ_REF' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4117) : warning C4013: 'SIPTAG_CSEQ_STR_REF' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4139) : warning C4013: 'sip_is_to' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4140) : warning C4047: '=' : 'sip_to_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4143) : warning C4013: 'sip_to_make' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4143) : warning C4047: '=' : 'sip_to_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4148) : warning C4047: '=' : 'sip_from_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4151) : warning C4047: '=' : 'sip_from_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4153) : warning C4013: 'sip_from_make' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4153) : warning C4047: '=' : 'sip_from_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4156) : warning C4047: '=' : 'sip_route_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4160) : warning C4013: 'sip_contact_init' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4163) : warning C4047: '=' : 'sip_contact_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4215) : warning C4047: '=' : 'sip_call_id_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4217) : warning C4013: 'sip_call_id_make' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4217) : warning C4047: '=' : 'sip_call_id_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4580) : warning C4013: 'sip_replaces_format' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4582) : warning C4047: 'return' : 'sip_replaces_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(4597) : warning C4013: 'sip_call_id_init' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5069) : warning C4047: '=' : 'sip_contact_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5220) : warning C4047: '=' : 'sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5270) : warning C4013: 'sip_request_copy' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5270) : warning C4047: '=' : 'sip_request_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5271) : warning C4013: 'sip_from_copy' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5271) : warning C4047: '=' : 'sip_from_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5272) : warning C4013: 'sip_to_copy' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5272) : warning C4047: '=' : 'sip_to_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5273) : warning C4013: 'sip_call_id_copy' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5273) : warning C4047: '=' : 'sip_call_id_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5274) : warning C4013: 'sip_cseq_copy' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5274) : warning C4047: '=' : 'sip_cseq_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5275) : warning C4047: '=' : 'sip_via_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5288) : warning C4013: 'sip_record_route_copy' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5288) : warning C4047: '=' : 'sip_record_route_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5295) : warning C4013: 'sip_timestamp_copy' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(5295) : warning C4047: '=' : 'sip_timestamp_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(6310) : warning C4047: '=' : 'sip_from_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(6312) : warning C4047: '=' : 'sip_to_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(6314) : warning C4047: '=' : 'sip_call_id_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(6316) : warning C4047: '=' : 'sip_cseq_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(6318) : warning C4047: '=' : 'sip_via_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(6349) : warning C4047: 'initializing' : 'sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(6407) : warning C4047: '=' : 'sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(6485) : warning C4047: 'initializing' : 'sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(7224) : warning C4047: '=' : 'sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(7315) : warning C4047: 'function' : 'sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(7315) : warning C4024: 'sip_add_tl' : different types for formal and actual parameter 2
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(7685) : warning C4047: '=' : 'sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(7861) : warning C4047: 'initializing' : 'const sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(8055) : warning C4047: 'initializing' : 'sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(8057) : warning C4013: 'sip_timestamp_format' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(8058) : warning C4047: 'initializing' : 'sip_timestamp_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(8200) : warning C4047: 'initializing' : 'sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(8246) : warning C4047: 'initializing' : 'sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(8364) : warning C4013: 'sip_supported_init' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(9449) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(9449) : warning C4024: 'outgoing_ackmsg' : different types for formal and actual parameter 4
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(9449) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_type_t'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(9449) : warning C4024: 'outgoing_ackmsg' : different types for formal and actual parameter 5
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(9469) : warning C4047: 'initializing' : 'sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(9470) : warning C4047: 'initializing' : 'sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(9488) : warning C4013: 'sip_header_remove' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(9611) : warning C4047: '=' : 'sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(9624) : warning C4013: 'sip_retry_after_init' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(9624) : error C2223: left of '->af_delta' must point to struct/union
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(10800) : warning C4047: '=' : 'sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(10834) : warning C4047: 'initializing' : 'sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(10913) : warning C4013: 'sip_rseq_init' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(10918) : error C2065: 'sip_require_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(10918) : warning C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(10918) : warning C4024: 'sip_add_make' : different types for formal and actual parameter 3
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(11081) : warning C4047: '=' : 'sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(11341) : warning C4047: '=' : 'sip_to_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(11427) : warning C4013: 'sip_rack_init' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(11482) : warning C4047: '=' : 'sip_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(11490) : error C2223: left of '->r_url' must point to struct/union
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(11491) : warning C4047: '=' : 'sip_route_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(11497) : warning C4047: '=' : 'sip_route_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(11521) : warning C4013: 'SIPTAG_RACK' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(11521) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(11522) : warning C4047: ':' : 'const tag_type_s *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(11522) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'const tag_type_s *'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta.c(11522) : warning C4024: 'sip_add_tl' : different types for formal and actual parameter 4
1>nta_check.c
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(83) : error C2220: warning treated as error - no 'object' file generated
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(83) : warning C4013: 'SIPTAG_UNSUPPORTED' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(83) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(83) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(84) : warning C4013: 'SIPTAG_SUPPORTED' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(124) : warning C4013: 'SIPTAG_REQUIRE' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(124) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(124) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(125) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_type_t'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(125) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 5
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(169) : warning C4013: 'SIPTAG_ALLOW' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(169) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(169) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(170) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_type_t'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(170) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 5
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(175) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(175) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(176) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_type_t'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(176) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 5
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(262) : warning C4013: 'SIPTAG_ACCEPT' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(262) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(262) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(263) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_type_t'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(263) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 5
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(335) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(335) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(336) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_type_t'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(336) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 5
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(376) : warning C4013: 'sip_min_se_init' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(376) : error C2223: left of '->min_delta' must point to struct/union
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(380) : warning C4013: 'SIPTAG_MIN_SE' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(380) : warning C4047: 'function' : 'tag_type_t' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(380) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 4
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(381) : warning C4047: 'function' : 'tag_value_t' differs in levels of indirection from 'tag_type_t'
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta_check.c(381) : warning C4024: 'nta_incoming_treply' : different types for formal and actual parameter 5
1>nta_tag.c
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta_tag.c(178) : error C2065: 'sip_contact_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta_tag.c(178) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta_tag.c(184) : error C2065: 'sip_contact_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta_tag.c(184) : error C2099: initializer is not a constant
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta_tag.c(191) : error C2065: 'sip_contact_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\nta\nta_tag.c(191) : error C2099: initializer is not a constant
1>nta_tag_ref.c
1>sl_read_payload.c
1>sl_utils_log.c
1>sl_utils_print.c
1>tport.c
1>tport_logging.c
1>tport_stub_sigcomp.c
1>tport_stub_stun.c
1>tport_tag.c
1>tport_tag_ref.c
1>tport_tls.c
1>tport_type_connect.c
1>..\..\sofia-sip\libsofia-sip-ua\tport\tport_type_connect.c(173) : error C2220: warning treated as error - no 'object' file generated
1>..\..\sofia-sip\libsofia-sip-ua\tport\tport_type_connect.c(173) : warning C4013: 'http_request_format' undefined; assuming extern returning int
1>..\..\sofia-sip\libsofia-sip-ua\tport\tport_type_connect.c(173) : warning C4047: '=' : 'http_request_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\tport\tport_type_connect.c(179) : error C2065: 'http_host_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\tport\tport_type_connect.c(179) : warning C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\tport\tport_type_connect.c(179) : warning C4024: 'msg_header_add_make' : different types for formal and actual parameter 3
1>..\..\sofia-sip\libsofia-sip-ua\tport\tport_type_connect.c(180) : error C2065: 'http_separator_class' : undeclared identifier
1>..\..\sofia-sip\libsofia-sip-ua\tport\tport_type_connect.c(180) : warning C4047: 'function' : 'msg_hclass_t *' differs in levels of indirection from 'int'
1>..\..\sofia-sip\libsofia-sip-ua\tport\tport_type_connect.c(180) : warning C4024: 'msg_header_add_make' : different types for formal and actual parameter 3
1>tport_type_tcp.c
1>tport_type_tls.c
1>tport_type_udp.c
1>Generating Code...
1>Compiling...
1>sdp.c
1>sdp_parse.c
1>sdp_print.c
1>sdp_tag.c
1>sdp_tag_ref.c
1>soa.c
1>soa_static.c
1>soa_tag.c
1>soa_tag_ref.c
1>inet_ntop.c
1>Generating Code...
1>Creating browse information file...
1>Microsoft Browse Information Maintenance Utility Version 9.00.21022
1>Copyright (C) Microsoft Corporation. All rights reserved.
1>BSCMAKE: error BK1506 : cannot open file '.\Debug\sip_basic.sbr': No such file or directory
1>Build log was saved at "file://c:\c4dev\freeswitch\libs\win32\sofia\Debug\BuildLog.htm"
1>libsofia_sip_ua_static - 899 error(s), 1735 warning(s)
========== Rebuild All: 0 succeeded, 1 failed, 0 skipped ==========
?
?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110815/22149e47/attachment-0001.html 

From msc at freeswitch.org  Mon Aug 15 23:03:09 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 15 Aug 2011 14:03:09 -0500
Subject: [Freeswitch-users] please help!!! playing tone only when 3rd
 party joins in bridging conference
In-Reply-To: <4E4419D7.4030509@gosilverplus.com>
References: <4E4419D7.4030509@gosilverplus.com>
Message-ID: <CAKzWOxUHgpW8Xv4SfFzZQnTzG93zxvPVO5PwgEsPe5NvT=N9=Q@mail.gmail.com>

You can now modify the enter and exit sounds on a conference. (I know,
because it's pretty much the only thing I've ever added to mod_conference.
:)

conference <conf name> enter_sound on|off|none|file <filename>

and

conference <conf name> exit_sound on|off|none|file <filename>

Start the conference without an enter sound and then turn on the enter sound
after the two parties are connected.

-MC

On Thu, Aug 11, 2011 at 1:05 PM, ran zhang <rzhang at gosilverplus.com> wrote:

> I have a bridging conference established with 2 people in it, and i want
> to play a tone when 3rd party joins in. I can't set the conference's
> 'enter-sound' to play the tone since it will play the tone when first 2
> people establish the bridging conference.
>
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110815/56c2a8b9/attachment.html 

From anthony.minessale at gmail.com  Mon Aug 15 23:04:43 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Aug 2011 14:04:43 -0500
Subject: [Freeswitch-users] Mod event socket error
In-Reply-To: <CALV5c5NZXjzyy6e4kDv7+5X1gZHgwaXNRx_PY5r63Mu7Ts=6DQ@mail.gmail.com>
References: <CALV5c5NZXjzyy6e4kDv7+5X1gZHgwaXNRx_PY5r63Mu7Ts=6DQ@mail.gmail.com>
Message-ID: <CAKbxfG_QFm+U_ZGMpaWA5F3bvYe+DP1xjtwJkQm7yMLOz=qaOw@mail.gmail.com>

it means you are subscribing to events and not reading them off the
socket fast enough.
you must do a continuos uninterrupted loop to read events from a
socket once subscription begins.


On Mon, Aug 15, 2011 at 1:21 PM, Leonardo P. Bidinoto
<leonardo.bidinoto at voicetechnology.com.br> wrote:
> Hi Guys,
>
> Im getting a lot of errors in my test machines, like these:
>
> 2011-08-15 10:12:39.433325 [CRIT] mod_event_socket.c:378 Lost 244 events!
> 2011-08-15 10:12:47.689947 [CRIT] mod_event_socket.c:378 Lost 32 events!
> 2011-08-15 10:12:48.770116 [CRIT] mod_event_socket.c:378 Lost 3 events!
> 2011-08-15 10:12:50.771600 [CRIT] mod_event_socket.c:378 Lost 54 events!
> 2011-08-15 10:12:50.791728 [CRIT] mod_event_socket.c:378 Lost 9 events!
> 2011-08-15 10:12:51.012380 [CRIT] mod_event_socket.c:378 Lost 39 events!
> 2011-08-15 10:12:51.012380 [CRIT] mod_event_socket.c:378 Lost 1 events!
>
> What this kind of error means?
>
> --
> Leonardo Pires Bidinoto
> Voice Technology
> www.voicetechnology.com.br
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From jeff at jefflenk.com  Mon Aug 15 23:26:52 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Mon, 15 Aug 2011 12:26:52 -0700 (PDT)
Subject: [Freeswitch-users] Weird: Sofia-sip failed to be compiled on
 Visual Studio 2008 but it was OK three weeks ago
In-Reply-To: <1313434112.88678.YahooMailClassic@web39707.mail.mud.yahoo.com>
References: <CALV5c5NZXjzyy6e4kDv7+5X1gZHgwaXNRx_PY5r63Mu7Ts=6DQ@mail.gmail.com>
	<1313434112.88678.YahooMailClassic@web39707.mail.mud.yahoo.com>
Message-ID: <1313436412335-6688832.post@n2.nabble.com>

king2kin,

Please do not hijack existing threads always send a new mail. Do not reply
to an existing email and change the subject as this causes lots of confusion
when trying to reply to known topics by readers of this list.

Only by accident did I even see your email. You must set autocrlf=false
before you checkout from git.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Mod-event-socket-error-tp6688617p6688832.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From yungwei at resolvity.com  Mon Aug 15 23:33:35 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Mon, 15 Aug 2011 15:33:35 -0400
Subject: [Freeswitch-users] voicemail is not saved
Message-ID: <33095823FD21DF429B481B5163264B7950FF12FD15@VMBX102.ihostexchange.net>

Hi,

I left several voicemails (Each is longer than 3 sec) to a user account, but none is available when I check the mailbox. 
Relevant settings are listed below. What am I missing here? Thanks.

In conf/autoload_configs/modules.conf.xml, mod_voicemail is already loaded.
freeswitch at internal> load mod_voicemail
+OK Reloading XML
-ERR [Module already loaded]
freeswitch at internal> 2011-08-15 14:32:10.666978 [WARNING] switch_loadable_module.c:998 Module mod_voicemail Already Loaded!

Here's the content of conf/autoload_configs/voicemail.conf.xml:
<configuration name="voicemail.conf" description="Voicemail">
  <settings>
  </settings>
  <profiles>
    <profile name="default">
      <param name="file-extension" value="wav"/>
      <param name="terminator-key" value="#"/>
      <param name="max-login-attempts" value="3"/>
      <param name="digit-timeout" value="10000"/>
      <param name="min-record-len" value="3"/>
      <param name="max-record-len" value="300"/>
      <param name="max-retries" value="3"/>
      <param name="tone-spec" value="%(1000, 0, 640)"/>
      <param name="callback-dialplan" value="XML"/>
      <param name="callback-context" value="default"/>
      <param name="play-new-messages-key" value="1"/>
      <param name="play-saved-messages-key" value="2"/>
      <param name="login-keys" value="0"/>
      <param name="main-menu-key" value="0"/>
      <param name="config-menu-key" value="5"/>
      <param name="record-greeting-key" value="1"/>
      <param name="choose-greeting-key" value="2"/>
      <param name="change-pass-key" value="6"/>
      <param name="record-name-key" value="3"/>
      <param name="record-file-key" value="3"/>
      <param name="listen-file-key" value="1"/>
      <param name="save-file-key" value="2"/>
      <param name="delete-file-key" value="7"/>
      <param name="undelete-file-key" value="8"/>
      <param name="email-key" value="4"/>
      <param name="pause-key" value="0"/>
      <param name="restart-key" value="1"/>
      <param name="ff-key" value="6"/>
      <param name="rew-key" value="4"/>
      <param name="skip-greet-key" value="#"/>
      <param name="previous-message-key" value="1"/>
      <param name="next-message-key" value="3"/>
      <param name="skip-info-key" value="*"/>
      <param name="repeat-message-key" value="0"/>
      <param name="record-silence-threshold" value="200"/>
      <param name="record-silence-hits" value="2"/>
      <param name="web-template-file" value="web-vm.tpl"/>
      <!-- if you need to change the sample rate of the recorded files e.g. gmail voicemail player -->
      <!--<param name="record-sample-rate" value="11025"/>-->
      <!-- the next two both must be set for this to be enabled
           the extension is in the format of <dest> [<dialplan>] [<context>]
       -->
      <param name="operator-extension" value="operator XML default"/>
      <param name="operator-key" value="9"/>
      <param name="vmain-extension" value="vmain XML default"/>
      <param name="vmain-key" value="*"/>
      <!-- playback created files as soon as they were recorded by default -->
      <!--<param name="auto-playback-recordings" value="true"/>-->
      <email>
        <param name="template-file" value="voicemail.tpl"/>
        <param name="notify-template-file" value="notify-voicemail.tpl"/>
        <!-- this is the format voicemail_time will have -->
        <param name="date-fmt" value="%A, %B %d %Y, %I %M %p"/>
        <param name="email-from" value="${voicemail_account}@${voicemail_domain}"/>
      </email>
      <!--<param name="storage-dir" value="/tmp"/>-->
      <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
      <!--<param name="record-comment" value="Your Comment"/>-->
      <!--<param name="record-title" value="Your Title"/>-->
      <!--<param name="record-copyright" value="Your Copyright"/>-->
    </profile>
  </profiles>
</configuration>

In conf/directory/default.xml, user 91000 is defined in domain voicemail_2.
  <domain name="voicemail_2">
    <groups>
      <group name="default">
        <users>
          <user bluebox_id="2" id="91000">
            <params>
              <param name="sip-forbid-register" value="true"/>
              <param name="vm-password" value="xxxx"/>
              <param name="vm-message-ext" value="wav"/>
              <param name="vm-mailto" value="xxx at xxx.com"/>
              <param name="email-addr" value="xxx at xxx.com"/>
              <param name="vm-delete-file" value="false"/>
              <param name="vm-attach-file" value="true"/>
              <param name="vm-email-all-messages" value="true"/>
            </params>
            <variables>
              <variable name="timezone" value="America/Chicago"/>
            </variables>
          </user>
        </users>
      </group>
    </groups>
  </domain>

In my dialplan, calls to 1112223333 will be sent to user 91000's voicemail box if they are not answered.
    <extension name="main_number_2" continue="true">
      <condition field="destination_number" expression="^1112223333$">
        <action application="set" bluebox="settingEndBridge" data="hangup_after_bridge=true"/>
        <action application="set" bluebox="settingFail" data="continue_on_fail=true"/>
        <action application="set" bluebox="autoPlay" data="vm_auto_play=false"/>
        <action application="set" bluebox="settingTimeout" data="call_timeout=30"/>
        <action application="set" bluebox="settingRing" data="ringback=${us-ring}"/>
        <action application="set" bluebox="settingRingXfer" data="transfer_ringback=${us-ring}"/>
        <action application="export" bluebox="sipCalleeIdName" data="sip_callee_id_name=m1"/>
        <action application="export" bluebox="sipCalleeIdNumber" data="sip_callee_id_number=1112223333"/>
        <action application="bridge" data="user/91000@$${location_3}"/>
        <action application="answer"/>
        <action application="voicemail" data="default voicemail_2 91000"/>
        <action application="hangup"/>
      </condition>
    </extension>


From anthony.minessale at gmail.com  Tue Aug 16 01:52:07 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 15 Aug 2011 16:52:07 -0500
Subject: [Freeswitch-users] odbc basic_calls
In-Reply-To: <CALNZuw=-PhE-OK38h8YPQ=00O+BPe3YE=C=XWVCXOUS+qX-W4Q@mail.gmail.com>
References: <CALNZuwk20r8=rmu9rOOd-aChOS6XOGVm7mxKNhd1-DnC+id7Hg@mail.gmail.com>
	<CAKbxfG-8E=5Do_mLf62tokgL5eN8=LfH954tUNRawcNCtmMTLA@mail.gmail.com>
	<CALNZuwkwgFmdmWxYMzkzGeW8u-TiRg2SbeJ1zFpOfxEPiyS2xg@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59EBABB8AD@cooper>
	<CALNZuw=uWC8kGww5Gjo_P7mxJ_QyrCDB1ipmamrG4QXVXb8TwQ@mail.gmail.com>
	<CALNZuw=-PhE-OK38h8YPQ=00O+BPe3YE=C=XWVCXOUS+qX-W4Q@mail.gmail.com>
Message-ID: <CAKbxfG8Y7Cdzp4oGUNTNopErnFaF6evnsKQ9dftPSu_WU3bDNA@mail.gmail.com>

I think i fixed it in tree now so it will

On Sun, Aug 14, 2011 at 12:00 PM, Avi Marcus <avi at avimarcus.net> wrote:
> I dropped and recreated the channels table, which doesn't have callee_name
> but sent_callee_name now. Then when I created the view manually, it worked,
> but none of this got triggered to happen automatically.
> -Avi
>
>
> On Fri, Aug 12, 2011 at 3:05 PM, Avi Marcus <avi at avimarcus.net> wrote:
>>
>> Error in query: Unknown column 'a.sent_callee_name' in 'field list'
>> So table channels.. no, I don't see that in there. I see?callee_name?in
>> there though. Did it miss some channel table alterations?
>> Huh, tables complete, db_data, interfaces and a few more all seem new.
>> I've been running odbc in mysql for a year? but I've never seen these until
>> now.
>> -Avi Marcus
>>
>> p.s. move this to jira now that it's back up..?
>>
>> On Thu, Aug 11, 2011 at 5:02 PM, Peter Olsson
>> <peter.olsson at visionutveckling.se> wrote:
>>>
>>> Could you try to create it manually? The view's definition is in
>>> switch_core_sqldb.c on line 1721.
>>>
>>> /Peter
>>> ________________________________________
>>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
>>> [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Avi Marcus
>>> [avi at avimarcus.net]
>>> Skickat: den 11 augusti 2011 15:48
>>> Till: FreeSWITCH Users Help
>>> ?mne: Re: [Freeswitch-users] odbc basic_calls
>>>
>>> I don't see anything in the startup log about it checking the tables:
>>> http://pastebin.freeswitch.org/17013
>>> This happens when I do "show calls". I know you changed something for
>>> show calls to even show 1 legged IVRs since my last update, but not having
>>> looked at the code, I don't see how that would be related.
>>> I haven't heard of this basic_calls table before today.
>>>
>>> -Avi Marcus
>>>
>>> On Thu, Aug 11, 2011 at 4:33 PM, Anthony Minessale
>>> <anthony.minessale at gmail.com<mailto:anthony.minessale at gmail.com>> wrote:
>>> its a view that should be created? perhaps there is an error on
>>> startup creating the view in mysuckwell?
>>>
>>>
>>> On Thu, Aug 11, 2011 at 6:28 AM, Avi Marcus
>>> <avi at avimarcus.net<mailto:avi at avimarcus.net>> wrote:
>>> > I just upgraded FS since.. 7 weeks ago I think. Now: FreeSWITCH Version
>>> > 1.0.head (git-9d98d49 2011-08-10 08-38-55 -0500)
>>> > While testing the new build clean, when I hit F4 for show calls, I now
>>> > get:
>>> > freeswitch at internal> 2011-08-11 14:24:04.574797 [ERR]
>>> > switch_core_sqldb.c:825 ERR: [select * from basic_calls where
>>> > hostname='sip2' order by call_created_epoch]
>>> > [STATE: 42S02 CODE 1146 ERROR: [unixODBC][MySQL][ODBC 3.51
>>> > Driver][mysqld-5.1.41-3ubuntu12.10-log]Table 'freeswitch.basic_calls'
>>> > doesn't exist
>>> > Aren't the core odbc tables supposed to be auto-created?
>>> > -Avi
>>> > _______________________________________________
>>> > Join us at ClueCon 2011, Aug 9-11, Chicago
>>> > http://www.cluecon.com 877-7-4ACLUE
>>> >
>>> > FreeSWITCH-users mailing list
>>> >
>>> > FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>>
>>> MSN:anthony_minessale at hotmail.com<mailto:MSN%3Aanthony_minessale at hotmail.com>
>>>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>>
>>> sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org>
>>>
>>> googletalk:conf+888 at conference.freeswitch.org<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:+19193869900<tel:%2B19193869900>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>>
>>> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> !DSPAM:4e43de1032765001342651!
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From lakersman2006 at yahoo.com  Tue Aug 16 02:17:19 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Mon, 15 Aug 2011 15:17:19 -0700 (PDT)
Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
In-Reply-To: <CAKbxfG_802p+G1a9HtLEh_i-K1C06fpZgyR4_NP=jZJR68YK8g@mail.gmail.com>
References: <4E4164C0.8030507@tiendalinux.com>
	<1312937649.7702.YahooMailNeo@web161011.mail.bf1.yahoo.com>
	<CAKbxfG_802p+G1a9HtLEh_i-K1C06fpZgyR4_NP=jZJR68YK8g@mail.gmail.com>
Message-ID: <1313446639.81086.YahooMailNeo@web161008.mail.bf1.yahoo.com>

Anthony,

My gripe was not about simply having a DIALSTATUS variable in Freeswitch which copies what is from "originate_disposition" what I wanted is to be able to get the status of the B-Leg because right now when early media is played (which i wanted)? "originate_disposition" shows "ANSWER" which I think is caused by me explitly called the "answer" app in my dialplan before the bridge app, this is because my DID provider requires an answer/sip 200 or else it will keep re-sending the sip invite, therefore causing freeswitch to keep creating new channels. All I want is to be able to get the proper sip/hangup/dial statuses of the B-leg.



________________________________
From: Anthony Minessale <anthony.minessale at gmail.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, August 10, 2011 8:52 AM
Subject: Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide


=D?

ok, sure. ?If that's your only complaint.... see commit?9d98d49f0556fb79656c8403f285ae0a615439d3



Some caveats

1) There is actually less?specific, more generalized data in this DIALSTATUS variable than what we already report, when you're ready to move on see the originate_disposition variable: ?It's kind of like going from reporting the precise geo-location of a cafe in Paris to generalizing it to "EUROPE"?

We follow the Q.850 standard for call cause codes and follow the SIP RFC to map sip response codes to/from the Q.850?equivalent. ?Also each module has its own version "sip_hangup_disposition" for sip so you have both the real sip response code AND the official Q.850 equiv variables set on each call.


2) We don't have a torture feature so we never return that code.


3) Since our originate can return before a call is answered I added "EARLY" which means the originate succeeded but its still not answered.

4) For any others that do not map directly to FreeSWITCH, I just installed a copy of originate_disposition for good measure.

P.S?

This email took longer to compose than the patch did while sitting in the middle of a crowded room so you probably could have simply parsed the originate originate_disposition yourself but if it helps people get over being stuck in a?paradigm, it's worth it for me to write........
?


On Tue, Aug 9, 2011 at 7:54 PM, Sam <lakersman2006 at yahoo.com> wrote:

I find that Asterisk's AGI is much easier to use, they allow you to retrieve the dial status much easier than freeswitch's api's. Come on freeswitch, if you want to be better than asterisk, you should make it easier to get the dialstatus, etc. At this point asterisk is still defacto.
>
>
>
>
>________________________________
> From: Nestor A Diaz <nestor at tiendalinux.com>
>To: freeswitch-users at lists.freeswitch.org
>Sent: Tuesday, August 9, 2011 9:48 AM
>Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
>
>
>Hi Guys.
>
>I am starting to use FreeSWITCH, i am an asterisk user since the
1.0.7
release appears as a package on the debian distribution, at the
beginning i was amazed by the fact i can build a PBX for my own
business
and i did, later i began to install this system for my customers and
sooner i meet the problems, however being the software open source i
always find a way to fix things using patchs from others, sometimes i
felt how my life was at risk when the system stops working and that
usually happens when i have to use queues and dealing with digium
hardware.
>
>Fixing those problems either by applying patches or by changing
the
hardware where the digium cards were supposed to be installed helps me,
but that was to much stress for me and seeking for a balance that will
let me invest more time on services, configuration and hoping to have
better hardware
options brings me to freeswitch.
>
>I agree with freeswitch philosophy that instead of having
thousands of
modules
that don't work fine i prefer just a few that works the way it should
be, a rock solid system for a corporate pbx and a call center is what i
want.
>
>So here i am trying to begin the conversion, and i hope the
information
we can transcript in this list will help others that want to try
another
alternative to asterisk.
>
>First of all i think the saner for a migration is to have the two
systems
running either on the same machine or different and use the stable
features of each one.
>
>So could you please freeswitch users help me with this rosetta
stone migration guide in order to post it to voip-info.org or
freeswitch wiki (i list only the ones i currently use ):
>
>
>
>Technology Asterisk Freeswitch 
>PSTN Connectivity (Digium /
Sangoma) dahdi freetdm 
>IAX2 mod_iax ?? none stable yet.
>Use Asterisk to forward traffic via SIP.
>Enable Hardware HPET for IAX2 trunk if card not available for Asterisk  
>Bluetooth Channel chan_mobile ??
>Use asterisk via SIP
> 
>Skype Skypeforasterisk (no longer for sale) mod_skypeopen 
>CDR Stadistics Arternic cdr-stats  ?? 
>Queue Statistics  Asteriskguru queue-stats ?? 
>Web Management Freepbx ?? 
>IVR AGI / AMI Event Socket 
>Codec G.729 Transcodind Cards
>G.729 licenses
>Free G.729 (Intel IPP) Transcodind Cards
>G.729 licenses
>fsg729 Intel IPP(any experience with it ? ) 
>Fax Handling Iaxmodem with Hylafax ??
>Iaxmodem via asterisk to FS via SIP ?
> 
>SIP chan_sip sofia 
>ACD app_queue mod_callcenter 
>
>Thank you all
>
>
>-- 
>Nestor A. Diaz
>Ingeniero de Sistemas
>Tel. +57 1-485-3020 x 211
>Cel. +57 316-227-3593
>Tel. SIP: sip:211 at tiendalinux.com
>Email/MSN: nestor at tiendalinux.com
>http://www.tiendalinux.com/
>Bogota, Colombia 
> 
>
>
>_______________________________________________
>Join us at ClueCon 2011, Aug 9-11, Chicago
>http://www.cluecon.com 877-7-4ACLUE
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>
>
>_______________________________________________
>Join us at ClueCon 2011, Aug 9-11, Chicago
>http://www.cluecon.com 877-7-4ACLUE
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900

_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110815/f5c61a9d/attachment-0001.html 

From lakersman2006 at yahoo.com  Tue Aug 16 02:35:14 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Mon, 15 Aug 2011 15:35:14 -0700 (PDT)
Subject: [Freeswitch-users] ORIGINATE_DISPOSITION
Message-ID: <1313447714.89555.YahooMailNeo@web161001.mail.bf1.yahoo.com>

For the ORIGINATE_DISPOSITION channel variable, does "SUCCESS" mean the call was ANSWERED?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110815/492f2123/attachment.html 

From db3l.net at gmail.com  Tue Aug 16 02:48:30 2011
From: db3l.net at gmail.com (David Bolen)
Date: Mon, 15 Aug 2011 18:48:30 -0400
Subject: [Freeswitch-users] DTMF bleed through in conference
Message-ID: <m2liuuz3y9.fsf@valheru.db3l.homeip.net>

I'm encountering what I would probably best describe as some bleed
through of inbound DTMF when in a conference, and I was wondering if
anyone might have suggestions for likely causes or troubleshooting?

DTMF detection itself seems fine, and appears to properly be using
rfc2833 (the SDPs all show include "a=rtpmap:101 telephone-event/8000").
I get "RTP RECV DTMF" logs for each digit that is recognized.
However, a reasonable percentage of the time, some small fraction of
the DTMF appears to be heard by the conference as a whole.  Never the
whole thing, and sometimes it's a bit distorted (though that may just
be due to it being so short).  Almost as if a small portion of a sound
sample makes it through.

For testing, I'm using internal phones with direct SIP connections to
the switch so no PSTN or gateways involved, though it also occurs with
a PSTN caller coming in through multiple gateways.  For the internal
phones, I've tried both a physical phone through a PAP2T, as well as a
PC soft phone.

This is with git-868d823 from Jul 29, so a few weeks old but reviewing
the more recent commits there weren't any seemingly related to DTMF or
conferencing.

Any suggestions on places to look or things to try would be most
appreciated.

-- David



From covici at ccs.covici.com  Tue Aug 16 02:57:58 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Mon, 15 Aug 2011 18:57:58 -0400
Subject: [Freeswitch-users] how do I fix perl interface to streamFile?
Message-ID: <26749.1313449078@ccs.covici.com>

I want to stream a file and have the dtmf digits move forward and
backward in the file by a certain number of seconds.  I first tried to
set n inputcallback function and use uuid_fileman, but that does not
work because when the digit is received, the file is actually stopped,
so the seek is meaningless.  Now the streamFile itself has provision for
this, except that the perl interface just has the file and the sample
count, omitting the callback function and its args.  What do I have to
change to putthe args in, so I can get this to work?

Thanks in advance for any suggestions.

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From msc at freeswitch.org  Tue Aug 16 03:00:38 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 15 Aug 2011 16:00:38 -0700
Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
In-Reply-To: <1313446639.81086.YahooMailNeo@web161008.mail.bf1.yahoo.com>
References: <4E4164C0.8030507@tiendalinux.com>
	<1312937649.7702.YahooMailNeo@web161011.mail.bf1.yahoo.com>
	<CAKbxfG_802p+G1a9HtLEh_i-K1C06fpZgyR4_NP=jZJR68YK8g@mail.gmail.com>
	<1313446639.81086.YahooMailNeo@web161008.mail.bf1.yahoo.com>
Message-ID: <CAKzWOxUUK0ODV0vFoPnmL7BAUi1PsDqZnYCfrNbQaf4pXRMS1g@mail.gmail.com>

On Mon, Aug 15, 2011 at 3:17 PM, Sam <lakersman2006 at yahoo.com> wrote:

> Anthony,
>
> My gripe was not about simply having a DIALSTATUS variable in Freeswitch
> which copies what is from "originate_disposition" what I wanted is to be
> able to get the status of the B-Leg because right now when early media is
> played (which i wanted)  "originate_disposition" shows "ANSWER" which I
> think is caused by me explitly called the "answer" app in my dialplan before
> the bridge app, this is because my DID provider requires an answer/sip 200
> or else it will keep re-sending the sip invite, therefore causing freeswitch
> to keep creating new channels. All I want is to be able to get the proper
> sip/hangup/dial statuses of the B-leg.
>

I think we understand where you are coming from. I believe in my other post
that I mentioned that the bridge app and a few other channel variables will
most likely let you tailor your dialplan to your exact needs without
explicitly needing to poll the "dial status" of the b-leg. In fact, I think
I left out a few options:

execute_on_answer
execute_on_ring
execute_on_media

Those channel variables are quite handy, especially the execute_on_answer.
If you execute a dp transfer when the b leg is answered then whatever
happens after your bridge (or originate API if you are doing that) will
always be some sort of failed call attempt. There's also a handy
"transfer_on_fail" channel variable that lets you explicitly send the call
to another dp extension on a failed bridge attempt.

It may seem unintuitive to be transferring the calls all over the dialplan,
but if you think about it you can create contexts to handle specific
scenarios and then you're done. Call failed? Transfer to "CALL_FAILED"
context and process. Call was answered? Transfer to "CALL_ANSWERED" context
and process. No scripting required, and it's really fast.

Just my $0.02...

-MC



> ------------------------------
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Wednesday, August 10, 2011 8:52 AM
> *Subject:* Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
> =D
>
> ok, sure.  If that's your only complaint.... see
> commit 9d98d49f0556fb79656c8403f285ae0a615439d3
>
>
>
> Some caveats
>
> 1) There is actually less specific, more generalized data in this
> DIALSTATUS variable than what we already report, when you're ready to move
> on see the originate_disposition variable:  It's kind of like going from
> reporting the precise geo-location of a cafe in Paris to generalizing it to
> "EUROPE"
>
> We follow the Q.850 standard for call cause codes and follow the SIP RFC to
> map sip response codes to/from the Q.850 equivalent.  Also each module has
> its own version "sip_hangup_disposition" for sip so you have both the real
> sip response code AND the official Q.850 equiv variables set on each call.
>
>
> 2) We don't have a torture feature so we never return that code.
>
>
> 3) Since our originate can return before a call is answered I added "EARLY"
> which means the originate succeeded but its still not answered.
>
> 4) For any others that do not map directly to FreeSWITCH, I just installed
> a copy of originate_disposition for good measure.
>
> P.S
>
> This email took longer to compose than the patch did while sitting in the
> middle of a crowded room so you probably could have simply parsed the
> originate originate_disposition yourself but if it helps people get over
> being stuck in a paradigm, it's worth it for me to write........
>
>
> On Tue, Aug 9, 2011 at 7:54 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> I find that Asterisk's AGI is much easier to use, they allow you to
> retrieve the dial status much easier than freeswitch's api's. Come on
> freeswitch, if you want to be better than asterisk, you should make it
> easier to get the dialstatus, etc. At this point asterisk is still defacto.
>
> ------------------------------
> *From:* Nestor A Diaz <nestor at tiendalinux.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Tuesday, August 9, 2011 9:48 AM
> *Subject:* [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
> Hi Guys.
>
> I am starting to use FreeSWITCH, i am an asterisk user since the 1.0.7
> release appears as a package on the debian distribution, at the beginning i
> was amazed by the fact i can build a PBX for my own business and i did,
> later i began to install this system for my customers and sooner i meet the
> problems, however being the software open source i always find a way to fix
> things using patchs from others, sometimes i felt how my life was at risk
> when the system stops working and that usually happens when i have to use
> queues and dealing with digium hardware.
>
> Fixing those problems either by applying patches or by changing the
> hardware where the digium cards were supposed to be installed helps me, but
> that was to much stress for me and seeking for a balance that will let me
> invest more time on services, configuration and hoping to have better
> hardware options brings me to freeswitch.
>
> I agree with freeswitch philosophy that instead of having thousands of
> modules that don't work fine i prefer just a few that works the way it
> should be, a rock solid system for a corporate pbx and a call center is what
> i want.
>
> So here i am trying to begin the conversion, and i hope the information we
> can transcript in this list will help others that want to try another
> alternative to asterisk.
>
> First of all i think the saner for a migration is to have the two systems
> running either on the same machine or different and use the stable features
> of each one.
>
> So could you please freeswitch users help me with this rosetta stone
> migration guide in order to post it to voip-info.org or freeswitch wiki (i
> list only the ones i currently use ):
>
>
>   *Technology* *Asterisk* *Freeswitch*  PSTN Connectivity (Digium /
> Sangoma) dahdi freetdm  IAX2 mod_iax ?? none stable yet.
> Use Asterisk to forward traffic via SIP.
> Enable Hardware HPET for IAX2 trunk if card not available for Asterisk  Bluetooth
> Channel chan_mobile ??
> Use asterisk via SIP
>   Skype Skypeforasterisk (no longer for sale) mod_skypeopen  CDR
> Stadistics Arternic cdr-stats ??  Queue Statistics  Asteriskguru
> queue-stats ??  Web Management Freepbx ??  IVR AGI / AMI Event Socket  Codec
> G.729 Transcodind Cards
> G.729 licenses
> Free G.729 (Intel IPP) Transcodind Cards
> G.729 licenses
> fsg729 Intel IPP(any experience with it ? )  Fax Handling Iaxmodem with
> Hylafax ??
> Iaxmodem via asterisk to FS via SIP ?
>   SIP chan_sip sofia  ACD app_queue mod_callcenter
>
> Thank you all
>
>
> --
> Nestor A. Diaz
> Ingeniero de Sistemas
> Tel. +57 1-485-3020 x 211
> Cel. +57 316-227-3593
> Tel. SIP: sip:211 at tiendalinux.com
> Email/MSN: nestor at tiendalinux.com
> http://www.tiendalinux.com/
> Bogota, Colombia
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110815/630d8642/attachment-0001.html 

From msc at freeswitch.org  Tue Aug 16 03:09:41 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 15 Aug 2011 16:09:41 -0700
Subject: [Freeswitch-users] DTMF bleed through in conference
In-Reply-To: <m2liuuz3y9.fsf@valheru.db3l.homeip.net>
References: <m2liuuz3y9.fsf@valheru.db3l.homeip.net>
Message-ID: <CAKzWOxU_oaWg1_c=92W30g-pZyWkjWWW2YV5bh_hG-r1mZdVZA@mail.gmail.com>

Are the devices that send the DTMFs using 2833? Sounds like possibly they
are using inband. I would get a pcap with RTP of the conference and see
what, exactly, the device is sending. Some tips for this can be found here:

http://wiki.freeswitch.org/wiki/Packet_Capture

-MC

On Mon, Aug 15, 2011 at 3:48 PM, David Bolen <db3l.net at gmail.com> wrote:

> I'm encountering what I would probably best describe as some bleed
> through of inbound DTMF when in a conference, and I was wondering if
> anyone might have suggestions for likely causes or troubleshooting?
>
> DTMF detection itself seems fine, and appears to properly be using
> rfc2833 (the SDPs all show include "a=rtpmap:101 telephone-event/8000").
> I get "RTP RECV DTMF" logs for each digit that is recognized.
> However, a reasonable percentage of the time, some small fraction of
> the DTMF appears to be heard by the conference as a whole.  Never the
> whole thing, and sometimes it's a bit distorted (though that may just
> be due to it being so short).  Almost as if a small portion of a sound
> sample makes it through.
>
> For testing, I'm using internal phones with direct SIP connections to
> the switch so no PSTN or gateways involved, though it also occurs with
> a PSTN caller coming in through multiple gateways.  For the internal
> phones, I've tried both a physical phone through a PAP2T, as well as a
> PC soft phone.
>
> This is with git-868d823 from Jul 29, so a few weeks old but reviewing
> the more recent commits there weren't any seemingly related to DTMF or
> conferencing.
>
> Any suggestions on places to look or things to try would be most
> appreciated.
>
> -- David
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110815/2b52d445/attachment.html 

From gcd at i.ph  Tue Aug 16 03:21:03 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Tue, 16 Aug 2011 07:21:03 +0800
Subject: [Freeswitch-users] Help with dual IP gateways
In-Reply-To: <005501cc5b67$be0ace00$3a206a00$@yahoo.com>
References: <005501cc5b67$be0ace00$3a206a00$@yahoo.com>
Message-ID: <CAHKqMb54iA2Wrfrd4DN7Qcuk8gPxeX+u9AB-Wx2-trO=QC80NQ@mail.gmail.com>

hi lars,

what you need is bandwidth management and QoS features in your router. i
changed my router firmware with DD-WRT and tailored the ff:

1. SIP and esp RTP are given top priority for transmission
2. Web traffice are given lesser priority, P2P the last priority
3. ACK packets are also given top priority to prevent download overflow

using scripts I found in the web.

another option is to use VLAN to segregate your network. Data VLAN goes to
one router and the Voice VLAN to another router.

-nandy


On Tue, Aug 16, 2011 at 12:23 AM, Lars Zeb <larclap at yahoo.com> wrote:

> Currently my LAN is connected to the internet via DSL. The FreeSWITCH box
> is
> on this subnet. To save money, I am moving the data portion of my LAN to a
> new ISP and I want to segregate the VOIP to another ISP. I am tired of
> having a bad VOIP connection during lengthy downloads.
>
> My VOIP and FreeSWITCH skills are minimal. I have used FreeSWITCH for over
> a
> year in a home/business environment. The only reason it is working is with
> the help of this list.
>
> My knowledge of IP is similar. I do not know how to setup a LAN with two
> gateways with all nodes seeing one another. I do want to be able to call
> out
> via FreeSWITCH from a softphone on the data portion of the new LAN.
>
> A friend suggested I need a dual ported WAN firewall/router with load
> balancing to enable all the nodes to be on the same subnet. Can anyone help
> me with suggestions? Is there a consultant I can hire to help with this?
>
> Thanks, Lars
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110816/4b4372e8/attachment.html 

From msc at freeswitch.org  Tue Aug 16 03:31:10 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 15 Aug 2011 16:31:10 -0700
Subject: [Freeswitch-users] ORIGINATE_DISPOSITION
In-Reply-To: <1313447714.89555.YahooMailNeo@web161001.mail.bf1.yahoo.com>
References: <1313447714.89555.YahooMailNeo@web161001.mail.bf1.yahoo.com>
Message-ID: <CAKzWOxVwDaxsFJAr1dVP5nfmDuFXK2tptvai+oqtwZgph3u1kw@mail.gmail.com>

I think you may be wanting "endpoint_disposition" depending on exactly what
you're looking at.

-MC

On Mon, Aug 15, 2011 at 3:35 PM, Sam <lakersman2006 at yahoo.com> wrote:

> For the ORIGINATE_DISPOSITION channel variable, does "SUCCESS" mean the
> call was ANSWERED?
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110815/41777b4f/attachment.html 

From gcd at i.ph  Tue Aug 16 03:39:37 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Tue, 16 Aug 2011 07:39:37 +0800
Subject: [Freeswitch-users] Help with dual IP gateways
In-Reply-To: <CAHKqMb54iA2Wrfrd4DN7Qcuk8gPxeX+u9AB-Wx2-trO=QC80NQ@mail.gmail.com>
References: <005501cc5b67$be0ace00$3a206a00$@yahoo.com>
	<CAHKqMb54iA2Wrfrd4DN7Qcuk8gPxeX+u9AB-Wx2-trO=QC80NQ@mail.gmail.com>
Message-ID: <CAHKqMb66jRafAtwO9W52+x=9WzsZrDnYOkBrb88-X_HKbMZfzA@mail.gmail.com>

more ...

QoS (DSCP) values must be set to prioritize voice packets in the Internet.

-nandy

On Tue, Aug 16, 2011 at 7:21 AM, Nandy Dagondon <gcd at i.ph> wrote:

> hi lars,
>
> what you need is bandwidth management and QoS features in your router. i
> changed my router firmware with DD-WRT and tailored the ff:
>
> 1. SIP and esp RTP are given top priority for transmission
> 2. Web traffice are given lesser priority, P2P the last priority
> 3. ACK packets are also given top priority to prevent download overflow
>
> using scripts I found in the web.
>
> another option is to use VLAN to segregate your network. Data VLAN goes to
> one router and the Voice VLAN to another router.
>
> -nandy
>
>
>
> On Tue, Aug 16, 2011 at 12:23 AM, Lars Zeb <larclap at yahoo.com> wrote:
>
>> Currently my LAN is connected to the internet via DSL. The FreeSWITCH box
>> is
>> on this subnet. To save money, I am moving the data portion of my LAN to a
>> new ISP and I want to segregate the VOIP to another ISP. I am tired of
>> having a bad VOIP connection during lengthy downloads.
>>
>> My VOIP and FreeSWITCH skills are minimal. I have used FreeSWITCH for over
>> a
>> year in a home/business environment. The only reason it is working is with
>> the help of this list.
>>
>> My knowledge of IP is similar. I do not know how to setup a LAN with two
>> gateways with all nodes seeing one another. I do want to be able to call
>> out
>> via FreeSWITCH from a softphone on the data portion of the new LAN.
>>
>> A friend suggested I need a dual ported WAN firewall/router with load
>> balancing to enable all the nodes to be on the same subnet. Can anyone
>> help
>> me with suggestions? Is there a consultant I can hire to help with this?
>>
>> Thanks, Lars
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110816/1b7d0952/attachment.html 

From lakersman2006 at yahoo.com  Tue Aug 16 03:49:14 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Mon, 15 Aug 2011 16:49:14 -0700 (PDT)
Subject: [Freeswitch-users] ORIGINATE_DISPOSITION
In-Reply-To: <CAKzWOxVwDaxsFJAr1dVP5nfmDuFXK2tptvai+oqtwZgph3u1kw@mail.gmail.com>
References: <1313447714.89555.YahooMailNeo@web161001.mail.bf1.yahoo.com>
	<CAKzWOxVwDaxsFJAr1dVP5nfmDuFXK2tptvai+oqtwZgph3u1kw@mail.gmail.com>
Message-ID: <1313452154.28177.YahooMailNeo@web161001.mail.bf1.yahoo.com>

I want to know the B-leg status of the call.



________________________________
From: Michael Collins <msc at freeswitch.org>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Monday, August 15, 2011 4:31 PM
Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION


I think you may be wanting "endpoint_disposition" depending on exactly what you're looking at.

-MC


On Mon, Aug 15, 2011 at 3:35 PM, Sam <lakersman2006 at yahoo.com> wrote:

For the ORIGINATE_DISPOSITION channel variable, does "SUCCESS" mean the call was ANSWERED?
>
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>


FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110815/312725ac/attachment-0001.html 

From gcd at i.ph  Tue Aug 16 03:50:34 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Tue, 16 Aug 2011 07:50:34 +0800
Subject: [Freeswitch-users] DTMF issue when using execute_extension with
	play_and_get_digits
In-Reply-To: <CAA1Qw2sLHtu8ZHQn7GJFOs=f2k9H2pLa+mgu6Kt+FEtGcwWBOw@mail.gmail.com>
References: <CAA1Qw2sLHtu8ZHQn7GJFOs=f2k9H2pLa+mgu6Kt+FEtGcwWBOw@mail.gmail.com>
Message-ID: <CAHKqMb4JOtHDKoQJB3nop9aQs1KOL8XX=K03wnnpmtP8NrYqeg@mail.gmail.com>

perhaps you forgot the "extension" objects!

<extension name="bind-5-app">
   <condition field="destination_number" expression="^4567$"
break="on-true">
<action application="play_and_get_digits" data="1 5 3 7000 #
'/voice_files/A.wav' '/voice_files/failed.wav' dtmf_received \d+"/>
   </condition>
</extension>

<extension name="catch-bind-5">
   <condition field="destination_number" expression="^.*$" break="on-true">
<action application="answer"/>
<action application="bind_meta_app" data="5 b o execute_extension::4567 XML
default"/>
<action application="bridge" data="freetdm/1/a/9952248266"/>
<action application="park"/>
   </condition>
</extension>

On Tue, Aug 9, 2011 at 11:07 PM, Nagalenoj H. <nagalenoj at gmail.com> wrote:

> Hi Friends,
>      Facing an issue when using bind_meta_app and execute_extension(with
> play_and_get_digits) combined.
>
> Here is my dialplan,
> <condition field="destination_number" expression="^4567$" break="on-true">
> <action application="play_and_get_digits" data="1 5 3 7000 #
> '/voice_files/A.wav' '/voice_files/failed.wav' dtmf_received \d+"/>
> </condition>
> <condition field="destination_number" expression="^.*$" break="on-true">
> <action application="answer"/>
> <action application="bind_meta_app" data="5 b o execute_extension::4567 XML
> default"/>
> <action application="bridge" data="freetdm/1/a/9952248266"/>
> <action application="park"/>
> </condition>
>
> So, when callee enters *5,  I want the caller to enter a number. I get the
> extension executed as expected. The caller is able to hear the voice file
> played and when he enters the digits, it is not received. Digits are not
> even present in FS log.
>
> In the normal cases, there is no issues in getting DTMFs. I don't know,
> what am I doing wrong here. Kindly, help me to resolve this.
>
> --
> Regards,
> Nagalenoj H.
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110816/1d187f47/attachment.html 

From gcd at i.ph  Tue Aug 16 03:52:06 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Tue, 16 Aug 2011 07:52:06 +0800
Subject: [Freeswitch-users] voicemail is not saved
In-Reply-To: <33095823FD21DF429B481B5163264B7950FF12FD15@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B7950FF12FD15@VMBX102.ihostexchange.net>
Message-ID: <CAHKqMb6=znNGaG0pDOj_AbaZM-P1=JOJ2hCH3PDZ0Pqwm5CUEg@mail.gmail.com>

check the directory/file permissions
-nandy

On Tue, Aug 16, 2011 at 3:33 AM, Yungwei Chen <yungwei at resolvity.com> wrote:

> Hi,
>
> I left several voicemails (Each is longer than 3 sec) to a user account,
> but none is available when I check the mailbox.
> Relevant settings are listed below. What am I missing here? Thanks.
>
> In conf/autoload_configs/modules.conf.xml, mod_voicemail is already loaded.
> freeswitch at internal> load mod_voicemail
> +OK Reloading XML
> -ERR [Module already loaded]
> freeswitch at internal> 2011-08-15 14:32:10.666978 [WARNING]
> switch_loadable_module.c:998 Module mod_voicemail Already Loaded!
>
> Here's the content of conf/autoload_configs/voicemail.conf.xml:
> <configuration name="voicemail.conf" description="Voicemail">
>  <settings>
>  </settings>
>  <profiles>
>    <profile name="default">
>      <param name="file-extension" value="wav"/>
>      <param name="terminator-key" value="#"/>
>      <param name="max-login-attempts" value="3"/>
>      <param name="digit-timeout" value="10000"/>
>      <param name="min-record-len" value="3"/>
>      <param name="max-record-len" value="300"/>
>      <param name="max-retries" value="3"/>
>      <param name="tone-spec" value="%(1000, 0, 640)"/>
>      <param name="callback-dialplan" value="XML"/>
>      <param name="callback-context" value="default"/>
>      <param name="play-new-messages-key" value="1"/>
>      <param name="play-saved-messages-key" value="2"/>
>      <param name="login-keys" value="0"/>
>      <param name="main-menu-key" value="0"/>
>      <param name="config-menu-key" value="5"/>
>      <param name="record-greeting-key" value="1"/>
>      <param name="choose-greeting-key" value="2"/>
>      <param name="change-pass-key" value="6"/>
>      <param name="record-name-key" value="3"/>
>      <param name="record-file-key" value="3"/>
>      <param name="listen-file-key" value="1"/>
>      <param name="save-file-key" value="2"/>
>      <param name="delete-file-key" value="7"/>
>      <param name="undelete-file-key" value="8"/>
>      <param name="email-key" value="4"/>
>      <param name="pause-key" value="0"/>
>      <param name="restart-key" value="1"/>
>      <param name="ff-key" value="6"/>
>      <param name="rew-key" value="4"/>
>      <param name="skip-greet-key" value="#"/>
>      <param name="previous-message-key" value="1"/>
>      <param name="next-message-key" value="3"/>
>      <param name="skip-info-key" value="*"/>
>      <param name="repeat-message-key" value="0"/>
>      <param name="record-silence-threshold" value="200"/>
>      <param name="record-silence-hits" value="2"/>
>      <param name="web-template-file" value="web-vm.tpl"/>
>      <!-- if you need to change the sample rate of the recorded files e.g.
> gmail voicemail player -->
>      <!--<param name="record-sample-rate" value="11025"/>-->
>      <!-- the next two both must be set for this to be enabled
>           the extension is in the format of <dest> [<dialplan>] [<context>]
>       -->
>      <param name="operator-extension" value="operator XML default"/>
>      <param name="operator-key" value="9"/>
>      <param name="vmain-extension" value="vmain XML default"/>
>      <param name="vmain-key" value="*"/>
>      <!-- playback created files as soon as they were recorded by default
> -->
>      <!--<param name="auto-playback-recordings" value="true"/>-->
>      <email>
>        <param name="template-file" value="voicemail.tpl"/>
>        <param name="notify-template-file" value="notify-voicemail.tpl"/>
>        <!-- this is the format voicemail_time will have -->
>        <param name="date-fmt" value="%A, %B %d %Y, %I %M %p"/>
>        <param name="email-from" value="${voicemail_account}@
> ${voicemail_domain}"/>
>      </email>
>      <!--<param name="storage-dir" value="/tmp"/>-->
>      <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
>      <!--<param name="record-comment" value="Your Comment"/>-->
>      <!--<param name="record-title" value="Your Title"/>-->
>      <!--<param name="record-copyright" value="Your Copyright"/>-->
>    </profile>
>  </profiles>
> </configuration>
>
> In conf/directory/default.xml, user 91000 is defined in domain voicemail_2.
>  <domain name="voicemail_2">
>    <groups>
>      <group name="default">
>        <users>
>          <user bluebox_id="2" id="91000">
>            <params>
>              <param name="sip-forbid-register" value="true"/>
>              <param name="vm-password" value="xxxx"/>
>              <param name="vm-message-ext" value="wav"/>
>              <param name="vm-mailto" value="xxx at xxx.com"/>
>              <param name="email-addr" value="xxx at xxx.com"/>
>              <param name="vm-delete-file" value="false"/>
>              <param name="vm-attach-file" value="true"/>
>              <param name="vm-email-all-messages" value="true"/>
>            </params>
>            <variables>
>              <variable name="timezone" value="America/Chicago"/>
>            </variables>
>          </user>
>        </users>
>      </group>
>    </groups>
>  </domain>
>
> In my dialplan, calls to 1112223333 will be sent to user 91000's voicemail
> box if they are not answered.
>    <extension name="main_number_2" continue="true">
>      <condition field="destination_number" expression="^1112223333$">
>        <action application="set" bluebox="settingEndBridge"
> data="hangup_after_bridge=true"/>
>        <action application="set" bluebox="settingFail"
> data="continue_on_fail=true"/>
>        <action application="set" bluebox="autoPlay"
> data="vm_auto_play=false"/>
>        <action application="set" bluebox="settingTimeout"
> data="call_timeout=30"/>
>        <action application="set" bluebox="settingRing"
> data="ringback=${us-ring}"/>
>        <action application="set" bluebox="settingRingXfer"
> data="transfer_ringback=${us-ring}"/>
>        <action application="export" bluebox="sipCalleeIdName"
> data="sip_callee_id_name=m1"/>
>        <action application="export" bluebox="sipCalleeIdNumber"
> data="sip_callee_id_number=1112223333"/>
>        <action application="bridge" data="user/91000@$${location_3}"/>
>        <action application="answer"/>
>        <action application="voicemail" data="default voicemail_2 91000"/>
>        <action application="hangup"/>
>      </condition>
>    </extension>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110816/f5e43b6a/attachment.html 

From msc at freeswitch.org  Tue Aug 16 03:54:53 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 15 Aug 2011 16:54:53 -0700
Subject: [Freeswitch-users] ORIGINATE_DISPOSITION
In-Reply-To: <1313452154.28177.YahooMailNeo@web161001.mail.bf1.yahoo.com>
References: <1313447714.89555.YahooMailNeo@web161001.mail.bf1.yahoo.com>
	<CAKzWOxVwDaxsFJAr1dVP5nfmDuFXK2tptvai+oqtwZgph3u1kw@mail.gmail.com>
	<1313452154.28177.YahooMailNeo@web161001.mail.bf1.yahoo.com>
Message-ID: <CAKzWOxWYXy5Sa2g+cCDEDcxWwE10OtrMaJzUyoLdFSDGtDfbyg@mail.gmail.com>

And how is it being generated? WIth a bridge or originate or ... ?
-MC

On Mon, Aug 15, 2011 at 4:49 PM, Sam <lakersman2006 at yahoo.com> wrote:

> I want to know the B-leg status of the call.
>
> ------------------------------
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Monday, August 15, 2011 4:31 PM
> *Subject:* Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>
> I think you may be wanting "endpoint_disposition" depending on exactly what
> you're looking at.
>
> -MC
>
> On Mon, Aug 15, 2011 at 3:35 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> For the ORIGINATE_DISPOSITION channel variable, does "SUCCESS" mean the
> call was ANSWERED?
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110815/407f6b60/attachment-0001.html 

From lakersman2006 at yahoo.com  Tue Aug 16 03:57:02 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Mon, 15 Aug 2011 16:57:02 -0700 (PDT)
Subject: [Freeswitch-users] ORIGINATE_DISPOSITION
In-Reply-To: <CAKzWOxWYXy5Sa2g+cCDEDcxWwE10OtrMaJzUyoLdFSDGtDfbyg@mail.gmail.com>
References: <1313447714.89555.YahooMailNeo@web161001.mail.bf1.yahoo.com>
	<CAKzWOxVwDaxsFJAr1dVP5nfmDuFXK2tptvai+oqtwZgph3u1kw@mail.gmail.com>
	<1313452154.28177.YahooMailNeo@web161001.mail.bf1.yahoo.com>
	<CAKzWOxWYXy5Sa2g+cCDEDcxWwE10OtrMaJzUyoLdFSDGtDfbyg@mail.gmail.com>
Message-ID: <1313452622.11918.YahooMailNeo@web161012.mail.bf1.yahoo.com>

It is being generated with a bridge.



________________________________
From: Michael Collins <msc at freeswitch.org>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Monday, August 15, 2011 4:54 PM
Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION


And how is it being generated? WIth a bridge or originate or ... ?
-MC


On Mon, Aug 15, 2011 at 4:49 PM, Sam <lakersman2006 at yahoo.com> wrote:

I want to know the B-leg status of the call.
>
>
>
>
>________________________________
>From: Michael Collins <msc at freeswitch.org>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>Sent: Monday, August 15, 2011 4:31 PM
>Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>
>
>
>I think you may be wanting "endpoint_disposition" depending on exactly what you're looking at.
>
>
>-MC
>
>
>On Mon, Aug 15, 2011 at 3:35 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
>For the ORIGINATE_DISPOSITION channel variable, does "SUCCESS" mean the call was ANSWERED?
>>
>>
>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>>
>
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>
>
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>


FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110815/0f91a35e/attachment.html 

From msc at freeswitch.org  Tue Aug 16 03:58:37 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 15 Aug 2011 16:58:37 -0700
Subject: [Freeswitch-users] ORIGINATE_DISPOSITION
In-Reply-To: <1313452622.11918.YahooMailNeo@web161012.mail.bf1.yahoo.com>
References: <1313447714.89555.YahooMailNeo@web161001.mail.bf1.yahoo.com>
	<CAKzWOxVwDaxsFJAr1dVP5nfmDuFXK2tptvai+oqtwZgph3u1kw@mail.gmail.com>
	<1313452154.28177.YahooMailNeo@web161001.mail.bf1.yahoo.com>
	<CAKzWOxWYXy5Sa2g+cCDEDcxWwE10OtrMaJzUyoLdFSDGtDfbyg@mail.gmail.com>
	<1313452622.11918.YahooMailNeo@web161012.mail.bf1.yahoo.com>
Message-ID: <CAKzWOxVREFK=uzYzT95oXFT0-sURz7yeOwkuQMz6LKjQ6GvoXA@mail.gmail.com>

And how are you checking the variable? Do you have an event socket open or
... ?
-MC

On Mon, Aug 15, 2011 at 4:57 PM, Sam <lakersman2006 at yahoo.com> wrote:

> It is being generated with a bridge.
>
> ------------------------------
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Monday, August 15, 2011 4:54 PM
>
> *Subject:* Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>
> And how is it being generated? WIth a bridge or originate or ... ?
> -MC
>
> On Mon, Aug 15, 2011 at 4:49 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> I want to know the B-leg status of the call.
>
> ------------------------------
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Monday, August 15, 2011 4:31 PM
> *Subject:* Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>
> I think you may be wanting "endpoint_disposition" depending on exactly what
> you're looking at.
>
> -MC
>
> On Mon, Aug 15, 2011 at 3:35 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> For the ORIGINATE_DISPOSITION channel variable, does "SUCCESS" mean the
> call was ANSWERED?
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110815/56206f39/attachment.html 

From msc at freeswitch.org  Tue Aug 16 04:03:28 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 15 Aug 2011 17:03:28 -0700
Subject: [Freeswitch-users] getting disconnect cause for a leg after
 bridge in Lua
In-Reply-To: <20110813094624.167960@gmx.com>
References: <20110813094624.167960@gmx.com>
Message-ID: <CAKzWOxXh0czScXS1OadKyH98tiKj-ttRmmQaLdrnP9KCMjXPPg@mail.gmail.com>

No, you can't do this because the session you are checking is "gone" as soon
as the call leg is disconnected. You are better off using a hangup hook or
an event socket application if you need to get that value in realtime.
Dialplan scripts are good for connecting endpoints and doing simple logic
but they are absolutely not what you want for doing any kind of billing or
reporting.

-MC

On Sat, Aug 13, 2011 at 2:46 AM, Mi Ke <mi.ke at null.net> wrote:

> Hi All,
>
> Is there any way to get a real disconnection cause for leg B in the
> following script ?
>
>
> if (session_a:ready() and session_b:ready()) then
>
>    freeswitch.bridge(session_a,session_b)
>
>    -- session_b gets disconnect here ...
>
>     local session_b_hangup_cause = session_b:hangupCause()
>
>
>
> session_b_hangup_cause is always "SUCCESS" after debridging while log and
> CDR shows correct value - can get it to my script ?
>
> Thanks / Mike
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110815/9c77e458/attachment.html 

From lakersman2006 at yahoo.com  Tue Aug 16 04:05:26 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Mon, 15 Aug 2011 17:05:26 -0700 (PDT)
Subject: [Freeswitch-users] ORIGINATE_DISPOSITION
In-Reply-To: <CAKzWOxVREFK=uzYzT95oXFT0-sURz7yeOwkuQMz6LKjQ6GvoXA@mail.gmail.com>
References: <1313447714.89555.YahooMailNeo@web161001.mail.bf1.yahoo.com>
	<CAKzWOxVwDaxsFJAr1dVP5nfmDuFXK2tptvai+oqtwZgph3u1kw@mail.gmail.com>
	<1313452154.28177.YahooMailNeo@web161001.mail.bf1.yahoo.com>
	<CAKzWOxWYXy5Sa2g+cCDEDcxWwE10OtrMaJzUyoLdFSDGtDfbyg@mail.gmail.com>
	<1313452622.11918.YahooMailNeo@web161012.mail.bf1.yahoo.com>
	<CAKzWOxVREFK=uzYzT95oXFT0-sURz7yeOwkuQMz6LKjQ6GvoXA@mail.gmail.com>
Message-ID: <1313453126.47357.YahooMailNeo@web161010.mail.bf1.yahoo.com>

I am using perl's $session->get_variable("originate_disposition");

Also, how come the "hangup_time" shows zero on answered calls?



________________________________
From: Michael Collins <msc at freeswitch.org>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Monday, August 15, 2011 4:58 PM
Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION


And how are you checking the variable? Do you have an event socket open or ... ?
-MC


On Mon, Aug 15, 2011 at 4:57 PM, Sam <lakersman2006 at yahoo.com> wrote:

It is being generated with a bridge.
>
>
>
>
>________________________________
>From: Michael Collins <msc at freeswitch.org>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>Sent: Monday, August 15, 2011 4:54 PM
>
>Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>
>
>
>And how is it being generated? WIth a bridge or originate or ... ?
>-MC
>
>
>On Mon, Aug 15, 2011 at 4:49 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
>I want to know the B-leg status of the call.
>>
>>
>>
>>
>>________________________________
>>From: Michael Collins <msc at freeswitch.org>
>>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>Sent: Monday, August 15, 2011 4:31 PM
>>Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>
>>
>>
>>I think you may be wanting "endpoint_disposition" depending on exactly what you're looking at.
>>
>>
>>-MC
>>
>>
>>On Mon, Aug 15, 2011 at 3:35 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>
>>For the ORIGINATE_DISPOSITION channel variable, does "SUCCESS" mean the call was ANSWERED?
>>>
>>>
>>>FreeSWITCH-users mailing list
>>>FreeSWITCH-users at lists.freeswitch.org
>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>http://www.freeswitch.org
>>>
>>>
>>
>>
>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>>
>>
>>
>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>>
>
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>
>
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>


FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110815/af09718f/attachment-0001.html 

From msc at freeswitch.org  Tue Aug 16 04:30:55 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 15 Aug 2011 17:30:55 -0700
Subject: [Freeswitch-users] ORIGINATE_DISPOSITION
In-Reply-To: <1313453126.47357.YahooMailNeo@web161010.mail.bf1.yahoo.com>
References: <1313447714.89555.YahooMailNeo@web161001.mail.bf1.yahoo.com>
	<CAKzWOxVwDaxsFJAr1dVP5nfmDuFXK2tptvai+oqtwZgph3u1kw@mail.gmail.com>
	<1313452154.28177.YahooMailNeo@web161001.mail.bf1.yahoo.com>
	<CAKzWOxWYXy5Sa2g+cCDEDcxWwE10OtrMaJzUyoLdFSDGtDfbyg@mail.gmail.com>
	<1313452622.11918.YahooMailNeo@web161012.mail.bf1.yahoo.com>
	<CAKzWOxVREFK=uzYzT95oXFT0-sURz7yeOwkuQMz6LKjQ6GvoXA@mail.gmail.com>
	<1313453126.47357.YahooMailNeo@web161010.mail.bf1.yahoo.com>
Message-ID: <CAKzWOxWtxNoto1c6Chu5dZLPkt+ATcwCGpHOYtPpWydUrcCFfQ@mail.gmail.com>

On Mon, Aug 15, 2011 at 5:05 PM, Sam <lakersman2006 at yahoo.com> wrote:

> I am using perl's $session->get_variable("originate_disposition");
>
Are you looking at the b-leg's session?


>
> Also, how come the "hangup_time" shows zero on answered calls?
>
Because hangup_time refers to the point in time at which the call was hung
up. Since you are in the middle of a call (using the $session object) you
will never see the hangup_time because the object ceases to exist once the
call leg is disconnected.

I get the impression that you may be using the wrong tool for this
particular job, but I'm not sure without seeing it. If you don't mind
dropping it on pastebin we'll have a look and give you some suggestions.

-MC


> ------------------------------
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Monday, August 15, 2011 4:58 PM
>
> *Subject:* Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>
> And how are you checking the variable? Do you have an event socket open or
> ... ?
> -MC
>
> On Mon, Aug 15, 2011 at 4:57 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> It is being generated with a bridge.
>
> ------------------------------
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Monday, August 15, 2011 4:54 PM
>
> *Subject:* Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>
> And how is it being generated? WIth a bridge or originate or ... ?
> -MC
>
> On Mon, Aug 15, 2011 at 4:49 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> I want to know the B-leg status of the call.
>
> ------------------------------
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Monday, August 15, 2011 4:31 PM
> *Subject:* Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>
> I think you may be wanting "endpoint_disposition" depending on exactly what
> you're looking at.
>
> -MC
>
> On Mon, Aug 15, 2011 at 3:35 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> For the ORIGINATE_DISPOSITION channel variable, does "SUCCESS" mean the
> call was ANSWERED?
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110815/313dee42/attachment.html 

From kris at kriskinc.com  Tue Aug 16 06:05:39 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Mon, 15 Aug 2011 22:05:39 -0400
Subject: [Freeswitch-users] DTMF bleed through in conference
In-Reply-To: <CAKzWOxU_oaWg1_c=92W30g-pZyWkjWWW2YV5bh_hG-r1mZdVZA@mail.gmail.com>
References: <m2liuuz3y9.fsf@valheru.db3l.homeip.net>
	<CAKzWOxU_oaWg1_c=92W30g-pZyWkjWWW2YV5bh_hG-r1mZdVZA@mail.gmail.com>
Message-ID: <CAKDfjgf2bYTDPsFaoM3ubvu=NGoPxN-tSJR0aCo7NZyRcmAqFw@mail.gmail.com>

Agreed.  Sounds like an analog device not clamping inband DTMF -> RFC
2833 events properly (or quickly enough).

Grab a pcap and look at the audio.  See if you can see DTMF in the
audio stream itself.

On Mon, Aug 15, 2011 at 7:09 PM, Michael Collins <msc at freeswitch.org> wrote:
> Are the devices that send the DTMFs using 2833? Sounds like possibly they
> are using inband. I would get a pcap with RTP of the conference and see
> what, exactly, the device is sending. Some tips for this can be found here:
> http://wiki.freeswitch.org/wiki/Packet_Capture
> -MC
> On Mon, Aug 15, 2011 at 3:48 PM, David Bolen <db3l.net at gmail.com> wrote:
>>
>> I'm encountering what I would probably best describe as some bleed
>> through of inbound DTMF when in a conference, and I was wondering if
>> anyone might have suggestions for likely causes or troubleshooting?
>>
>> DTMF detection itself seems fine, and appears to properly be using
>> rfc2833 (the SDPs all show include "a=rtpmap:101 telephone-event/8000").
>> I get "RTP RECV DTMF" logs for each digit that is recognized.
>> However, a reasonable percentage of the time, some small fraction of
>> the DTMF appears to be heard by the conference as a whole. ?Never the
>> whole thing, and sometimes it's a bit distorted (though that may just
>> be due to it being so short). ?Almost as if a small portion of a sound
>> sample makes it through.
>>
>> For testing, I'm using internal phones with direct SIP connections to
>> the switch so no PSTN or gateways involved, though it also occurs with
>> a PSTN caller coming in through multiple gateways. ?For the internal
>> phones, I've tried both a physical phone through a PAP2T, as well as a
>> PC soft phone.
>>
>> This is with git-868d823 from Jul 29, so a few weeks old but reviewing
>> the more recent commits there weren't any seemingly related to DTMF or
>> conferencing.
>>
>> Any suggestions on places to look or things to try would be most
>> appreciated.
>>
>> -- David
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Kristian Kielhofner


From xing2kin at yahoo.com  Tue Aug 16 06:38:54 2011
From: xing2kin at yahoo.com (king2kin)
Date: Mon, 15 Aug 2011 19:38:54 -0700 (PDT)
Subject: [Freeswitch-users] Weird: Sofia-sip failed to be compiled on
	Visual Studio 2008 but it was OK three weeks ago
In-Reply-To: <1313436412335-6688832.post@n2.nabble.com>
Message-ID: <1313462334.33615.YahooMailClassic@web39705.mail.mud.yahoo.com>

Thank you for your advice. 

Yes, I did simply reply some exisiting email and change its subject as my own topic to take quickly this group-email address while I was sending a mail with my own topic.

Sorry, I didn't know this action still has impact on the early thread.

--- On Mon, 8/15/11, Jeff Lenk <jeff at jefflenk.com> wrote:

> From: Jeff Lenk <jeff at jefflenk.com>
> Subject: Re: [Freeswitch-users] Weird: Sofia-sip failed to be compiled on Visual Studio 2008 but it was OK three weeks ago
> To: freeswitch-users at lists.freeswitch.org
> Date: Monday, August 15, 2011, 12:26 PM
> king2kin,
> 
> Please do not hijack existing threads always send a new
> mail. Do not reply
> to an existing email and change the subject as this causes
> lots of confusion
> when trying to reply to known topics by readers of this
> list.
> 
> Only by accident did I even see your email. You must set
> autocrlf=false
> before you checkout from git.
> 
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Mod-event-socket-error-tp6688617p6688832.html
> Sent from the freeswitch-users mailing list archive at
> Nabble.com.
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


From kbdfck at gmail.com  Tue Aug 16 13:30:43 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Tue, 16 Aug 2011 13:30:43 +0400
Subject: [Freeswitch-users] ORIGINATE_DISPOSITION
In-Reply-To: <CAKzWOxWtxNoto1c6Chu5dZLPkt+ATcwCGpHOYtPpWydUrcCFfQ@mail.gmail.com>
References: <1313447714.89555.YahooMailNeo@web161001.mail.bf1.yahoo.com>
	<CAKzWOxVwDaxsFJAr1dVP5nfmDuFXK2tptvai+oqtwZgph3u1kw@mail.gmail.com>
	<1313452154.28177.YahooMailNeo@web161001.mail.bf1.yahoo.com>
	<CAKzWOxWYXy5Sa2g+cCDEDcxWwE10OtrMaJzUyoLdFSDGtDfbyg@mail.gmail.com>
	<1313452622.11918.YahooMailNeo@web161012.mail.bf1.yahoo.com>
	<CAKzWOxVREFK=uzYzT95oXFT0-sURz7yeOwkuQMz6LKjQ6GvoXA@mail.gmail.com>
	<1313453126.47357.YahooMailNeo@web161010.mail.bf1.yahoo.com>
	<CAKzWOxWtxNoto1c6Chu5dZLPkt+ATcwCGpHOYtPpWydUrcCFfQ@mail.gmail.com>
Message-ID: <CA+GmxbZ2RG+RS79F8yjXAR4eeFwFy3EMV0k4YiESwq3qMCbwkA@mail.gmail.com>

Seems we need to clear things about B-leg disposition in wiki.
AFAIK there is no method to get correct disposition from B leg without
analyzing events in case when A-leg was answered by FS itself.

If A leg was not answered, we can use A-leg disposition for call
disposition.

2011/8/16 Michael Collins <msc at freeswitch.org>

>
>
> On Mon, Aug 15, 2011 at 5:05 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
>> I am using perl's $session->get_variable("originate_disposition");
>>
> Are you looking at the b-leg's session?
>
>
>>
>> Also, how come the "hangup_time" shows zero on answered calls?
>>
> Because hangup_time refers to the point in time at which the call was hung
> up. Since you are in the middle of a call (using the $session object) you
> will never see the hangup_time because the object ceases to exist once the
> call leg is disconnected.
>
> I get the impression that you may be using the wrong tool for this
> particular job, but I'm not sure without seeing it. If you don't mind
> dropping it on pastebin we'll have a look and give you some suggestions.
>
> -MC
>
>
>> ------------------------------
>> *From:* Michael Collins <msc at freeswitch.org>
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Sent:* Monday, August 15, 2011 4:58 PM
>>
>> *Subject:* Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>
>> And how are you checking the variable? Do you have an event socket open or
>> ... ?
>> -MC
>>
>> On Mon, Aug 15, 2011 at 4:57 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>
>> It is being generated with a bridge.
>>
>> ------------------------------
>> *From:* Michael Collins <msc at freeswitch.org>
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Sent:* Monday, August 15, 2011 4:54 PM
>>
>> *Subject:* Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>
>> And how is it being generated? WIth a bridge or originate or ... ?
>> -MC
>>
>> On Mon, Aug 15, 2011 at 4:49 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>
>> I want to know the B-leg status of the call.
>>
>> ------------------------------
>> *From:* Michael Collins <msc at freeswitch.org>
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Sent:* Monday, August 15, 2011 4:31 PM
>> *Subject:* Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>
>> I think you may be wanting "endpoint_disposition" depending on exactly
>> what you're looking at.
>>
>> -MC
>>
>> On Mon, Aug 15, 2011 at 3:35 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>
>> For the ORIGINATE_DISPOSITION channel variable, does "SUCCESS" mean the
>> call was ANSWERED?
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Best regards,

Dmitry Sytchev,
IT Engineer
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110816/be15e94a/attachment-0001.html 

From dhairya.blogs at gmail.com  Tue Aug 16 13:32:26 2011
From: dhairya.blogs at gmail.com (Dhairya Vora)
Date: Tue, 16 Aug 2011 15:02:26 +0530
Subject: [Freeswitch-users] [Newbie] Unable to play .mp3 file
Message-ID: <CA+P_fE-V0vVmcRnnXHBvdbYFv=ULzjDF4D4y-j29WTcTunCyiA@mail.gmail.com>

I have just tried a simple mp3 file, but freeswitch gives an error when call
is established...

freeswitch at internal> 2011-08-16 08:58:51.464482 [NOTICE] sofia.c:5725
Channel [sofia/external/00919879459879] has been answered
2011-08-16 08:58:51.464482 [NOTICE] switch_ivr.c:1663 Transfer
sofia/external/00919879459879 to inline[socket:127.0.0.1:8084 async
full at default]
2011-08-16 08:58:51.464482 [INFO] switch_channel.c:2638
sofia/external/00919879459879 Flipping CID from "" <00919879459879> to
"Outbound Call" <gateway/flowroute/00919879459879>
2011-08-16 08:58:52.044467 [ERR] mod_shout.c:806 Error: MPG123 Error at
/usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c:627.
2011-08-16 08:58:52.044467 [ERR] mod_shout.c:809 Error from mpg123: Invalid
mpg123 handle. (code 10)
2011-08-16 08:58:52.304451 [ERR] mod_shout.c:806 Error: MPG123 Error at
/usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c:627.
2011-08-16 08:58:52.304451 [ERR] mod_shout.c:809 Error from mpg123: Invalid
mpg123 handle. (code 10)
2011-08-16 08:58:52.564453 [ERR] mod_shout.c:806 Error: MPG123 Error at
/usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c:627.
2011-08-16 08:58:52.564453 [ERR] mod_shout.c:809 Error from mpg123: Invalid
mpg123 handle. (code 10)
2011-08-16 08:58:52.824447 [ERR] mod_shout.c:806 Error: MPG123 Error at
/usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c:627.
2011-08-16 08:58:52.824447 [ERR] mod_shout.c:809 Error from mpg123: Invalid
mpg123 handle. (code 10)
2011-08-16 08:58:52.844500 [NOTICE] mod_dptools.c:916 Hangup
sofia/external/00919879459879 [CS_RESET] [NORMAL_CLEARING]
2011-08-16 08:58:52.884443 [NOTICE] switch_core_session.c:1347 Session 3
(sofia/external/00919879459879) Ended
2011-08-16 08:58:52.884443 [NOTICE] switch_core_session.c:1349 Close Channel
sofia/external/00919879459879 [CS_DESTROY]

I checked that mod_shout is enable. The .mp3 file is not currupted. I am
able to play in mp3 player.

Please help solving this error:

-Dhairya


P.S. I am using plivo php file to make this call
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110816/b90aad35/attachment.html 

From gmaruzz at gmail.com  Tue Aug 16 13:42:14 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 16 Aug 2011 11:42:14 +0200
Subject: [Freeswitch-users] [Newbie] Unable to play .mp3 file
In-Reply-To: <CA+P_fE-V0vVmcRnnXHBvdbYFv=ULzjDF4D4y-j29WTcTunCyiA@mail.gmail.com>
References: <CA+P_fE-V0vVmcRnnXHBvdbYFv=ULzjDF4D4y-j29WTcTunCyiA@mail.gmail.com>
Message-ID: <CALXCt0qgOx68CvMfWF+rSzfe0aFPNENLRZ42-sKbPsw5Qy79-Q@mail.gmail.com>

have you installed mpg123 (and is it where plivo is expecting it to be)?

-giovanni

On Tue, Aug 16, 2011 at 11:32 AM, Dhairya Vora <dhairya.blogs at gmail.com> wrote:
> I have just tried a simple mp3 file, but freeswitch gives an error when call
> is established...
>
> freeswitch at internal> 2011-08-16 08:58:51.464482 [NOTICE] sofia.c:5725
> Channel [sofia/external/00919879459879] has been answered
> 2011-08-16 08:58:51.464482 [NOTICE] switch_ivr.c:1663 Transfer
> sofia/external/00919879459879 to inline[socket:127.0.0.1:8084 async
> full at default]
> 2011-08-16 08:58:51.464482 [INFO] switch_channel.c:2638
> sofia/external/00919879459879 Flipping CID from "" <00919879459879> to
> "Outbound Call" <gateway/flowroute/00919879459879>
> 2011-08-16 08:58:52.044467 [ERR] mod_shout.c:806 Error: MPG123 Error at
> /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c:627.
> 2011-08-16 08:58:52.044467 [ERR] mod_shout.c:809 Error from mpg123: Invalid
> mpg123 handle. (code 10)
> 2011-08-16 08:58:52.304451 [ERR] mod_shout.c:806 Error: MPG123 Error at
> /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c:627.
> 2011-08-16 08:58:52.304451 [ERR] mod_shout.c:809 Error from mpg123: Invalid
> mpg123 handle. (code 10)
> 2011-08-16 08:58:52.564453 [ERR] mod_shout.c:806 Error: MPG123 Error at
> /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c:627.
> 2011-08-16 08:58:52.564453 [ERR] mod_shout.c:809 Error from mpg123: Invalid
> mpg123 handle. (code 10)
> 2011-08-16 08:58:52.824447 [ERR] mod_shout.c:806 Error: MPG123 Error at
> /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c:627.
> 2011-08-16 08:58:52.824447 [ERR] mod_shout.c:809 Error from mpg123: Invalid
> mpg123 handle. (code 10)
> 2011-08-16 08:58:52.844500 [NOTICE] mod_dptools.c:916 Hangup
> sofia/external/00919879459879 [CS_RESET] [NORMAL_CLEARING]
> 2011-08-16 08:58:52.884443 [NOTICE] switch_core_session.c:1347 Session 3
> (sofia/external/00919879459879) Ended
> 2011-08-16 08:58:52.884443 [NOTICE] switch_core_session.c:1349 Close Channel
> sofia/external/00919879459879 [CS_DESTROY]
>
> I checked that mod_shout is enable. The .mp3 file is not currupted. I am
> able to play in mp3 player.
>
> Please help solving this error:
>
> -Dhairya
>
>
> P.S. I am using plivo php file to make this call
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 gmaruzz at gmail.com  Tue Aug 16 13:53:23 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Tue, 16 Aug 2011 11:53:23 +0200
Subject: [Freeswitch-users] [Freeswitch-dev] Meetup Paris?
In-Reply-To: <CAPHrvXHx-3u0hvQ2WzRZ7M1K5OTO6J-Bur=hC1z5wxXN43p2rA@mail.gmail.com>
References: <E58D1B62-B7E9-4D15-A0AF-DC993AA70786@avgs.ca>
	<CAPHrvXHx-3u0hvQ2WzRZ7M1K5OTO6J-Bur=hC1z5wxXN43p2rA@mail.gmail.com>
Message-ID: <CALXCt0pzErc5beBzNXJ_niDVm+4X5W1CTkV8Ts28vK=hoDD5ww@mail.gmail.com>

I have fond memories of a conference I gave couple of years ago in
Hyderabad and the nice reception has had.

Certainly there are lot of people and companies that can be interested
in FreeSWITCH, and Cluecon India.

Most people is still on Asterisk, but I believe they'll be very
interested into knowing first hand what are the advantages of using
FS, maybe in powerful combination with OpenSips.

Let's have the ball rolling, try to gauge what interest you can
gather, and which companies can act as sponsors (money matters ;) ).

-giovanni


On Tue, Aug 16, 2011 at 8:28 AM, Prashant Lamba
<prashant.lamba at gmail.com> wrote:
> Colleagues,
> After seeing the success of ClueCon, I have always been keen to promote open
> source telephony here in India and possibly organize an "India" leg of
> something like ClueCon here in Hyderabad. I can start the ground work of
> gathering people, speakers, venue etc.
> Hyderabad is probably the best technology hub here in India. Any
> suggestions?
> Prashant
> --
> Prashant Lamba
> Phonologies (India) Private Limited
> e: prashant at phonologies.com
> m: +91 9867.22.1975
>
> On Fri, Aug 12, 2011 at 10:58 AM, Mathieu Rene <mrene_lists at avgs.ca> wrote:
>>
>> English version follows.
>>
>> En cette fin de ClueCon, je pars sur Paris demain soir (donc j'arrive
>> samedi en matin?, bien d?cal?), et j'y resterai jusqu'? mercredi. ?Je
>> propose d'organiser un petit meetup afin de joindre l'utile ? l'agr?able et
>> de voir un peu le visage de la communaut? freeswitch parisienne. Est-ce
>> quelqu'un peut proposer un bar sympa ou on pourrait faire un truc?
>>
>> ---
>>
>> ClueCon is over and I'm leaving to Paris tomorrow (landing saturday
>> morning, well jetlagged) and staying until Wednesday. I'm proposing to
>> organize a little meetup in order to get to know the face of the parisian
>> freeswitch community. Does anyone know a bar where we could meet up?
>>
>> Mathieu Rene
>> Avant-Garde Solutions Inc
>> Office: + 1 (514) 664-1044 x100
>> Cell: +1 (514) 664-1044 x200
>> mrene at avgs.ca
>>
>>
>>
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From dhairya.blogs at gmail.com  Tue Aug 16 14:51:27 2011
From: dhairya.blogs at gmail.com (Dhairya Vora)
Date: Tue, 16 Aug 2011 16:21:27 +0530
Subject: [Freeswitch-users] [Newbie] Unable to play .mp3 file
In-Reply-To: <CALXCt0qgOx68CvMfWF+rSzfe0aFPNENLRZ42-sKbPsw5Qy79-Q@mail.gmail.com>
References: <CA+P_fE-V0vVmcRnnXHBvdbYFv=ULzjDF4D4y-j29WTcTunCyiA@mail.gmail.com>
	<CALXCt0qgOx68CvMfWF+rSzfe0aFPNENLRZ42-sKbPsw5Qy79-Q@mail.gmail.com>
Message-ID: <CA+P_fE8X5rsQopzxgLyjZZOJBY7EFgxjutz436xZsAy39gWybg@mail.gmail.com>

I have installed mpg123 at /usr/src/mpg123-1.12.4

I am searching where plivo is expecting it to be.

On Tue, Aug 16, 2011 at 3:12 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

> have you installed mpg123 (and is it where plivo is expecting it to be)?
>
> -giovanni
>
> On Tue, Aug 16, 2011 at 11:32 AM, Dhairya Vora <dhairya.blogs at gmail.com>
> wrote:
> > I have just tried a simple mp3 file, but freeswitch gives an error when
> call
> > is established...
> >
> > freeswitch at internal> 2011-08-16 08:58:51.464482 [NOTICE] sofia.c:5725
> > Channel [sofia/external/00919879459879] has been answered
> > 2011-08-16 08:58:51.464482 [NOTICE] switch_ivr.c:1663 Transfer
> > sofia/external/00919879459879 to inline[socket:127.0.0.1:8084 async
> > full at default]
> > 2011-08-16 08:58:51.464482 [INFO] switch_channel.c:2638
> > sofia/external/00919879459879 Flipping CID from "" <00919879459879> to
> > "Outbound Call" <gateway/flowroute/00919879459879>
> > 2011-08-16 08:58:52.044467 [ERR] mod_shout.c:806 Error: MPG123 Error at
> > /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c:627.
> > 2011-08-16 08:58:52.044467 [ERR] mod_shout.c:809 Error from mpg123:
> Invalid
> > mpg123 handle. (code 10)
> > 2011-08-16 08:58:52.304451 [ERR] mod_shout.c:806 Error: MPG123 Error at
> > /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c:627.
> > 2011-08-16 08:58:52.304451 [ERR] mod_shout.c:809 Error from mpg123:
> Invalid
> > mpg123 handle. (code 10)
> > 2011-08-16 08:58:52.564453 [ERR] mod_shout.c:806 Error: MPG123 Error at
> > /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c:627.
> > 2011-08-16 08:58:52.564453 [ERR] mod_shout.c:809 Error from mpg123:
> Invalid
> > mpg123 handle. (code 10)
> > 2011-08-16 08:58:52.824447 [ERR] mod_shout.c:806 Error: MPG123 Error at
> > /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c:627.
> > 2011-08-16 08:58:52.824447 [ERR] mod_shout.c:809 Error from mpg123:
> Invalid
> > mpg123 handle. (code 10)
> > 2011-08-16 08:58:52.844500 [NOTICE] mod_dptools.c:916 Hangup
> > sofia/external/00919879459879 [CS_RESET] [NORMAL_CLEARING]
> > 2011-08-16 08:58:52.884443 [NOTICE] switch_core_session.c:1347 Session 3
> > (sofia/external/00919879459879) Ended
> > 2011-08-16 08:58:52.884443 [NOTICE] switch_core_session.c:1349 Close
> Channel
> > sofia/external/00919879459879 [CS_DESTROY]
> >
> > I checked that mod_shout is enable. The .mp3 file is not currupted. I am
> > able to play in mp3 player.
> >
> > Please help solving this error:
> >
> > -Dhairya
> >
> >
> > P.S. I am using plivo php file to make this call
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110816/ac02a834/attachment.html 

From dhairya.blogs at gmail.com  Tue Aug 16 14:53:47 2011
From: dhairya.blogs at gmail.com (Dhairya Vora)
Date: Tue, 16 Aug 2011 16:23:47 +0530
Subject: [Freeswitch-users] [Newbie] Unable to play .mp3 file
In-Reply-To: <CA+P_fE8X5rsQopzxgLyjZZOJBY7EFgxjutz436xZsAy39gWybg@mail.gmail.com>
References: <CA+P_fE-V0vVmcRnnXHBvdbYFv=ULzjDF4D4y-j29WTcTunCyiA@mail.gmail.com>
	<CALXCt0qgOx68CvMfWF+rSzfe0aFPNENLRZ42-sKbPsw5Qy79-Q@mail.gmail.com>
	<CA+P_fE8X5rsQopzxgLyjZZOJBY7EFgxjutz436xZsAy39gWybg@mail.gmail.com>
Message-ID: <CA+P_fE8cxhNNC3gSNp06Xur5Mi3nnvG+eD_06F3mAUY-2u6Byw@mail.gmail.com>

There is also a setup at /usr/src/freeswitch/libs/mpg123-1.13.2/


On Tue, Aug 16, 2011 at 4:21 PM, Dhairya Vora <dhairya.blogs at gmail.com>wrote:

> I have installed mpg123 at /usr/src/mpg123-1.12.4
>
> I am searching where plivo is expecting it to be.
>
>
> On Tue, Aug 16, 2011 at 3:12 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:
>
>> have you installed mpg123 (and is it where plivo is expecting it to be)?
>>
>> -giovanni
>>
>> On Tue, Aug 16, 2011 at 11:32 AM, Dhairya Vora <dhairya.blogs at gmail.com>
>> wrote:
>> > I have just tried a simple mp3 file, but freeswitch gives an error when
>> call
>> > is established...
>> >
>> > freeswitch at internal> 2011-08-16 08:58:51.464482 [NOTICE] sofia.c:5725
>> > Channel [sofia/external/00919879459879] has been answered
>> > 2011-08-16 08:58:51.464482 [NOTICE] switch_ivr.c:1663 Transfer
>> > sofia/external/00919879459879 to inline[socket:127.0.0.1:8084 async
>> > full at default]
>> > 2011-08-16 08:58:51.464482 [INFO] switch_channel.c:2638
>> > sofia/external/00919879459879 Flipping CID from "" <00919879459879> to
>> > "Outbound Call" <gateway/flowroute/00919879459879>
>> > 2011-08-16 08:58:52.044467 [ERR] mod_shout.c:806 Error: MPG123 Error at
>> > /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c:627.
>> > 2011-08-16 08:58:52.044467 [ERR] mod_shout.c:809 Error from mpg123:
>> Invalid
>> > mpg123 handle. (code 10)
>> > 2011-08-16 08:58:52.304451 [ERR] mod_shout.c:806 Error: MPG123 Error at
>> > /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c:627.
>> > 2011-08-16 08:58:52.304451 [ERR] mod_shout.c:809 Error from mpg123:
>> Invalid
>> > mpg123 handle. (code 10)
>> > 2011-08-16 08:58:52.564453 [ERR] mod_shout.c:806 Error: MPG123 Error at
>> > /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c:627.
>> > 2011-08-16 08:58:52.564453 [ERR] mod_shout.c:809 Error from mpg123:
>> Invalid
>> > mpg123 handle. (code 10)
>> > 2011-08-16 08:58:52.824447 [ERR] mod_shout.c:806 Error: MPG123 Error at
>> > /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c:627.
>> > 2011-08-16 08:58:52.824447 [ERR] mod_shout.c:809 Error from mpg123:
>> Invalid
>> > mpg123 handle. (code 10)
>> > 2011-08-16 08:58:52.844500 [NOTICE] mod_dptools.c:916 Hangup
>> > sofia/external/00919879459879 [CS_RESET] [NORMAL_CLEARING]
>> > 2011-08-16 08:58:52.884443 [NOTICE] switch_core_session.c:1347 Session 3
>> > (sofia/external/00919879459879) Ended
>> > 2011-08-16 08:58:52.884443 [NOTICE] switch_core_session.c:1349 Close
>> Channel
>> > sofia/external/00919879459879 [CS_DESTROY]
>> >
>> > I checked that mod_shout is enable. The .mp3 file is not currupted. I am
>> > able to play in mp3 player.
>> >
>> > Please help solving this error:
>> >
>> > -Dhairya
>> >
>> >
>> > P.S. I am using plivo php file to make this call
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at 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
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110816/2ec4578f/attachment-0001.html 

From asilva at wirelessmundi.com  Tue Aug 16 15:15:47 2011
From: asilva at wirelessmundi.com (Antonio)
Date: Tue, 16 Aug 2011 13:15:47 +0200
Subject: [Freeswitch-users] Problem receiving fax
Message-ID: <1313493347.30552.80.camel@marces.madrid.commsmundi.com>

Hi to all,

I'm having problems receiving fax in a pri E1 line. 

When a restart freeswitch or the server and i can receive one fax and
then it starts to fail again.

I didn't open a jira bug, because I'm not sure where to look for the
problem.

The log can be found at http://pastebin.freeswitch.org/17047


Thanks,

Ant?nio Silva





From benkokakao at gmail.com  Tue Aug 16 15:46:47 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Tue, 16 Aug 2011 13:46:47 +0200
Subject: [Freeswitch-users] Problem receiving fax
In-Reply-To: <1313493347.30552.80.camel@marces.madrid.commsmundi.com>
References: <1313493347.30552.80.camel@marces.madrid.commsmundi.com>
Message-ID: <CAAMQ8bROOOci2ocWZkcn_qxh-vc8kY1Tg+3deUJMs3Lvf0PMpw@mail.gmail.com>

On 16 August 2011 13:15, Antonio <asilva at wirelessmundi.com> wrote:
> I'm having problems receiving fax in a pri E1 line.
> The log can be found at http://pastebin.freeswitch.org/17047

Hi!

I had the same issue a few days ago("FLOW T.30 Bad HDLC CRC received").
Recompiling&Reinstalling libpri&FreeSWITCH helped.

hthu2
Christian


From jcgpoza at gmail.com  Tue Aug 16 16:08:26 2011
From: jcgpoza at gmail.com (Dissident)
Date: Tue, 16 Aug 2011 05:08:26 -0700 (PDT)
Subject: [Freeswitch-users] Goip GSM Gateway works great with FreeSwitch!
Message-ID: <1313496506598-6691087.post@n2.nabble.com>

http://freeswitch-users.2379917.n2.nabble.com/file/n6691087/single-channel-gsm-gateway-goip-gateway.jpg 

Hi, I've configured this GSM gateway with Freeswitch and it works great. I'm
posting this so that if somebody tries to configure this Gateway won't have
to spend as much time as a I did :) 

This is the configuration:

*Gateway*
*freeswitch/conf/sip_profiles/external/goip.xml*

*Tips to check that everything is OK
 - Remember to remove the PIN number from your SIM card. Double check your
SIM card with a Phone. It should not ask you for a  PIN. Disable it! Period
:) 
 - *sofia status profile internal* (the Goip extension user must appear, I
chose ext. 1000 for it...)
- * sofia status* (to check that the Goip gateway is properly configured.
 - On the Goip status page the Phone Status and your GSM Status must be
LOGIN 
 - On the Goip status page GSM operator GSM signal must show something. If
it doesn't it's because it has not logged properly with your GSM Sim card.
Double check it with a Phone.
 - If the Goip is logged properly with FS you can use your favorite SIP
client to log directly into GOIP that is to say. if your GOIP IP is
192.168.2.2 you can log into 1000 at 192.168.2.2 *(again this is not your FS IP
is the GOIP IP)* and make an outbound call to an ordinary cell phone. This
is useful in case you have trouble with the outbound call.

* Things to find out :??
- First I used X-Lite, Windows XP and miniSip server 20 clients to test this
gateway. It's kind of strange that with miniSip every time I call to the
extension I chose for Goip a tone appears an then I can dial a cell phone
number. FS says that this extension is not available, it shows up with the
sofia status profile internal though.

*I've used with Linux Ubuntu:
 - FreeSWITCH Version 1.0.head (git-decfdbb 2011-08-11 14-15-26 -0500)
 - Jitsy Java Sip Client 
 - 3 Android Phones with CSipSimple, Sipdroid and 3CXPhone Sip Client
 - 2 linksys spa941
 - 1 Goip GSM GateWay

http://goip.com.ua/download/goip-series-manual.pdf
http://goip.com.ua/download/goip-series-manual.pdf 
http://www.alibaba.com/trade/search?SearchText=goip+gsm+gateway&Country=&IndexArea=product_en&fsb=y&CatId=509
http://www.alibaba.com/trade/search?SearchText=goip+gsm+gateway&Country=&IndexArea=product_en&fsb=y&CatId=509 


I hope you find this info useful, if you have any comments or requests or
corrections... please let me know...

best regards.














--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Goip-GSM-Gateway-works-great-with-FreeSwitch-tp6691087p6691087.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From asilva at wirelessmundi.com  Tue Aug 16 16:14:33 2011
From: asilva at wirelessmundi.com (Antonio)
Date: Tue, 16 Aug 2011 14:14:33 +0200
Subject: [Freeswitch-users] Problem receiving fax
In-Reply-To: <CAAMQ8bROOOci2ocWZkcn_qxh-vc8kY1Tg+3deUJMs3Lvf0PMpw@mail.gmail.com>
References: <1313493347.30552.80.camel@marces.madrid.commsmundi.com>
	<CAAMQ8bROOOci2ocWZkcn_qxh-vc8kY1Tg+3deUJMs3Lvf0PMpw@mail.gmail.com>
Message-ID: <1313496873.30552.82.camel@marces.madrid.commsmundi.com>


I'm using libpri-1.4.11 and freeswitch head. I'm going to try with the
latest libpri-1.4.12.

And post the results.

Thanks,
Ant?nio


On Tue, 2011-08-16 at 13:46 +0200, Christian Benke wrote: 
> On 16 August 2011 13:15, Antonio <asilva at wirelessmundi.com> wrote:
> > I'm having problems receiving fax in a pri E1 line.
> > The log can be found at http://pastebin.freeswitch.org/17047
> 
> Hi!
> 
> I had the same issue a few days ago("FLOW T.30 Bad HDLC CRC received").
> Recompiling&Reinstalling libpri&FreeSWITCH helped.
> 
> hthu2
> Christian
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 

Un cordial saludo / Best regards, 

 _________________________

Ant?nio Silva

E-mail:asilva at wirelessmundi.com



From avi at avimarcus.net  Tue Aug 16 16:44:11 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 16 Aug 2011 15:44:11 +0300
Subject: [Freeswitch-users] Inband DTMF when rfc2833 negotiated?
Message-ID: <CALNZuwkD8q5L7ErTw7rLTTRD=ZB4QNmNF_nbyLuhL5cAwPdFmw@mail.gmail.com>

Why is this call using inband when the SDP says rfc2833? I got complaints
that the DTMF wasn't working, despite the dtmf numbers in the log being
correct.

Trace on leg B: http://pastebin.freeswitch.org/17053


Thanks,
-Avi

p.s. I think I've seen this with toll free gateway, too.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110816/274d9ed3/attachment.html 

From tculjaga at gmail.com  Tue Aug 16 16:44:16 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 16 Aug 2011 14:44:16 +0200
Subject: [Freeswitch-users] FS performance using ESL
In-Reply-To: <CAKbxfG_SnLS2c6CnepKcy=ea7CY=JnM5Lyy_Rv8ahUYpqJdooA@mail.gmail.com>
References: <CAJoKKdU9VPzOS-C3jHCjG6Y0V-Jo_j7hqYwt6-Zh7ohb1pq3xQ@mail.gmail.com>
	<CAKbxfG-SWBzu0dBv_5B0v7710q7vLeeduT9L5V_cnceYHjrGJQ@mail.gmail.com>
	<CAJoKKdUcA=9CvSdRPhaoMopK+f4K2W1DO0=0b5mck_KEjOc9pQ@mail.gmail.com>
	<CAJoKKdXR64qNfCYj0uyYFda4YLavwd_NFn3bH6Td+2eSGo7Y1w@mail.gmail.com>
	<CAKbxfG_SnLS2c6CnepKcy=ea7CY=JnM5Lyy_Rv8ahUYpqJdooA@mail.gmail.com>
Message-ID: <CAJoKKdX_sPBeWGv2xq7NOX4gd2Q1QQvWpm0wiSLv+67PvaGFVQ@mail.gmail.com>

On Mon, Aug 15, 2011 at 6:30 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> You must have something setup strangely cos it would definitely reduce
> your overall cps to use ESL but not down to 2 CPS.
>
> Did you look over the server stats like top etc and look for any
> misconfiguration?
>
> Hello Anthony, thanks for your response....

yay, i found the cause .... testserver and FS were running on the same
server. The server had just 1GB of RAM and of course ... by forking
testserver (on 8 CPS) took all the remaining RAM ending to write into swap
... this triggered a domino effect on the entire server becoming less and
less responsive as testserver started to run from swap!!!... yay .. really
bad... didn't see it happen until i started nmon ... top/htop didn't make it
in time to show this issue..

anyhow, i moved testserver to another machine reaching 35 CPS ... really
nice indeed.


now, having a referent point (testserver) im trying to reach that 35 CPS
with a java application.

yes, i notices few issues :=)

please check: http://pastebin.freeswitch.org/17052


------------------------------------snipp-----------------------------------

Control: full


// here i subscribe to all events ... well not good idea but its a start
events plain all

Content-Type: command/reply
Reply-Text: +OK event listener enabled plain

//and here i do a filter per uuid
filter Unique-ID f7a7b97b-df96-41f3-a6a3-fdf24350a45c

Content-Type: command/reply
Reply-Text: +OK filter added.
[Unique-ID]=[f7a7b97b-df96-41f3-a6a3-fdf24350a45c]

linger

Content-Type: command/reply
Reply-Text: +OK will linger


// here i send answer in sync mode ( i could change it into async)
sendmsg
call-command: execute
execute-app-name: answer
event-lock: true

Content-Type: command/reply
Reply-Text: +OK

Content-Length: 1805
Content-Type: text/event-plain


--------------------------------------------------------------------------------




so my questions:

if i use <action application="socket" data="192.168.254.99:8084 async
full"/> and if i subscribe to "myevents" i don't need to set a filter on
uuid and i could gain performance.
if i use <action application="socket" data="192.168.254.99:8084 async"/>  i
will be getting events for the call in question only... so no special
filters needed and i could limit the number of events im subscribing


what is a better approach in a matter of performance ?
What do i loose/gain by using async full vs async mode ?



Thanks for your answer,
Tihomir.









>
> On Thu, Aug 11, 2011 at 6:56 PM, Tihomir Culjaga <tculjaga at gmail.com>
> wrote:
> > is there any other method than esl to controll calls on FS from an
> eternal
> > application?
> > will mod_curl or mod_xml_curl get better performance?
> >
> > T.
> >
> > On Fri, Aug 12, 2011 at 1:33 AM, Tihomir Culjaga <tculjaga at gmail.com>
> wrote:
> >>
> >> Hi Anthony, thanks for your response ...
> >>
> >>
> >> this is what i have:
> >>
> >>         esl_filter(&handle, "unique-id",
> >> esl_event_get_header(handle.info_event, "caller-unique-id"));
> >>         esl_events(&handle, ESL_EVENT_TYPE_PLAIN, "CHANNEL_DATA
> >> CHANNEL_EXECUTE_COMPLETE CHANNEL_HANGUP");
> >>
> >> what do you suggest i put there ?
> >>
> >>
> >> is the inbound method less costly ?
> >>
> >>
> >>
> >>
> >> I modified testserver.c just a bit...
> >>
> >> #include <sys/types.h>  /* include this before any other sys headers */
> >> #include <sys/wait.h>   /* header for waitpid() and various macros */
> >> #include <signal.h>     /* header for signal functions */
> >> #include <stdio.h>      /* header for fprintf() */
> >> #include <unistd.h>     /* header for fork() */
> >> #include <stdlib.h>
> >> #include <esl.h>
> >>
> >> void sig_chld(int);     /* prototype for our SIGCHLD handler */
> >>
> >> static void mycallback(esl_socket_t server_sock, esl_socket_t
> client_sock,
> >> struct sockaddr_in *addr)
> >> {
> >>         esl_handle_t handle = {{0}};
> >>         int done = 0;
> >>         esl_status_t status;
> >>         time_t exp = 0;
> >>
> >>         if (fork() != 0) {
> >>                 close(client_sock);
> >>                 return;
> >>         }
> >>
> >>         esl_attach_handle(&handle, client_sock, addr);
> >>
> >>         esl_log(ESL_LOG_INFO, "Connected! %d\n", handle.sock);
> >>
> >>         esl_filter(&handle, "unique-id",
> >> esl_event_get_header(handle.info_event, "caller-unique-id"));
> >>         esl_events(&handle, ESL_EVENT_TYPE_PLAIN, "CHANNEL_DATA
> >> CHANNEL_EXECUTE_COMPLETE CHANNEL_HANGUP");
> >>
> >>         esl_send_recv(&handle, "linger");
> >>
> >>         esl_execute(&handle, "answer", NULL, NULL);
> >>         //esl_execute(&handle, "conference", "3000 at default", NULL);
> >>         esl_execute(&handle, "playback", "/home/tculjaga/myWavFile.wav",
> >> NULL);
> >>         //esl_execute(&handle, "sleep", "1000", NULL);
> >>         //esl_execute(&handle, "hangup", NULL, NULL);
> >>
> >>         while((status = esl_recv_timed(&handle, 1000)) != ESL_FAIL) {
> >>                 if (done) {
> >>                         if (time(NULL) >= exp) {
> >>                                 break;
> >>                         }
> >>                 } else if (status == ESL_SUCCESS) {
> >>                         const char *type =
> >> esl_event_get_header(handle.last_event, "content-type");
> >>                         if (type && !strcasecmp(type,
> >> "text/disconnect-notice")) {
> >>                                 const char *dispo =
> >> esl_event_get_header(handle.last_event, "content-disposition");
> >>                                 esl_log(ESL_LOG_INFO, "Got a
> disconnection
> >> notice dispostion: [%s]\n", dispo ? dispo : "");
> >>                                 if (!strcmp(dispo, "linger")) {
> >>                                         done = 1;
> >>                                         esl_log(ESL_LOG_INFO, "Waiting 5
> >> seconds for any remaining events.\n");
> >>                                         exp = time(NULL) + 5;
> >>                                 }
> >>                         }
> >>                 }
> >>         }
> >>
> >>         esl_log(ESL_LOG_INFO, "Disconnected! %d\n", handle.sock);
> >>         esl_disconnect(&handle);
> >>
> >>         close(client_sock);
> >>
> >>         _exit(0);
> >> }
> >>
> >> /*
> >>  * The signal handler function -- only gets called when a SIGCHLD
> >>  * is received, ie when a child terminates
> >>  */
> >> void sig_chld(int signo)
> >> {
> >>     int status;
> >>
> >>     /* Wait for any child without blocking */
> >>     if (waitpid(-1, &status, WNOHANG) < 0)
> >>     {
> >>         /*
> >>          * calling standard I/O functions like fprintf() in a
> >>          * signal handler is not recommended, but probably OK
> >>          * in toy programs like this one.
> >>          */
> >>         fprintf(stderr, "waitpid failed\n");
> >>         return;
> >>     }
> >> }
> >>
> >> int main(void)
> >> {
> >>         struct sigaction act;
> >>
> >>         /* Assign sig_chld as our SIGCHLD handler */
> >>         act.sa_handler = sig_chld;
> >>
> >>         /* We don't want to block any other signals in this example */
> >>         sigemptyset(&act.sa_mask);
> >>
> >>         /*
> >>          * We're only interested in children that have terminated, not
> >> ones
> >>          * which have been stopped (eg user pressing control-Z at
> >> terminal)
> >>          */
> >>         act.sa_flags = SA_NOCLDSTOP;
> >>
> >>         /*
> >>          * Make these values effective. If we were writing a real
> >>          * application, we would probably save the old value instead of
> >>          * passing NULL.
> >>          */
> >> /*      if (sigaction(SIGCHLD, &act, NULL) < 0)
> >>         {
> >>                 fprintf(stderr, "sigaction failed\n");
> >>                 return 1;
> >>         }
> >> */
> >>         signal(SIGCHLD, SIG_IGN);
> >>
> >>         esl_global_set_default_logger(0);
> >>         esl_listen("localhost", 8088, mycallback);
> >>
> >>         return 0;
> >> }
> >>
> >>
> >>
> >>
> >> On Thu, Aug 11, 2011 at 9:59 PM, Anthony Minessale
> >> <anthony.minessale at gmail.com> wrote:
> >>>
> >>> try removing the filter and event subscriptions
> >>> it's costly to consume all of the events especially at 75cps.
> >>>
> >>>
> >>> On Thu, Aug 11, 2011 at 5:23 AM, Tihomir Culjaga <tculjaga at gmail.com>
> >>> wrote:
> >>> > hello,
> >>> >
> >>> > im wondering how much performance do we loose when using ESL instead
> of
> >>> > running it via dialplan?
> >>> >
> >>> >
> >>> > without ESL with a fine tuned FS and a short dialplan ( answer,
> >>> > playback
> >>> > like 20 seconds file, hangup ) im able to service 75 CPS. On the same
> >>> > FS,
> >>> > when i use ESL to answer the call, playback the same file and hangup,
> >>> > im not
> >>> > able to run more than 2 CPS... this is a huge impact and i really
> can't
> >>> > believe it.
> >>> >
> >>> > I'm using event-socket outbound e.g.:
> >>> >
> >>> > <action application="socket" data="127.0.0.1:8088 async full"/>
> >>> >
> >>> > my extension looks like:
> >>> >
> >>> > <extension name="ESL_C">
> >>> >   <condition field="destination_number" expression="^(6666)$">
> >>> >     <action application="socket" data="127.0.0.1:8088 async full"/>
> >>> >     <action application="sleep" data="1000"/>
> >>> >     <action application="hangup"/>
> >>> >   </condition>
> >>> > </extension>
> >>> >
> >>> >
> >>> > im using testserver from lib/esl/ and i just removed the conference
> >>> > command
> >>> > and added the playback one.... also i moved the esl_debug lvl to 0
> >>> >
> >>> >
> >>> > anyhow, FS cannot run more than 2 CPS compared to 75 CPS when the
> >>> > playback
> >>> > is done from the dialplan.
> >>> >
> >>> >
> >>> > Please, can someone give me a clue on what is going on?
> >>> > Maybe im doing something wrong?
> >>> > how to get maximum FS performance using ESL ?
> >>> >
> >>> >
> >>> >
> >>> > Regards,
> >>> > Tihomir.
> >>> >
> >>> >
> >>> > _______________________________________________
> >>> > Join us at ClueCon 2011, Aug 9-11, Chicago
> >>> > http://www.cluecon.com 877-7-4ACLUE
> >>> >
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >
> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > http://www.freeswitch.org
> >>> >
> >>> >
> >>>
> >>>
> >>>
> >>> --
> >>> Anthony Minessale II
> >>>
> >>> 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
> >>> googletalk:conf+888 at conference.freeswitch.org
> >>> pstn:+19193869900
> >>>
> >>> _______________________________________________
> >>> Join us at ClueCon 2011, Aug 9-11, Chicago
> >>> http://www.cluecon.com 877-7-4ACLUE
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110816/20f9bfc1/attachment-0001.html 

From x.liu at hw.ac.uk  Tue Aug 16 19:53:30 2011
From: x.liu at hw.ac.uk (xl127)
Date: Tue, 16 Aug 2011 16:53:30 +0100
Subject: [Freeswitch-users] any IVR example in C/C++?
In-Reply-To: <CAPerZE93NU+ccKdmGEHufb11rRv0S+L1m0=riKibtgUX2VLU1A@mail.gmail.com>
References: <4E43EE36.60209@hw.ac.uk>	<CAA4nhyAnC0HfxJJPGpOdiBGaG__gBf6Bp9KVi6spjDQgjLTZ5w@mail.gmail.com>	<CALXCt0paCyLs3ojFVeWk9uO7VuuuEgXd_owR_10o0p5L-WStag@mail.gmail.com>	<31462.1313279925@ccs.covici.com>
	<CAPerZE93NU+ccKdmGEHufb11rRv0S+L1m0=riKibtgUX2VLU1A@mail.gmail.com>
Message-ID: <4E4A927A.4050102@hw.ac.uk>

Thanks for everybody's points, helpful!

Cheers,
Xing


On 14/08/11 04:04, Campbell Steven wrote:
> Try here:
>
> http://fisheye.freeswitch.org/browse/freeswitch.git/src/mod/applications/mod_protovm
>
> Campbell
>
> On Sun, Aug 14, 2011 at 11:58 AM,<covici at ccs.covici.com>  wrote:
>> Where can I find Mock's voicemail -- I don't see it in contrib?
>>
>> Giovanni Maruzzelli<gmaruzz at gmail.com>  wrote:
>>
>>> Also, you can check mod-voicemail.c and the new mod-voicemail made by
>>> Moc, those are the only IVR written in C that I know about (they're
>>> written in C because voicemail is considered a base feature, and been
>>> written in C assure stability because people does not fiddle with
>>> them)
>>>
>>> On 8/13/11, Moises Silva<moises.silva at gmail.com>  wrote:
>>>> On Thu, Aug 11, 2011 at 10:59 AM, xl127<x.liu at hw.ac.uk>  wrote:
>>>>> I am wondering how I could do this for a C/C++ application?
>>>>> And in the scripts languages I can set a callback method, e.g.
>>>>>    session.setInputCallback(myInputCallback)
>>>>> but I didn't find how to do this in C/C++.
>>>> The default question here is, why do you need C/C++ for an IVR?
>>>> FreeSWITCH allows you to use simpler/safer languages to build IVR's.
>>>>
>>>> You can certainly do it, but the reason you don't find examples is
>>>> probably because most people understand there is no need for C/C++
>>>> there. Having said that, you can take a look at the IVR/say/play API's
>>>> in switch_ivr_play_say.c to find out how to provide a callback to the
>>>> different API's thru the switch_input_args_t structure.
>>>>
>>>> Moises Silva
>>>> Senior Software Engineer, Software Development Manager
>>>> Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
>>>> L3R 9R6 Canada
>>>> t. 1 905 474 1990 x128 | e. moy at sangoma.com
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>> --
>>> Sent from my mobile device
>>>
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> Cell : +39-347-2665618
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> --
>> Your life is like a penny.  You're going to lose it.  The question is:
>> How do
>> you spend it?
>>
>>          John Covici
>>          covici at ccs.covici.com
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Heriot-Watt University is a Scottish charity
registered under charity number SC000278.



From bryan at bryanlemon.com  Tue Aug 16 17:46:16 2011
From: bryan at bryanlemon.com (Bryan Lemon)
Date: Tue, 16 Aug 2011 09:46:16 -0400
Subject: [Freeswitch-users] Question about ext-rtp-ip and ext-sip-ip
Message-ID: <CAEjbKNQ3tgWrVrr5uiTVKzv53avDLp3cRoavLuZsOiJfUaW_fg@mail.gmail.com>

>From what I am seeing, freeswitch is not honoring the ext-*-ip variables in
the invite messages. Using the following command entered on
fs_cli: originate
{origination_caller_id_name='Something.com',origination_caller_id_number=5555551212,userid=7,rowid=ROWID,phonenumber=5555551212,initial=2,prompt=0,thankyou=0,whattosay='',ignore_early_media=true}sofia/gateway/didforsale/15555551212
&javascript(somejavascript.js), the invite message is below. Shouldn't the
instances of 10.0.10.144 be replaced with the ext-*-ip of 204.111.*.*? This
is causing the rtp packets to be sent to the incorrect location, and
resulting in 1-way audio.


send 1089 bytes to udp/[209.216.*.*]:5060 at 05:56:00.276988:
   ------------------------------------------------------------------------
   INVITE sip:13044150838<https://www.google.com/voice/m/caller?number=+13044150838>@209.216.*.*
SIP/2.0
   Via: SIP/2.0/UDP 10.0.10.144:5080;rport;branch=z9hG4bKH0e2DU1Bc2KgD
   Max-Forwards: 69
   From: "SomeName" <sip:bryan_bryanlemon_com at 209.216.
*.*;transport=udp>;tag=HpU27XSQHmX1g
   To: <sip:15555551212 at 209.216.*.*>
   Call-ID: 425293d9-426f-122f-8fb5-f04da2846e9a
   CSeq: 16401016 INVITE
   Contact: <sip:gw+didforsale at 204.111.*.*:5080;transport=udp;gw=didforsale>
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-decfdbb 2011-08-11 14-15-26
-0500
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, hold, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 203
   X-FS-Support: update_display
   Remote-Party-ID: "SomeName" <sip:15555551212 at 10.0.10.144
>;party=calling;screen=yes;privacy=off

   v=0
   o=FreeSWITCH 1313441448 1313441449 IN IP4 10.0.10.144
   s=FreeSWITCH
   c=IN IP4 10.0.10.144
   t=0 0
   m=audio 32712 RTP/AVP 8 0 3 101 13
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20




sofia status profile internal
=================================================================================================
Name              internal
Domain Name       N/A
Auto-NAT          true
DBName            sofia_reg_internal
Pres Hosts        10.0.10.144,10.0.10.144
Dialplan          XML
Context           public
Challenge Realm   auto_from
RTP-IP            10.0.10.144
Ext-RTP-IP        204.111.*.*
SIP-IP            10.0.10.144
Ext-SIP-IP        204.111.*.*
URL               sip:mod_sofia at 10.0.10.144:5060
BIND-URL          sip:mod_sofia at 10.0.10.144:5060


freeswitch at internal> sofia status profile external
=================================================================================================
Name              external
Domain Name       N/A
Auto-NAT          true
DBName            sofia_reg_external
Pres Hosts
Dialplan          XML
Context           public
Challenge Realm   auto_to
RTP-IP            10.0.10.144
Ext-RTP-IP        204.111.*.*
SIP-IP            10.0.10.144
Ext-SIP-IP        204.111.*.*
URL               sip:mod_sofia at 10.0.10.144:5080
BIND-URL          sip:mod_sofia at 10.0.10.144:5080


Thank you,
Bryan Lemon
(302) 648-2747 <https://www.google.com/voice/m/caller?number=+13026482747>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110816/2c9fdb0a/attachment.html 

From jchavanton at gmail.com  Tue Aug 16 18:13:36 2011
From: jchavanton at gmail.com (Julien Chavanton)
Date: Tue, 16 Aug 2011 10:13:36 -0400
Subject: [Freeswitch-users] g729 licenses usage
Message-ID: <CAKmcL2nwh-Jp_1wAf4+MtUxe0c_PWZbwy1cJzg7pPOb1mT5PAw@mail.gmail.com>

Hi,

Before we purchase more g729 licenses we would like to confirm, how many
licenses are used when you have two call leg (g.729)  bridged with session
recording ?

I would expect 2 licenses, is this correct ?

------------------------------------------------------

I am sure I am not alone to hate this kind of licensing limitation, after
all theses years do we still need g.729, if it did not exist it would be
replaced by something just as good or better.

I know this as nothing to do with FS, since the market abuse or takeover of
g.729 is enforced everywhere in the network switches.


<freeswitch-users at lists.freeswitch.org>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110816/826caf68/attachment.html 

From lrmipsum0 at gmail.com  Tue Aug 16 18:58:14 2011
From: lrmipsum0 at gmail.com (Lorem Ipsum)
Date: Tue, 16 Aug 2011 16:58:14 +0200
Subject: [Freeswitch-users] Problem with receiving a NOTIFY after sending a
	SUBSCRIBE request
Message-ID: <CANSA4prMK6zbUX4xBQdBKkFS1qxuEg0cRacSpsJY=Jn51B_e6w@mail.gmail.com>

Hello,

I'm testing a SIP stack for an embedded device. The device, among other
things, is capable of informing a user about pending messages on the
voicemail. It does that by subscribing to the message-summary. Below some
wireshark traces (172.16.30.68 is my device, 172.16.31.10 is FreeSWITCH):

REGISTER sip:172.16.31.10 SIP/2.0
Via: SIP/2.0/UDP 172.16.30.68:5080
;rport;branch=z9hG4bKPj051b000000035ea40edf
Route: <sip:172.16.31.10:5060;lr>
Max-Forwards: 70
From: <sip:399510002 at 172.16.31.10>;tag=051b000000025ea40edf
To: <sip:399510002 at 172.16.31.10>
Call-ID: 051b000000015ea40edf
CSeq: 1 REGISTER
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY
User-Agent: My_Sip_Device
Contact: <sip:399510002 at 172.16.30.68:5080>;transport=udp
Content-Length:  0

SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.16.30.68:5080
;rport=5080;branch=z9hG4bKPj051b000000035ea40edf
From: <sip:399510002 at 172.16.31.10>;tag=051b000000025ea40edf
To: <sip:399510002 at 172.16.31.10>;tag=mQrcXUcvrtUmS
Call-ID: 051b000000015ea40edf
CSeq: 1 REGISTER
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-38e3f5f 2011-08-09 03-09-19
-0400
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
WWW-Authenticate: Digest realm="172.16.31.10",
nonce="3139f996-c814-11e0-93d2-05aa0ee343d6", algorithm=MD5, qop="auth"
Content-Length: 0

REGISTER sip:172.16.31.10 SIP/2.0
Via: SIP/2.0/UDP 172.16.30.68:5080
;rport;branch=z9hG4bKPj051b000000065ea40edf
Route: <sip:172.16.31.10:5060;lr>
Max-Forwards: 70
From: <sip:399510002 at 172.16.31.10>;tag=051b000000045ea40edf
To: <sip:399510002 at 172.16.31.10>
Call-ID: 051b000000015ea40edf
CSeq: 2 REGISTER
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY
User-Agent: My_Sip_Device
Contact: <sip:399510002 at 172.16.30.68:5080>;transport=udp
Authorization: Digest username="399510002", realm="172.16.31.10",
nonce="3139f996-c814-11e0-93d2-05aa0ee343d6", uri="sip:172.16.31.10",
response="1e95409a562c074cbe6df148a85107ef", algorithm=MD5,
cnonce="051b000000055ea40edf", qop=auth, nc=00000001
Content-Length:  0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.30.68:5080
;rport=5080;branch=z9hG4bKPj051b000000065ea40edf
From: <sip:399510002 at 172.16.31.10>;tag=051b000000045ea40edf
To: <sip:399510002 at 172.16.31.10>;tag=N0H5ypXZN3H7m
Call-ID: 051b000000015ea40edf
CSeq: 2 REGISTER
Contact: <sip:399510002 at 172.16.30.68:5080>;transport=udp;expires=180
Date: Tue, 16 Aug 2011 14:29:47 GMT
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-38e3f5f 2011-08-09 03-09-19
-0400
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Content-Length: 0


SUBSCRIBE sip:172.16.31.10 SIP/2.0
Via: SIP/2.0/UDP 172.16.30.68:5080
;rport;branch=z9hG4bKPj051b0000000a5ea40edf
Max-Forwards: 69
From: "399510002" <sip:399510002 at 172.16.31.10>;tag=051b000000085ea40edf
To: <sip:172.16.31.10>
Contact: <sip:399510002 at 172.16.30.68:5080>
Call-ID: 051b000000095ea40edf
CSeq: 1 SUBSCRIBE
Event: message-summary
Accept: application/simple-message-summary
Allow-Events: message-summary
User-Agent: My_Sip_Device
X-Serialnumber: LMZ091218000026
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY
Route: <sip:172.16.31.10:5060;lr>
Content-Length:  0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.30.68:5080
;rport=5080;branch=z9hG4bKPj051b0000000a5ea40edf
From: "399510002" <sip:399510002 at 172.16.31.10>;tag=051b000000085ea40edf
To: <sip:172.16.31.10>;tag=p9ay0He3jc8Sg
Call-ID: 051b000000095ea40edf
CSeq: 1 SUBSCRIBE
Contact: <sip:@172.16.31.10:5060>
Expires: 60
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-38e3f5f 2011-08-09 03-09-19
-0400
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
Subscription-State: active;expires=60
Content-Length: 0


After sending the 200 OK, FreeSWITCH does not send the NOTIFY.
If you look at the Contact header of the answer to the SUBSCRIBE you will
notice that the part before the "@" is missing. I guess this is because
SUBSCRIBE request does not contain the whole URI, just the host part. That
is because our customer wants it done this way; the request line should look
like this:
SUBSCRIBE sip:voicemail_server SIP/2.0

and the To: header should look like this:
To: <sip:voicemail_server>

My question is: how can I make FreeSWITCH (the NOTIFY part anyway, other
things are working OK) work with such a device? Thanks.

Regards,
Tom
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110816/3eded18f/attachment-0001.html 

From SureshM at telesoftlabs.com  Tue Aug 16 12:07:08 2011
From: SureshM at telesoftlabs.com (Suresh M)
Date: Tue, 16 Aug 2011 13:37:08 +0530
Subject: [Freeswitch-users] mod_event_socket Outbound connection in Java.
Message-ID: <81BCC027A3DF104B89991D9E3621F9B42AF328@tslsrv.TSL.local>

Hi,
 
I am trying mod_event_socket in FreeSwitch [outbound ] using Java and I
am stuck:
 
I tried the following C# sample successfully.
 
ESLconnection eslConnection = new
ESLconnection(sckClient.Handle.ToInt32());
 
I need to do the same in Java. But what should I pass to new
ESLconnection constructor in Java in place of socket handle?!  I know
basically this is a Java question for which I tried a lot to find answer
but in vain.  Hope somebody out there would already have come across
this and got a solution.  Any help or clue or alternative method to
achieve the same is greatly appreciated.
 
Thanks in advance.
 
Suresh
 
 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110816/f799f7ce/attachment.html 

From brian at freeswitch.org  Tue Aug 16 21:03:14 2011
From: brian at freeswitch.org (Brian West)
Date: Tue, 16 Aug 2011 12:03:14 -0500
Subject: [Freeswitch-users] Question about ext-rtp-ip and ext-sip-ip
In-Reply-To: <CAEjbKNQ3tgWrVrr5uiTVKzv53avDLp3cRoavLuZsOiJfUaW_fg@mail.gmail.com>
References: <CAEjbKNQ3tgWrVrr5uiTVKzv53avDLp3cRoavLuZsOiJfUaW_fg@mail.gmail.com>
Message-ID: <65727391-DF08-4074-BB7F-BDB766DF7942@freeswitch.org>

Bryan,
	Can you provide the sofia profile xml?  

/b

On Aug 16, 2011, at 8:46 AM, Bryan Lemon wrote:

>> From what I am seeing, freeswitch is not honoring the ext-*-ip variables in
> the invite messages. Using the following command entered on
> fs_cli: originate
> {origination_caller_id_name='Something.com',origination_caller_id_number=5555551212,userid=7,rowid=ROWID,phonenumber=5555551212,initial=2,prompt=0,thankyou=0,whattosay='',ignore_early_media=true}sofia/gateway/didforsale/15555551212
> &javascript(somejavascript.js), the invite message is below. Shouldn't the
> instances of 10.0.10.144 be replaced with the ext-*-ip of 204.111.*.*? This
> is causing the rtp packets to be sent to the incorrect location, and
> resulting in 1-way audio.
> 
> 
> send 1089 bytes to udp/[209.216.*.*]:5060 at 05:56:00.276988:
>   ------------------------------------------------------------------------
>   INVITE sip:13044150838<https://www.google.com/voice/m/caller?number=+13044150838>@209.216.*.*
> SIP/2.0
>   Via: SIP/2.0/UDP 10.0.10.144:5080;rport;branch=z9hG4bKH0e2DU1Bc2KgD
>   Max-Forwards: 69
>   From: "SomeName" <sip:bryan_bryanlemon_com at 209.216.
> *.*;transport=udp>;tag=HpU27XSQHmX1g
>   To: <sip:15555551212 at 209.216.*.*>
>   Call-ID: 425293d9-426f-122f-8fb5-f04da2846e9a
>   CSeq: 16401016 INVITE
>   Contact: <sip:gw+didforsale at 204.111.*.*:5080;transport=udp;gw=didforsale>
>   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-decfdbb 2011-08-11 14-15-26
> -0500
>   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY
>   Supported: timer, precondition, path, replaces
>   Allow-Events: talk, hold, refer
>   Content-Type: application/sdp
>   Content-Disposition: session
>   Content-Length: 203
>   X-FS-Support: update_display
>   Remote-Party-ID: "SomeName" <sip:15555551212 at 10.0.10.144
>> ;party=calling;screen=yes;privacy=off
> 
>   v=0
>   o=FreeSWITCH 1313441448 1313441449 IN IP4 10.0.10.144
>   s=FreeSWITCH
>   c=IN IP4 10.0.10.144
>   t=0 0
>   m=audio 32712 RTP/AVP 8 0 3 101 13
>   a=rtpmap:101 telephone-event/8000
>   a=fmtp:101 0-16
>   a=ptime:20
> 
> 
> 
> 
> sofia status profile internal
> =================================================================================================
> Name              internal
> Domain Name       N/A
> Auto-NAT          true
> DBName            sofia_reg_internal
> Pres Hosts        10.0.10.144,10.0.10.144
> Dialplan          XML
> Context           public
> Challenge Realm   auto_from
> RTP-IP            10.0.10.144
> Ext-RTP-IP        204.111.*.*
> SIP-IP            10.0.10.144
> Ext-SIP-IP        204.111.*.*
> URL               sip:mod_sofia at 10.0.10.144:5060
> BIND-URL          sip:mod_sofia at 10.0.10.144:5060
> 
> 
> freeswitch at internal> sofia status profile external
> =================================================================================================
> Name              external
> Domain Name       N/A
> Auto-NAT          true
> DBName            sofia_reg_external
> Pres Hosts
> Dialplan          XML
> Context           public
> Challenge Realm   auto_to
> RTP-IP            10.0.10.144
> Ext-RTP-IP        204.111.*.*
> SIP-IP            10.0.10.144
> Ext-SIP-IP        204.111.*.*
> URL               sip:mod_sofia at 10.0.10.144:5080
> BIND-URL          sip:mod_sofia at 10.0.10.144:5080
> 
> 
> Thank you,
> Bryan Lemon
> (302) 648-2747 <https://www.google.com/voice/m/caller?number=+13026482747>
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From lakersman2006 at yahoo.com  Tue Aug 16 21:22:23 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Tue, 16 Aug 2011 10:22:23 -0700 (PDT)
Subject: [Freeswitch-users] ORIGINATE_DISPOSITION
In-Reply-To: <CAKzWOxWtxNoto1c6Chu5dZLPkt+ATcwCGpHOYtPpWydUrcCFfQ@mail.gmail.com>
References: <1313447714.89555.YahooMailNeo@web161001.mail.bf1.yahoo.com>
	<CAKzWOxVwDaxsFJAr1dVP5nfmDuFXK2tptvai+oqtwZgph3u1kw@mail.gmail.com>
	<1313452154.28177.YahooMailNeo@web161001.mail.bf1.yahoo.com>
	<CAKzWOxWYXy5Sa2g+cCDEDcxWwE10OtrMaJzUyoLdFSDGtDfbyg@mail.gmail.com>
	<1313452622.11918.YahooMailNeo@web161012.mail.bf1.yahoo.com>
	<CAKzWOxVREFK=uzYzT95oXFT0-sURz7yeOwkuQMz6LKjQ6GvoXA@mail.gmail.com>
	<1313453126.47357.YahooMailNeo@web161010.mail.bf1.yahoo.com>
	<CAKzWOxWtxNoto1c6Chu5dZLPkt+ATcwCGpHOYtPpWydUrcCFfQ@mail.gmail.com>
Message-ID: <1313515343.27543.YahooMailNeo@web161004.mail.bf1.yahoo.com>

MC,

Here is the pastebin http://pastebin.freeswitch.org/17056 of my perl script. For the script I just made it display the various channel variables so I can see what values freeswitch will provide me after the call has been bridged or not.



________________________________
From: Michael Collins <msc at freeswitch.org>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Monday, August 15, 2011 5:30 PM
Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION





On Mon, Aug 15, 2011 at 5:05 PM, Sam <lakersman2006 at yahoo.com> wrote:

I am using perl's $session->get_variable("originate_disposition");
Are you looking at the b-leg's session?
?

>
>Also, how come the "hangup_time" shows zero on answered calls?
Because hangup_time refers to the point in time at which the call was hung up. Since you are in the middle of a call (using the $session object) you will never see the hangup_time because the object ceases to exist once the call leg is disconnected.

I get the impression that you may be using the wrong tool for this particular job, but I'm not sure without seeing it. If you don't mind dropping it on pastebin we'll have a look and give you some suggestions.

-MC


>
>
>________________________________
> From: Michael Collins <msc at freeswitch.org>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>Sent: Monday, August 15, 2011 4:58 PM
>
>Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>
>
>
>And how are you checking the variable? Do you have an event socket open or ... ?
>-MC
>
>
>On Mon, Aug 15, 2011 at 4:57 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
>It is being generated with a bridge.
>>
>>
>>
>>
>>________________________________
>>From: Michael Collins <msc at freeswitch.org>
>>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>Sent: Monday, August 15, 2011 4:54 PM
>>
>>Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>
>>
>>
>>And how is it being generated? WIth a bridge or originate or ... ?
>>-MC
>>
>>
>>On Mon, Aug 15, 2011 at 4:49 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>
>>I want to know the B-leg status of the call.
>>>
>>>
>>>
>>>
>>>________________________________
>>>From: Michael Collins <msc at freeswitch.org>
>>>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>Sent: Monday, August 15, 2011 4:31 PM
>>>Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>>
>>>
>>>
>>>I think you may be wanting "endpoint_disposition" depending on exactly what you're looking at.
>>>
>>>
>>>-MC
>>>
>>>
>>>On Mon, Aug 15, 2011 at 3:35 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>>
>>>For the ORIGINATE_DISPOSITION channel variable, does "SUCCESS" mean the call was ANSWERED?
>>>>
>>>>
>>>>FreeSWITCH-users mailing list
>>>>FreeSWITCH-users at lists.freeswitch.org
>>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>>FreeSWITCH-users mailing list
>>>FreeSWITCH-users at lists.freeswitch.org
>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>>FreeSWITCH-users mailing list
>>>FreeSWITCH-users at lists.freeswitch.org
>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>http://www.freeswitch.org
>>>
>>>
>>
>>
>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>>
>>
>>
>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>>
>
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>
>
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>


FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110816/e19d2520/attachment-0001.html 

From lakersman2006 at yahoo.com  Tue Aug 16 21:26:25 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Tue, 16 Aug 2011 10:26:25 -0700 (PDT)
Subject: [Freeswitch-users] ORIGINATE_DISPOSITION
In-Reply-To: <CA+GmxbZ2RG+RS79F8yjXAR4eeFwFy3EMV0k4YiESwq3qMCbwkA@mail.gmail.com>
References: <1313447714.89555.YahooMailNeo@web161001.mail.bf1.yahoo.com>
	<CAKzWOxVwDaxsFJAr1dVP5nfmDuFXK2tptvai+oqtwZgph3u1kw@mail.gmail.com>
	<1313452154.28177.YahooMailNeo@web161001.mail.bf1.yahoo.com>
	<CAKzWOxWYXy5Sa2g+cCDEDcxWwE10OtrMaJzUyoLdFSDGtDfbyg@mail.gmail.com>
	<1313452622.11918.YahooMailNeo@web161012.mail.bf1.yahoo.com>
	<CAKzWOxVREFK=uzYzT95oXFT0-sURz7yeOwkuQMz6LKjQ6GvoXA@mail.gmail.com>
	<1313453126.47357.YahooMailNeo@web161010.mail.bf1.yahoo.com>
	<CAKzWOxWtxNoto1c6Chu5dZLPkt+ATcwCGpHOYtPpWydUrcCFfQ@mail.gmail.com>
	<CA+GmxbZ2RG+RS79F8yjXAR4eeFwFy3EMV0k4YiESwq3qMCbwkA@mail.gmail.com>
Message-ID: <1313515585.77893.YahooMailNeo@web161019.mail.bf1.yahoo.com>

Dmitry,

Yes, I agree. For an novice like me it was extremely confusing and frustrating that I could not get the proper B leg disposition when A leg is answered by FS itself.

I hope there can be some methods or functions that will allow us to get the proper B Leg disposition.



________________________________
From: Dmitry Sytchev <kbdfck at gmail.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, August 16, 2011 2:30 AM
Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION


Seems we need to clear things about B-leg disposition in wiki.
AFAIK there is no method to get correct disposition from B leg without analyzing events in case when A-leg was answered by FS itself.?

If A leg was not answered, we can use A-leg disposition for call disposition.


2011/8/16 Michael Collins <msc at freeswitch.org>


>
>
>On Mon, Aug 15, 2011 at 5:05 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
>I am using perl's $session->get_variable("originate_disposition");
>Are you looking at the b-leg's session?
>?
>
>>
>>Also, how come the "hangup_time" shows zero on answered calls?
>Because hangup_time refers to the point in time at which the call was hung up. Since you are in the middle of a call (using the $session object) you will never see the hangup_time because the object ceases to exist once the call leg is disconnected.
>
>
>I get the impression that you may be using the wrong tool for this particular job, but I'm not sure without seeing it. If you don't mind dropping it on pastebin we'll have a look and give you some suggestions.
>
>-MC
>
>
>
>>
>>
>>________________________________
>> From: Michael Collins <msc at freeswitch.org>
>>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>Sent: Monday, August 15, 2011 4:58 PM
>>
>>Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>
>>
>>
>>And how are you checking the variable? Do you have an event socket open or ... ?
>>-MC
>>
>>
>>On Mon, Aug 15, 2011 at 4:57 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>
>>It is being generated with a bridge.
>>>
>>>
>>>
>>>
>>>________________________________
>>>From: Michael Collins <msc at freeswitch.org>
>>>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>Sent: Monday, August 15, 2011 4:54 PM
>>>
>>>Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>>
>>>
>>>
>>>And how is it being generated? WIth a bridge or originate or ... ?
>>>-MC
>>>
>>>
>>>On Mon, Aug 15, 2011 at 4:49 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>>
>>>I want to know the B-leg status of the call.
>>>>
>>>>
>>>>
>>>>
>>>>________________________________
>>>>From: Michael Collins <msc at freeswitch.org>
>>>>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>>Sent: Monday, August 15, 2011 4:31 PM
>>>>Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>>>
>>>>
>>>>
>>>>I think you may be wanting "endpoint_disposition" depending on exactly what you're looking at.
>>>>
>>>>
>>>>-MC
>>>>
>>>>
>>>>On Mon, Aug 15, 2011 at 3:35 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>>>
>>>>For the ORIGINATE_DISPOSITION channel variable, does "SUCCESS" mean the call was ANSWERED?
>>>>>
>>>>>
>>>>>FreeSWITCH-users mailing list
>>>>>FreeSWITCH-users at lists.freeswitch.org
>>>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>>FreeSWITCH-users mailing list
>>>>FreeSWITCH-users at lists.freeswitch.org
>>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>>FreeSWITCH-users mailing list
>>>>FreeSWITCH-users at lists.freeswitch.org
>>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>>FreeSWITCH-users mailing list
>>>FreeSWITCH-users at lists.freeswitch.org
>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>>FreeSWITCH-users mailing list
>>>FreeSWITCH-users at lists.freeswitch.org
>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>http://www.freeswitch.org
>>>
>>>
>>
>>
>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>>
>>
>>
>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>>
>
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>


-- 
Best regards,

Dmitry Sytchev,
IT Engineer


FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110816/b274481a/attachment.html 

From nasida at live.ru  Tue Aug 16 21:46:35 2011
From: nasida at live.ru (Yuriy Nasida)
Date: Tue, 16 Aug 2011 21:46:35 +0400
Subject: [Freeswitch-users] Lua not playing wav files
Message-ID: <SNT127-W400A9FAAD362C48C2F3B5CD5290@phx.gbl>



Hi
Freeswitch-users,




My
simple lua script:




freeswitch.consoleLog("err","start
hello.lua\n")

session:answer();

message
= "ivr/ivr-enter_destination_telephone_number.wav"

session:execute("playback",
message)

session:hangup();




Script
looks fine I think, but FS doesn't play audio. If I use corresponding
XML dialplan all work fine.







logs
when I use lua:




2011-08-16
13:18:00.336003 [DEBUG] switch_core_state_machine.c:371
(sofia/external/79213777785 at 65.98.107.130:5080) State EXECUTE going
to sleep

2011-08-16
13:18:00.336003 [DEBUG] switch_core_state_machine.c:364
(sofia/external/79213777785 at 65.98.107.130:5080) State ROUTING

2011-08-16
13:18:00.336003 [DEBUG] mod_sofia.c:147
sofia/external/79213777785 at 65.98.107.130:5080 SOFIA ROUTING

2011-08-16
13:18:00.336003 [DEBUG] switch_core_state_machine.c:77
sofia/external/79213777785 at 65.98.107.130:5080 Standard ROUTING

2011-08-16
13:18:00.336003 [INFO] mod_dialplan_xml.c:331 Processing unknown
<79213777785>->inbound_type_uri in context public

2011-08-16
13:18:00.345157 [ERR] switch_cpp.cpp:1197 start hello.lua

2011-08-16
13:18:00.345157 [DEBUG] sofia_glue.c:4650 Audio Codec Compare
[PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000]

2011-08-16
13:18:00.345157 [DEBUG] sofia_glue.c:4650 Audio Codec Compare
[PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]

2011-08-16
13:18:00.345157 [DEBUG] sofia_glue.c:2773 Set Codec
sofia/external/79213777785 at 65.98.107.130:5080 PCMU/8000 20 ms 160
samples 64000 bits

2011-08-16
13:18:00.346166 [DEBUG] sofia_glue.c:4764 Set 2833 dtmf send/recv
payload to 101

2011-08-16
13:18:00.346166 [DEBUG] sofia_glue.c:3014 AUDIO RTP
[sofia/external/79213777785 at 65.98.107.130:5080] 65.98.107.130 port
26266 -> 212.232.72.134 port 49276 codec: 0 ms: 20

2011-08-16
13:18:00.346166 [DEBUG] switch_rtp.c:1623 Starting timer [soft] 160
bytes per 20ms

2011-08-16
13:18:00.347205 [DEBUG] sofia_glue.c:3276 Set 2833 dtmf send payload
to 101

2011-08-16
13:18:00.347205 [DEBUG] sofia_glue.c:3281 Set 2833 dtmf receive
payload to 101

2011-08-16
13:18:00.347205 [DEBUG] mod_sofia.c:681 Local SDP
sofia/external/79213777785 at 65.98.107.130:5080:

v=0

o=FreeSWITCH
1313488814 1313488815 IN IP4 65.98.107.130

s=FreeSWITCH

c=IN
IP4 65.98.107.130

t=0 0

m=audio
26266 RTP/AVP 0 101

a=rtpmap:0
PCMU/8000

a=rtpmap:101
telephone-event/8000

a=fmtp:101
0-16

a=silenceSupp:off
- - - -

a=ptime:20

a=sendrecv




2011-08-16
13:18:00.348214 [DEBUG] sofia.c:4761 Channel
sofia/external/79213777785 at 65.98.107.130:5080 entering state
[completed][200]

2011-08-16
13:18:00.348214 [DEBUG] switch_core_session.c:1939 Application
playback Requires media! pre_answering channel
sofia/external/79213777785 at 65.98.107.130:5080

2011-08-16
13:18:00.348214 [DEBUG] switch_cpp.cpp:618 CoreSession::hangup

2011-08-16
13:18:00.348214 [DEBUG] switch_cpp.cpp:988
sofia/external/79213777785 at 65.98.107.130:5080 destroy/unlink session
from object

2011-08-16
13:18:00.348214 [DEBUG] switch_core_state_machine.c:364
(sofia/external/79213777785 at 65.98.107.130:5080) State ROUTING going
to sleep"










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




logs
when I use XML dialplan:




2011-08-16
13:19:49.609765 [DEBUG] switch_core_state_machine.c:371
(sofia/external/79213777785 at 65.98.107.130:5080) State EXECUTE going
to sleep

2011-08-16
13:19:49.609765 [DEBUG] switch_core_state_machine.c:364
(sofia/external/79213777785 at 65.98.107.130:5080) State ROUTING

2011-08-16
13:19:49.609765 [DEBUG] mod_sofia.c:147
sofia/external/79213777785 at 65.98.107.130:5080 SOFIA ROUTING

2011-08-16
13:19:49.609765 [DEBUG] switch_core_state_machine.c:77
sofia/external/79213777785 at 65.98.107.130:5080 Standard ROUTING

2011-08-16
13:19:49.609765 [INFO] mod_dialplan_xml.c:331 Processing unknown
<79213777785>->inbound_type_uri in context public

2011-08-16
13:19:49.615894 [DEBUG] switch_core_state_machine.c:364
(sofia/external/79213777785 at 65.98.107.130:5080) State ROUTING going
to sleep

2011-08-16
13:19:49.615894 [DEBUG] switch_core_state_machine.c:371
(sofia/external/79213777785 at 65.98.107.130:5080) State EXECUTE

2011-08-16
13:19:49.615894 [DEBUG] mod_sofia.c:240
sofia/external/79213777785 at 65.98.107.130:5080 SOFIA EXECUTE

2011-08-16
13:19:49.615894 [DEBUG] switch_core_state_machine.c:157
sofia/external/79213777785 at 65.98.107.130:5080 Standard EXECUTE

2011-08-16
13:19:49.615894 [DEBUG] sofia_glue.c:4650 Audio Codec Compare
[PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000]

2011-08-16
13:19:49.615894 [DEBUG] sofia_glue.c:4650 Audio Codec Compare
[PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]

2011-08-16
13:19:49.616902 [DEBUG] sofia_glue.c:2773 Set Codec
sofia/external/79213777785 at 65.98.107.130:5080 PCMU/8000 20 ms 160
samples 64000 bits

2011-08-16
13:19:49.616902 [DEBUG] sofia_glue.c:4764 Set 2833 dtmf send/recv
payload to 101

2011-08-16
13:19:49.616902 [DEBUG] sofia_glue.c:3014 AUDIO RTP
[sofia/external/79213777785 at 65.98.107.130:5080] 65.98.107.130 port
23946 -> 212.232.72.134 port 49278 codec: 0 ms: 20

2011-08-16
13:19:49.616902 [DEBUG] switch_rtp.c:1623 Starting timer [soft] 160
bytes per 20ms

2011-08-16
13:19:49.617998 [DEBUG] sofia_glue.c:3276 Set 2833 dtmf send payload
to 101

2011-08-16
13:19:49.617998 [DEBUG] sofia_glue.c:3281 Set 2833 dtmf receive
payload to 101

2011-08-16
13:19:49.617998 [DEBUG] mod_sofia.c:681 Local SDP
sofia/external/79213777785 at 65.98.107.130:5080:

v=0

o=FreeSWITCH
1313491243 1313491244 IN IP4 65.98.107.130

s=FreeSWITCH

c=IN
IP4 65.98.107.130

t=0 0

m=audio
23946 RTP/AVP 0 101

a=rtpmap:0
PCMU/8000

a=rtpmap:101
telephone-event/8000

a=fmtp:101
0-16

a=silenceSupp:off
- - - -

a=ptime:20

a=sendrecv




2011-08-16
13:19:49.619008 [DEBUG] sofia.c:4761 Channel
sofia/external/79213777785 at 65.98.107.130:5080 entering state
[completed][200]

2011-08-16
13:19:49.631344 [DEBUG] switch_ivr_play_say.c:1278 Codec Activated
L16 at 8000hz 1 channels 20ms

2011-08-16
13:19:49.740067 [DEBUG] sofia.c:4761 Channel
sofia/external/79213777785 at 65.98.107.130:5080 entering state
[ready][200]

2011-08-16
13:19:52.879234 [DEBUG] switch_ivr_play_say.c:1648 done playing file

2011-08-16
13:19:52.880333 [NOTICE] switch_core_state_machine.c:189
sofia/external/79213777785 at 65.98.107.130:5080 has executed the last
dialplan instruction, hanging up.







I
have compared logs and saw that case without lua have some strings
unlike case with lua.




"2011-08-16
13:19:49.615894 [DEBUG] switch_core_state_machine.c:364
(sofia/external/79213777785 at 65.98.107.130:5080) State ROUTING going
to sleep

2011-08-16
13:19:49.615894 [DEBUG] switch_core_state_machine.c:371
(sofia/external/79213777785 at 65.98.107.130:5080) State EXECUTE

2011-08-16
13:19:49.615894 [DEBUG] mod_sofia.c:240
sofia/external/79213777785 at 65.98.107.130:5080 SOFIA EXECUTE

2011-08-16
13:19:49.615894 [DEBUG] switch_core_state_machine.c:157
sofia/external/79213777785 at 65.98.107.130:5080 Standard EXECUTE"






Many thanks to anyone who can help.




 		 	   		  
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110816/4e8668fd/attachment-0001.html 

From anthony.minessale at gmail.com  Tue Aug 16 21:54:47 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Aug 2011 12:54:47 -0500
Subject: [Freeswitch-users] ORIGINATE_DISPOSITION
In-Reply-To: <1313515585.77893.YahooMailNeo@web161019.mail.bf1.yahoo.com>
References: <1313447714.89555.YahooMailNeo@web161001.mail.bf1.yahoo.com>
	<CAKzWOxVwDaxsFJAr1dVP5nfmDuFXK2tptvai+oqtwZgph3u1kw@mail.gmail.com>
	<1313452154.28177.YahooMailNeo@web161001.mail.bf1.yahoo.com>
	<CAKzWOxWYXy5Sa2g+cCDEDcxWwE10OtrMaJzUyoLdFSDGtDfbyg@mail.gmail.com>
	<1313452622.11918.YahooMailNeo@web161012.mail.bf1.yahoo.com>
	<CAKzWOxVREFK=uzYzT95oXFT0-sURz7yeOwkuQMz6LKjQ6GvoXA@mail.gmail.com>
	<1313453126.47357.YahooMailNeo@web161010.mail.bf1.yahoo.com>
	<CAKzWOxWtxNoto1c6Chu5dZLPkt+ATcwCGpHOYtPpWydUrcCFfQ@mail.gmail.com>
	<CA+GmxbZ2RG+RS79F8yjXAR4eeFwFy3EMV0k4YiESwq3qMCbwkA@mail.gmail.com>
	<1313515585.77893.YahooMailNeo@web161019.mail.bf1.yahoo.com>
Message-ID: <CAKbxfG9N1TUmT=HZQqhJfkLk_YiKc=XPqbBoa-9Rm=+Q97PZJw@mail.gmail.com>

The originate disposition of A leg will always have the last known status from B

originate, by default, returns when:

1) media is established on a specific outgoing leg, answered or not
2) all outgoing legs are terminated.

if you call somewhere that uses early media, the originate will end
setting originate_disposition to SUCCESS meaning that a live channel
was produced.  Now the bridge will begin between the A and B leg.

When the bridge ends if the B leg is hungup, its cause will be stored
in "bridge_hangup_cause"

if you want to keep the originate from ending when early media is
established you can add {ignore_early_media=true} or
{bridge_early_media=true} prepended to your dial string and the
originate will never return untill all outbound legs are either hungup
or one is answered.


You should take some time to expand your mind to the different
paradigm in FreeSWITCH where you may have as many as 10 outbound legs
at once in a forked-dial situation and some of what you think is
simple and obvious will quickly dissolve.

Another thing you can do is set the variable "failed_xml_cdr_prefix"
on the A leg.
This prefix will be mixed with an incrementing variable for each
outbound call leg and in the case of a failure the entire XML cdr will
be set into a var on A leg.

for instance if you set failed_xml_cdr_prefix=foo you would get foo_1
foo_2 etc depending on the number of outbound call legs.

Additionally you can set copy_xml_cdr on the A leg and when the bridge
ends you will get a complete CDR for B in the "b_leg_cdr" variable on
A

Finally you should really go with the flow of how FreeSWITCH is
engineered and try to keep your accounting logic in a separate place
and monitor the XML-CDR, CDR-CSV or event_socket +
CHANNEL_HANGUP_COMPLETE events to process this information.  There is
much more to a call and what happens when it's transferred etc than
what you can get in a single monolithic perspective of inside the
channel.  Plus it's conter intuitive to put routing, application and
accounting logic in the same place.



On Tue, Aug 16, 2011 at 12:26 PM, Sam <lakersman2006 at yahoo.com> wrote:
> Dmitry,
> Yes, I agree. For an novice like me it was extremely confusing and
> frustrating that I could not get the proper B leg disposition when A leg is
> answered by FS itself.
> I hope there can be some methods or functions that will allow us to get the
> proper B Leg disposition.
> ________________________________
> From: Dmitry Sytchev <kbdfck at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Sent: Tuesday, August 16, 2011 2:30 AM
> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>
> Seems we need to clear things about B-leg disposition in wiki.
> AFAIK there is no method to get correct disposition from B leg without
> analyzing events in case when A-leg was answered by FS itself.
> If A leg was not answered, we can use A-leg disposition for call
> disposition.
>
> 2011/8/16 Michael Collins <msc at freeswitch.org>
>
>
> On Mon, Aug 15, 2011 at 5:05 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> I am using perl's $session->get_variable("originate_disposition");
>
> Are you looking at the b-leg's session?
>
>
> Also, how come the "hangup_time" shows zero on answered calls?
>
> Because hangup_time refers to the point in time at which the call was hung
> up. Since you are in the middle of a call (using the $session object) you
> will never see the hangup_time because the object ceases to exist once the
> call leg is disconnected.
> I get the impression that you may be using the wrong tool for this
> particular job, but I'm not sure without seeing it. If you don't mind
> dropping it on pastebin we'll have a look and give you some suggestions.
> -MC
>
> ________________________________
> From: Michael Collins <msc at freeswitch.org>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Sent: Monday, August 15, 2011 4:58 PM
> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>
> And how are you checking the variable? Do you have an event socket open or
> ... ?
> -MC
>
> On Mon, Aug 15, 2011 at 4:57 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> It is being generated with a bridge.
>
> ________________________________
> From: Michael Collins <msc at freeswitch.org>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Sent: Monday, August 15, 2011 4:54 PM
> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>
> And how is it being generated? WIth a bridge or originate or ... ?
> -MC
>
> On Mon, Aug 15, 2011 at 4:49 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> I want to know the B-leg status of the call.
>
> ________________________________
> From: Michael Collins <msc at freeswitch.org>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Sent: Monday, August 15, 2011 4:31 PM
> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>
> I think you may be wanting "endpoint_disposition" depending on exactly what
> you're looking at.
> -MC
>
> On Mon, Aug 15, 2011 at 3:35 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> For the ORIGINATE_DISPOSITION channel variable, does "SUCCESS" mean the call
> was ANSWERED?
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Best regards,
>
> Dmitry Sytchev,
> IT Engineer
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From Hector.Geraldino at ip-soft.net  Tue Aug 16 22:01:52 2011
From: Hector.Geraldino at ip-soft.net (Hector Geraldino)
Date: Tue, 16 Aug 2011 14:01:52 -0400
Subject: [Freeswitch-users] mod_event_socket Outbound connection in Java.
In-Reply-To: <81BCC027A3DF104B89991D9E3621F9B42AF328@tslsrv.TSL.local>
References: <81BCC027A3DF104B89991D9E3621F9B42AF328@tslsrv.TSL.local>
Message-ID: <6A6B4C284AD15042B429EB9D904544AD021FD8A873@NY1-EXMB-01.ip-soft.net>

Hi Suresh,

I haven't used the C# libraries before, but I can highly recommend you to use the Java ESL client library  (org.freeswitch.esl.client) that is listed on the wiki.

http://wiki.freeswitch.org/wiki/Java_ESL

I'm using it in one of my current developments and it works pretty good. As it uses the JBoss netty for connection management, you don't have to worry about connection handling, missing events or anything like that. It's well designed and the performance is acceptable.

Give it a shot and let us know if you have any questions.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Suresh M
Sent: Tuesday, August 16, 2011 4:07 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] mod_event_socket Outbound connection in Java.

Hi,

I am trying mod_event_socket in FreeSwitch [outbound ] using Java and I am stuck:

I tried the following C# sample successfully.

ESLconnection eslConnection = new ESLconnection(sckClient.Handle.ToInt32());

I need to do the same in Java. But what should I pass to new ESLconnection constructor in Java in place of socket handle?!  I know basically this is a Java question for which I tried a lot to find answer but in vain.  Hope somebody out there would already have come across this and got a solution.  Any help or clue or alternative method to achieve the same is greatly appreciated.

Thanks in advance.

Suresh


-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110816/6ef67e82/attachment.html 

From vipkilla at gmail.com  Tue Aug 16 22:03:43 2011
From: vipkilla at gmail.com (vip killa)
Date: Tue, 16 Aug 2011 14:03:43 -0400
Subject: [Freeswitch-users] enterprise deployment not working
Message-ID: <CAC-LwPPSAei6EyeQBtrHoin0DtN73f2zfd6GU5JyQm0Z8_rXPw@mail.gmail.com>

I'm following tutorial at
http://wiki.freeswitch.org/wiki/Enterprise_deployment_<http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS>
OpenSIPS <http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS> and
opensips is not forwarding the registration requests...
Running opensips-1.6.4-2-tls,
Here is what the debugging looks like when i try to register with the
opensip box
10.20.30.17 is the opensips box.
10.20.30.18 is the FreeSWITCH box.


Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:core:parse_msg:
SIP Request:
Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:core:parse_msg:
 method:  <REGISTER>
Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:core:parse_msg:
 uri:     <sip:10.20.30.17>
Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:core:parse_msg:
 version: <SIP/2.0>
Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
DBG:core:parse_headers: flags=2
Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
DBG:core:get_hdr_field: cseq <CSeq>: <43> <REGISTER>
Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
DBG:core:parse_via_param: found param type 232, <branch> =
<z9hG4bKda029da0-e604-1910-80bc-002622a67db9>; state=6
Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
DBG:core:parse_via_param: found param type 235, <rport> = <n/a>; state=17
Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:core:parse_via:
end of header reached, state=5
Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
DBG:core:parse_headers: via found, flags=2
Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
DBG:core:parse_headers: this is the first via
Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
DBG:core:receive_msg: After parse_msg...
Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
DBG:core:receive_msg: preparing to run routing scripts...
Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
DBG:core:parse_headers: flags=100
Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:core:parse_to:
end of header reached, state=10
Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:core:parse_to:
display={}, ruri={sip:1000 at 10.20.30.17}
Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
DBG:core:get_hdr_field: <To> [24]; uri=[sip:1000 at 10.20.30.17]
 ]  16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
DBG:core:get_hdr_field: to body [<sip:1000 at 10.20.30.17>
Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
DBG:core:get_hdr_field: content_length=0
Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
DBG:maxfwd:is_maxfwd_present: value = 70
Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:uri:has_totag:
no totag
Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
DBG:core:parse_to_param: tag=70ea9ca0-e604-1910-80bc-002622a67db9
Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:core:parse_to:
end of header reached, state=29
Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:core:parse_to:
display={}, ruri={sip:1000 at 10.20.30.17}
Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
DBG:dispatcher:ds_select_dst: set [1]
Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
DBG:dispatcher:ds_select_dst: alg hash [1], id [1]
Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
DBG:core:parse_headers: flags=ffffffffffffffff
Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
DBG:core:get_hdr_field: found end of header
Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
DBG:core:check_ip_address: params 72.237.213.162, 72.237.213.162, 0
Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
DBG:core:destroy_avp_list: destroying list (nil)
Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
DBG:core:receive_msg: cleaning up
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110816/5bad2962/attachment-0001.html 

From x.liu at hw.ac.uk  Tue Aug 16 22:17:13 2011
From: x.liu at hw.ac.uk (xl127)
Date: Tue, 16 Aug 2011 19:17:13 +0100
Subject: [Freeswitch-users] How to do "search" in freeswitch-users Archives?
In-Reply-To: <CAC-LwPPSAei6EyeQBtrHoin0DtN73f2zfd6GU5JyQm0Z8_rXPw@mail.gmail.com>
References: <CAC-LwPPSAei6EyeQBtrHoin0DtN73f2zfd6GU5JyQm0Z8_rXPw@mail.gmail.com>
Message-ID: <4E4AB429.8000401@hw.ac.uk>

Hi,

how you guys do "search" over all the previous posts in the 
freeswitch-users archives
to find the specific posts you are interested in?

Cheers,
Xing



-- 
Heriot-Watt University is a Scottish charity
registered under charity number SC000278.



From anthony.minessale at gmail.com  Tue Aug 16 22:21:02 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Aug 2011 13:21:02 -0500
Subject: [Freeswitch-users] FS performance using ESL
In-Reply-To: <CAJoKKdX_sPBeWGv2xq7NOX4gd2Q1QQvWpm0wiSLv+67PvaGFVQ@mail.gmail.com>
References: <CAJoKKdU9VPzOS-C3jHCjG6Y0V-Jo_j7hqYwt6-Zh7ohb1pq3xQ@mail.gmail.com>
	<CAKbxfG-SWBzu0dBv_5B0v7710q7vLeeduT9L5V_cnceYHjrGJQ@mail.gmail.com>
	<CAJoKKdUcA=9CvSdRPhaoMopK+f4K2W1DO0=0b5mck_KEjOc9pQ@mail.gmail.com>
	<CAJoKKdXR64qNfCYj0uyYFda4YLavwd_NFn3bH6Td+2eSGo7Y1w@mail.gmail.com>
	<CAKbxfG_SnLS2c6CnepKcy=ea7CY=JnM5Lyy_Rv8ahUYpqJdooA@mail.gmail.com>
	<CAJoKKdX_sPBeWGv2xq7NOX4gd2Q1QQvWpm0wiSLv+67PvaGFVQ@mail.gmail.com>
Message-ID: <CAKbxfG__pZe9yhB7QY2ovTvLg1QgBx3AyTxCP6GRznkinQMpXg@mail.gmail.com>

supplying full or not will not change performance it just controls
weather or not each socket has full control to do other event socket
commands besides ones that relate to the specific channel.

myevents is probably a touch more efficient than filtering on
unique-id but its negligible.

The best way to gain performance is to limit the number of events you
subscribe to, to the bare necessity.



On Tue, Aug 16, 2011 at 7:44 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:
>
>
> On Mon, Aug 15, 2011 at 6:30 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> You must have something setup strangely cos it would definitely reduce
>> your overall cps to use ESL but not down to 2 CPS.
>>
>> Did you look over the server stats like top etc and look for any
>> misconfiguration?
>>
> Hello Anthony, thanks for your response....
>
> yay, i found the cause .... testserver and FS were running on the same
> server. The server had just 1GB of RAM and of course ... by forking
> testserver (on 8 CPS) took all the remaining RAM ending to write into swap
> ... this triggered a domino effect on the entire server becoming less and
> less responsive as testserver started to run from swap!!!... yay .. really
> bad... didn't see it happen until i started nmon ... top/htop didn't make it
> in time to show this issue..
>
> anyhow, i moved testserver to another machine reaching 35 CPS ... really
> nice indeed.
>
>
> now, having a referent point (testserver) im trying to reach that 35 CPS
> with a java application.
>
> yes, i notices few issues :=)
>
> please check: http://pastebin.freeswitch.org/17052
>
>
> ------------------------------------snipp-----------------------------------
>
> Control: full
>
>
> // here i subscribe to all events ... well not good idea but its a start
> events plain all
>
> Content-Type: command/reply
> Reply-Text: +OK event listener enabled plain
>
> //and here i do a filter per uuid
> filter Unique-ID f7a7b97b-df96-41f3-a6a3-fdf24350a45c
>
> Content-Type: command/reply
> Reply-Text: +OK filter added.
> [Unique-ID]=[f7a7b97b-df96-41f3-a6a3-fdf24350a45c]
>
> linger
>
> Content-Type: command/reply
> Reply-Text: +OK will linger
>
>
> // here i send answer in sync mode ( i could change it into async)
> sendmsg
> call-command: execute
> execute-app-name: answer
> event-lock: true
>
> Content-Type: command/reply
> Reply-Text: +OK
>
> Content-Length: 1805
> Content-Type: text/event-plain
>
>
> --------------------------------------------------------------------------------
>
>
>
>
> so my questions:
>
> if i use <action application="socket" data="192.168.254.99:8084 async
> full"/> and if i subscribe to "myevents" i don't need to set a filter on
> uuid and i could gain performance.
> if i use <action application="socket" data="192.168.254.99:8084 async"/>? i
> will be getting events for the call in question only... so no special
> filters needed and i could limit the number of events im subscribing
>
>
> what is a better approach in a matter of performance ?
> What do i loose/gain by using async full vs async mode ?
>
>
>
> Thanks for your answer,
> Tihomir.
>
>
>
>
>
>
>
>
>>
>> On Thu, Aug 11, 2011 at 6:56 PM, Tihomir Culjaga <tculjaga at gmail.com>
>> wrote:
>> > is there any other method than esl to controll calls on FS from an
>> > eternal
>> > application?
>> > will mod_curl or mod_xml_curl get better performance?
>> >
>> > T.
>> >
>> > On Fri, Aug 12, 2011 at 1:33 AM, Tihomir Culjaga <tculjaga at gmail.com>
>> > wrote:
>> >>
>> >> Hi Anthony, thanks for your response ...
>> >>
>> >>
>> >> this is what i have:
>> >>
>> >> ??????? esl_filter(&handle, "unique-id",
>> >> esl_event_get_header(handle.info_event, "caller-unique-id"));
>> >> ??????? esl_events(&handle, ESL_EVENT_TYPE_PLAIN, "CHANNEL_DATA
>> >> CHANNEL_EXECUTE_COMPLETE CHANNEL_HANGUP");
>> >>
>> >> what do you suggest i put there ?
>> >>
>> >>
>> >> is the inbound method less costly ?
>> >>
>> >>
>> >>
>> >>
>> >> I modified testserver.c just a bit...
>> >>
>> >> #include <sys/types.h>? /* include this before any other sys headers */
>> >> #include <sys/wait.h>?? /* header for waitpid() and various macros */
>> >> #include <signal.h>???? /* header for signal functions */
>> >> #include <stdio.h>????? /* header for fprintf() */
>> >> #include <unistd.h>???? /* header for fork() */
>> >> #include <stdlib.h>
>> >> #include <esl.h>
>> >>
>> >> void sig_chld(int);???? /* prototype for our SIGCHLD handler */
>> >>
>> >> static void mycallback(esl_socket_t server_sock, esl_socket_t
>> >> client_sock,
>> >> struct sockaddr_in *addr)
>> >> {
>> >> ??????? esl_handle_t handle = {{0}};
>> >> ??????? int done = 0;
>> >> ??????? esl_status_t status;
>> >> ??????? time_t exp = 0;
>> >>
>> >> ??????? if (fork() != 0) {
>> >> ??????????????? close(client_sock);
>> >> ??????????????? return;
>> >> ??????? }
>> >>
>> >> ??????? esl_attach_handle(&handle, client_sock, addr);
>> >>
>> >> ??????? esl_log(ESL_LOG_INFO, "Connected! %d\n", handle.sock);
>> >>
>> >> ??????? esl_filter(&handle, "unique-id",
>> >> esl_event_get_header(handle.info_event, "caller-unique-id"));
>> >> ??????? esl_events(&handle, ESL_EVENT_TYPE_PLAIN, "CHANNEL_DATA
>> >> CHANNEL_EXECUTE_COMPLETE CHANNEL_HANGUP");
>> >>
>> >> ??????? esl_send_recv(&handle, "linger");
>> >>
>> >> ??????? esl_execute(&handle, "answer", NULL, NULL);
>> >> ??????? //esl_execute(&handle, "conference", "3000 at default", NULL);
>> >> ??????? esl_execute(&handle, "playback",
>> >> "/home/tculjaga/myWavFile.wav",
>> >> NULL);
>> >> ??????? //esl_execute(&handle, "sleep", "1000", NULL);
>> >> ??????? //esl_execute(&handle, "hangup", NULL, NULL);
>> >>
>> >> ??????? while((status = esl_recv_timed(&handle, 1000)) != ESL_FAIL) {
>> >> ??????????????? if (done) {
>> >> ??????????????????????? if (time(NULL) >= exp) {
>> >> ??????????????????????????????? break;
>> >> ??????????????????????? }
>> >> ??????????????? } else if (status == ESL_SUCCESS) {
>> >> ??????????????????????? const char *type =
>> >> esl_event_get_header(handle.last_event, "content-type");
>> >> ??????????????????????? if (type && !strcasecmp(type,
>> >> "text/disconnect-notice")) {
>> >> ??????????????????????????????? const char *dispo =
>> >> esl_event_get_header(handle.last_event, "content-disposition");
>> >> ??????????????????????????????? esl_log(ESL_LOG_INFO, "Got a
>> >> disconnection
>> >> notice dispostion: [%s]\n", dispo ? dispo : "");
>> >> ??????????????????????????????? if (!strcmp(dispo, "linger")) {
>> >> ??????????????????????????????????????? done = 1;
>> >> ??????????????????????????????????????? esl_log(ESL_LOG_INFO, "Waiting
>> >> 5
>> >> seconds for any remaining events.\n");
>> >> ??????????????????????????????????????? exp = time(NULL) + 5;
>> >> ??????????????????????????????? }
>> >> ??????????????????????? }
>> >> ??????????????? }
>> >> ??????? }
>> >>
>> >> ??????? esl_log(ESL_LOG_INFO, "Disconnected! %d\n", handle.sock);
>> >> ??????? esl_disconnect(&handle);
>> >>
>> >> ??????? close(client_sock);
>> >>
>> >> ??????? _exit(0);
>> >> }
>> >>
>> >> /*
>> >> ?* The signal handler function -- only gets called when a SIGCHLD
>> >> ?* is received, ie when a child terminates
>> >> ?*/
>> >> void sig_chld(int signo)
>> >> {
>> >> ??? int status;
>> >>
>> >> ??? /* Wait for any child without blocking */
>> >> ??? if (waitpid(-1, &status, WNOHANG) < 0)
>> >> ??? {
>> >> ??????? /*
>> >> ???????? * calling standard I/O functions like fprintf() in a
>> >> ???????? * signal handler is not recommended, but probably OK
>> >> ???????? * in toy programs like this one.
>> >> ???????? */
>> >> ??????? fprintf(stderr, "waitpid failed\n");
>> >> ??????? return;
>> >> ??? }
>> >> }
>> >>
>> >> int main(void)
>> >> {
>> >> ??????? struct sigaction act;
>> >>
>> >> ??????? /* Assign sig_chld as our SIGCHLD handler */
>> >> ??????? act.sa_handler = sig_chld;
>> >>
>> >> ??????? /* We don't want to block any other signals in this example */
>> >> ??????? sigemptyset(&act.sa_mask);
>> >>
>> >> ??????? /*
>> >> ???????? * We're only interested in children that have terminated, not
>> >> ones
>> >> ???????? * which have been stopped (eg user pressing control-Z at
>> >> terminal)
>> >> ???????? */
>> >> ??????? act.sa_flags = SA_NOCLDSTOP;
>> >>
>> >> ??????? /*
>> >> ???????? * Make these values effective. If we were writing a real
>> >> ???????? * application, we would probably save the old value instead of
>> >> ???????? * passing NULL.
>> >> ???????? */
>> >> /*????? if (sigaction(SIGCHLD, &act, NULL) < 0)
>> >> ??????? {
>> >> ??????????????? fprintf(stderr, "sigaction failed\n");
>> >> ??????????????? return 1;
>> >> ??????? }
>> >> */
>> >> ??????? signal(SIGCHLD, SIG_IGN);
>> >>
>> >> ??????? esl_global_set_default_logger(0);
>> >> ??????? esl_listen("localhost", 8088, mycallback);
>> >>
>> >> ??????? return 0;
>> >> }
>> >>
>> >>
>> >>
>> >>
>> >> On Thu, Aug 11, 2011 at 9:59 PM, Anthony Minessale
>> >> <anthony.minessale at gmail.com> wrote:
>> >>>
>> >>> try removing the filter and event subscriptions
>> >>> it's costly to consume all of the events especially at 75cps.
>> >>>
>> >>>
>> >>> On Thu, Aug 11, 2011 at 5:23 AM, Tihomir Culjaga <tculjaga at gmail.com>
>> >>> wrote:
>> >>> > hello,
>> >>> >
>> >>> > im wondering how much performance do we loose when using ESL instead
>> >>> > of
>> >>> > running it via dialplan?
>> >>> >
>> >>> >
>> >>> > without ESL with a fine tuned FS and a short dialplan ( answer,
>> >>> > playback
>> >>> > like 20 seconds file, hangup ) im able to service 75 CPS. On the
>> >>> > same
>> >>> > FS,
>> >>> > when i use ESL to answer the call, playback the same file and
>> >>> > hangup,
>> >>> > im not
>> >>> > able to run more than 2 CPS... this is a huge impact and i really
>> >>> > can't
>> >>> > believe it.
>> >>> >
>> >>> > I'm using event-socket outbound e.g.:
>> >>> >
>> >>> > <action application="socket" data="127.0.0.1:8088 async full"/>
>> >>> >
>> >>> > my extension looks like:
>> >>> >
>> >>> > <extension name="ESL_C">
>> >>> > ? <condition field="destination_number" expression="^(6666)$">
>> >>> > ??? <action application="socket" data="127.0.0.1:8088 async full"/>
>> >>> > ??? <action application="sleep" data="1000"/>
>> >>> > ??? <action application="hangup"/>
>> >>> > ? </condition>
>> >>> > </extension>
>> >>> >
>> >>> >
>> >>> > im using testserver from lib/esl/ and i just removed the conference
>> >>> > command
>> >>> > and added the playback one.... also i moved the esl_debug lvl to 0
>> >>> >
>> >>> >
>> >>> > anyhow, FS cannot run more than 2 CPS compared to 75 CPS when the
>> >>> > playback
>> >>> > is done from the dialplan.
>> >>> >
>> >>> >
>> >>> > Please, can someone give me a clue on what is going on?
>> >>> > Maybe im doing something wrong?
>> >>> > how to get maximum FS performance using ESL ?
>> >>> >
>> >>> >
>> >>> >
>> >>> > Regards,
>> >>> > Tihomir.
>> >>> >
>> >>> >
>> >>> > _______________________________________________
>> >>> > Join us at ClueCon 2011, Aug 9-11, Chicago
>> >>> > http://www.cluecon.com 877-7-4ACLUE
>> >>> >
>> >>> > FreeSWITCH-users mailing list
>> >>> > FreeSWITCH-users at lists.freeswitch.org
>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >
>> >>> >
>> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> > http://www.freeswitch.org
>> >>> >
>> >>> >
>> >>>
>> >>>
>> >>>
>> >>> --
>> >>> Anthony Minessale II
>> >>>
>> >>> 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
>> >>> googletalk:conf+888 at conference.freeswitch.org
>> >>> pstn:+19193869900
>> >>>
>> >>> _______________________________________________
>> >>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> >>> http://www.cluecon.com 877-7-4ACLUE
>> >>>
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>
>> >
>> >
>> > _______________________________________________
>> > Join us at ClueCon 2011, Aug 9-11, Chicago
>> > http://www.cluecon.com 877-7-4ACLUE
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> 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
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From Hector.Geraldino at ip-soft.net  Tue Aug 16 22:31:57 2011
From: Hector.Geraldino at ip-soft.net (Hector Geraldino)
Date: Tue, 16 Aug 2011 14:31:57 -0400
Subject: [Freeswitch-users] How to do "search" in freeswitch-users
 Archives?
In-Reply-To: <4E4AB429.8000401@hw.ac.uk>
References: <CAC-LwPPSAei6EyeQBtrHoin0DtN73f2zfd6GU5JyQm0Z8_rXPw@mail.gmail.com>
	<4E4AB429.8000401@hw.ac.uk>
Message-ID: <6A6B4C284AD15042B429EB9D904544AD021FD8A87E@NY1-EXMB-01.ip-soft.net>

I use google, constraining the query with: site:lists.freeswitch.org

mailman doesn't provide search functionalities, AFAIK

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of xl127
Sent: Tuesday, August 16, 2011 2:17 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] How to do "search" in freeswitch-users Archives?

Hi,

how you guys do "search" over all the previous posts in the 
freeswitch-users archives
to find the specific posts you are interested in?

Cheers,
Xing



-- 
Heriot-Watt University is a Scottish charity
registered under charity number SC000278.



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


From lakersman2006 at yahoo.com  Tue Aug 16 22:48:11 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Tue, 16 Aug 2011 11:48:11 -0700 (PDT)
Subject: [Freeswitch-users] ORIGINATE_DISPOSITION
In-Reply-To: <CAKbxfG9N1TUmT=HZQqhJfkLk_YiKc=XPqbBoa-9Rm=+Q97PZJw@mail.gmail.com>
References: <1313447714.89555.YahooMailNeo@web161001.mail.bf1.yahoo.com>
	<CAKzWOxVwDaxsFJAr1dVP5nfmDuFXK2tptvai+oqtwZgph3u1kw@mail.gmail.com>
	<1313452154.28177.YahooMailNeo@web161001.mail.bf1.yahoo.com>
	<CAKzWOxWYXy5Sa2g+cCDEDcxWwE10OtrMaJzUyoLdFSDGtDfbyg@mail.gmail.com>
	<1313452622.11918.YahooMailNeo@web161012.mail.bf1.yahoo.com>
	<CAKzWOxVREFK=uzYzT95oXFT0-sURz7yeOwkuQMz6LKjQ6GvoXA@mail.gmail.com>
	<1313453126.47357.YahooMailNeo@web161010.mail.bf1.yahoo.com>
	<CAKzWOxWtxNoto1c6Chu5dZLPkt+ATcwCGpHOYtPpWydUrcCFfQ@mail.gmail.com>
	<CA+GmxbZ2RG+RS79F8yjXAR4eeFwFy3EMV0k4YiESwq3qMCbwkA@mail.gmail.com>
	<1313515585.77893.YahooMailNeo@web161019.mail.bf1.yahoo.com>
	<CAKbxfG9N1TUmT=HZQqhJfkLk_YiKc=XPqbBoa-9Rm=+Q97PZJw@mail.gmail.com>
Message-ID: <1313520491.44910.YahooMailNeo@web161015.mail.bf1.yahoo.com>

"if you call somewhere that uses early media, the originate will end
setting originate_disposition to SUCCESS meaning that a live channel
was produced.? Now the bridge will begin between the A and B leg."

Does that mean it is safe to assume when "originate_disposition" is SUCCESS, that the call has been answered if I set the following:


{ignore_early_media=false}
{bridge_early_media=true}




________________________________
From: Anthony Minessale <anthony.minessale at gmail.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, August 16, 2011 10:54 AM
Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION

The originate disposition of A leg will always have the last known status from B

originate, by default, returns when:

1) media is established on a specific outgoing leg, answered or not
2) all outgoing legs are terminated.

if you call somewhere that uses early media, the originate will end
setting originate_disposition to SUCCESS meaning that a live channel
was produced.? Now the bridge will begin between the A and B leg.

When the bridge ends if the B leg is hungup, its cause will be stored
in "bridge_hangup_cause"

if you want to keep the originate from ending when early media is
established you can add {ignore_early_media=true} or
{bridge_early_media=true} prepended to your dial string and the
originate will never return untill all outbound legs are either hungup
or one is answered.


You should take some time to expand your mind to the different
paradigm in FreeSWITCH where you may have as many as 10 outbound legs
at once in a forked-dial situation and some of what you think is
simple and obvious will quickly dissolve.

Another thing you can do is set the variable "failed_xml_cdr_prefix"
on the A leg.
This prefix will be mixed with an incrementing variable for each
outbound call leg and in the case of a failure the entire XML cdr will
be set into a var on A leg.

for instance if you set failed_xml_cdr_prefix=foo you would get foo_1
foo_2 etc depending on the number of outbound call legs.

Additionally you can set copy_xml_cdr on the A leg and when the bridge
ends you will get a complete CDR for B in the "b_leg_cdr" variable on
A

Finally you should really go with the flow of how FreeSWITCH is
engineered and try to keep your accounting logic in a separate place
and monitor the XML-CDR, CDR-CSV or event_socket +
CHANNEL_HANGUP_COMPLETE events to process this information.? There is
much more to a call and what happens when it's transferred etc than
what you can get in a single monolithic perspective of inside the
channel.? Plus it's conter intuitive to put routing, application and
accounting logic in the same place.



On Tue, Aug 16, 2011 at 12:26 PM, Sam <lakersman2006 at yahoo.com> wrote:
> Dmitry,
> Yes, I agree. For an novice like me it was extremely confusing and
> frustrating that I could not get the proper B leg disposition when A leg is
> answered by FS itself.
> I hope there can be some methods or functions that will allow us to get the
> proper B Leg disposition.
> ________________________________
> From: Dmitry Sytchev <kbdfck at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Sent: Tuesday, August 16, 2011 2:30 AM
> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>
> Seems we need to clear things about B-leg disposition in wiki.
> AFAIK there is no method to get correct disposition from B leg without
> analyzing events in case when A-leg was answered by FS itself.
> If A leg was not answered, we can use A-leg disposition for call
> disposition.
>
> 2011/8/16 Michael Collins <msc at freeswitch.org>
>
>
> On Mon, Aug 15, 2011 at 5:05 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> I am using perl's $session->get_variable("originate_disposition");
>
> Are you looking at the b-leg's session?
>
>
> Also, how come the "hangup_time" shows zero on answered calls?
>
> Because hangup_time refers to the point in time at which the call was hung
> up. Since you are in the middle of a call (using the $session object) you
> will never see the hangup_time because the object ceases to exist once the
> call leg is disconnected.
> I get the impression that you may be using the wrong tool for this
> particular job, but I'm not sure without seeing it. If you don't mind
> dropping it on pastebin we'll have a look and give you some suggestions.
> -MC
>
> ________________________________
> From: Michael Collins <msc at freeswitch.org>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Sent: Monday, August 15, 2011 4:58 PM
> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>
> And how are you checking the variable? Do you have an event socket open or
> ... ?
> -MC
>
> On Mon, Aug 15, 2011 at 4:57 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> It is being generated with a bridge.
>
> ________________________________
> From: Michael Collins <msc at freeswitch.org>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Sent: Monday, August 15, 2011 4:54 PM
> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>
> And how is it being generated? WIth a bridge or originate or ... ?
> -MC
>
> On Mon, Aug 15, 2011 at 4:49 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> I want to know the B-leg status of the call.
>
> ________________________________
> From: Michael Collins <msc at freeswitch.org>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Sent: Monday, August 15, 2011 4:31 PM
> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>
> I think you may be wanting "endpoint_disposition" depending on exactly what
> you're looking at.
> -MC
>
> On Mon, Aug 15, 2011 at 3:35 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> For the ORIGINATE_DISPOSITION channel variable, does "SUCCESS" mean the call
> was ANSWERED?
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Best regards,
>
> Dmitry Sytchev,
> IT Engineer
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110816/84f6fc48/attachment-0001.html 

From anthony.minessale at gmail.com  Tue Aug 16 22:56:55 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Aug 2011 13:56:55 -0500
Subject: [Freeswitch-users] ORIGINATE_DISPOSITION
In-Reply-To: <1313520491.44910.YahooMailNeo@web161015.mail.bf1.yahoo.com>
References: <1313447714.89555.YahooMailNeo@web161001.mail.bf1.yahoo.com>
	<CAKzWOxVwDaxsFJAr1dVP5nfmDuFXK2tptvai+oqtwZgph3u1kw@mail.gmail.com>
	<1313452154.28177.YahooMailNeo@web161001.mail.bf1.yahoo.com>
	<CAKzWOxWYXy5Sa2g+cCDEDcxWwE10OtrMaJzUyoLdFSDGtDfbyg@mail.gmail.com>
	<1313452622.11918.YahooMailNeo@web161012.mail.bf1.yahoo.com>
	<CAKzWOxVREFK=uzYzT95oXFT0-sURz7yeOwkuQMz6LKjQ6GvoXA@mail.gmail.com>
	<1313453126.47357.YahooMailNeo@web161010.mail.bf1.yahoo.com>
	<CAKzWOxWtxNoto1c6Chu5dZLPkt+ATcwCGpHOYtPpWydUrcCFfQ@mail.gmail.com>
	<CA+GmxbZ2RG+RS79F8yjXAR4eeFwFy3EMV0k4YiESwq3qMCbwkA@mail.gmail.com>
	<1313515585.77893.YahooMailNeo@web161019.mail.bf1.yahoo.com>
	<CAKbxfG9N1TUmT=HZQqhJfkLk_YiKc=XPqbBoa-9Rm=+Q97PZJw@mail.gmail.com>
	<1313520491.44910.YahooMailNeo@web161015.mail.bf1.yahoo.com>
Message-ID: <CAKbxfG-3UDi59h3VcOm7+3+OXgaaX1fjNRLZiePrjVc6Ba10Cw@mail.gmail.com>

bridge_early_media implies ignore_early_media true so you can't use
them together but yes since there is no cause code for answered
because the call has not ended that would guarantee it was answered if
you exited originate with either ignore_early_media=true or
bridge_early_media=true

On Tue, Aug 16, 2011 at 1:48 PM, Sam <lakersman2006 at yahoo.com> wrote:
> "if you call somewhere that uses early media, the originate will end
> setting originate_disposition to SUCCESS meaning that a live channel
> was produced.? Now the bridge will begin between the A and B leg."
> Does that mean it is safe to assume when "originate_disposition" is SUCCESS,
> that the call has been answered if I set the following:
>
> {ignore_early_media=false}
> {bridge_early_media=true}
>
>
> ________________________________
> From: Anthony Minessale <anthony.minessale at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Sent: Tuesday, August 16, 2011 10:54 AM
> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>
> The originate disposition of A leg will always have the last known status
> from B
>
> originate, by default, returns when:
>
> 1) media is established on a specific outgoing leg, answered or not
> 2) all outgoing legs are terminated.
>
> if you call somewhere that uses early media, the originate will end
> setting originate_disposition to SUCCESS meaning that a live channel
> was produced.? Now the bridge will begin between the A and B leg.
>
> When the bridge ends if the B leg is hungup, its cause will be stored
> in "bridge_hangup_cause"
>
> if you want to keep the originate from ending when early media is
> established you can add {ignore_early_media=true} or
> {bridge_early_media=true} prepended to your dial string and the
> originate will never return untill all outbound legs are either hungup
> or one is answered.
>
>
> You should take some time to expand your mind to the different
> paradigm in FreeSWITCH where you may have as many as 10 outbound legs
> at once in a forked-dial situation and some of what you think is
> simple and obvious will quickly dissolve.
>
> Another thing you can do is set the variable "failed_xml_cdr_prefix"
> on the A leg.
> This prefix will be mixed with an incrementing variable for each
> outbound call leg and in the case of a failure the entire XML cdr will
> be set into a var on A leg.
>
> for instance if you set failed_xml_cdr_prefix=foo you would get foo_1
> foo_2 etc depending on the number of outbound call legs.
>
> Additionally you can set copy_xml_cdr on the A leg and when the bridge
> ends you will get a complete CDR for B in the "b_leg_cdr" variable on
> A
>
> Finally you should really go with the flow of how FreeSWITCH is
> engineered and try to keep your accounting logic in a separate place
> and monitor the XML-CDR, CDR-CSV or event_socket +
> CHANNEL_HANGUP_COMPLETE events to process this information.? There is
> much more to a call and what happens when it's transferred etc than
> what you can get in a single monolithic perspective of inside the
> channel.? Plus it's conter intuitive to put routing, application and
> accounting logic in the same place.
>
>
>
> On Tue, Aug 16, 2011 at 12:26 PM, Sam <lakersman2006 at yahoo.com> wrote:
>> Dmitry,
>> Yes, I agree. For an novice like me it was extremely confusing and
>> frustrating that I could not get the proper B leg disposition when A leg
>> is
>> answered by FS itself.
>> I hope there can be some methods or functions that will allow us to get
>> the
>> proper B Leg disposition.
>> ________________________________
>> From: Dmitry Sytchev <kbdfck at gmail.com>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Sent: Tuesday, August 16, 2011 2:30 AM
>> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>
>> Seems we need to clear things about B-leg disposition in wiki.
>> AFAIK there is no method to get correct disposition from B leg without
>> analyzing events in case when A-leg was answered by FS itself.
>> If A leg was not answered, we can use A-leg disposition for call
>> disposition.
>>
>> 2011/8/16 Michael Collins <msc at freeswitch.org>
>>
>>
>> On Mon, Aug 15, 2011 at 5:05 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>
>> I am using perl's $session->get_variable("originate_disposition");
>>
>> Are you looking at the b-leg's session?
>>
>>
>> Also, how come the "hangup_time" shows zero on answered calls?
>>
>> Because hangup_time refers to the point in time at which the call was hung
>> up. Since you are in the middle of a call (using the $session object) you
>> will never see the hangup_time because the object ceases to exist once the
>> call leg is disconnected.
>> I get the impression that you may be using the wrong tool for this
>> particular job, but I'm not sure without seeing it. If you don't mind
>> dropping it on pastebin we'll have a look and give you some suggestions.
>> -MC
>>
>> ________________________________
>> From: Michael Collins <msc at freeswitch.org>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Sent: Monday, August 15, 2011 4:58 PM
>> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>
>> And how are you checking the variable? Do you have an event socket open or
>> ... ?
>> -MC
>>
>> On Mon, Aug 15, 2011 at 4:57 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>
>> It is being generated with a bridge.
>>
>> ________________________________
>> From: Michael Collins <msc at freeswitch.org>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Sent: Monday, August 15, 2011 4:54 PM
>> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>
>> And how is it being generated? WIth a bridge or originate or ... ?
>> -MC
>>
>> On Mon, Aug 15, 2011 at 4:49 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>
>> I want to know the B-leg status of the call.
>>
>> ________________________________
>> From: Michael Collins <msc at freeswitch.org>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Sent: Monday, August 15, 2011 4:31 PM
>> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>
>> I think you may be wanting "endpoint_disposition" depending on exactly
>> what
>> you're looking at.
>> -MC
>>
>> On Mon, Aug 15, 2011 at 3:35 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>
>> For the ORIGINATE_DISPOSITION channel variable, does "SUCCESS" mean the
>> call
>> was ANSWERED?
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> --
>> Best regards,
>>
>> Dmitry Sytchev,
>> IT Engineer
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From lakersman2006 at yahoo.com  Tue Aug 16 23:18:55 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Tue, 16 Aug 2011 12:18:55 -0700 (PDT)
Subject: [Freeswitch-users] ORIGINATE_DISPOSITION
In-Reply-To: <CAKbxfG-3UDi59h3VcOm7+3+OXgaaX1fjNRLZiePrjVc6Ba10Cw@mail.gmail.com>
References: <1313447714.89555.YahooMailNeo@web161001.mail.bf1.yahoo.com>
	<CAKzWOxVwDaxsFJAr1dVP5nfmDuFXK2tptvai+oqtwZgph3u1kw@mail.gmail.com>
	<1313452154.28177.YahooMailNeo@web161001.mail.bf1.yahoo.com>
	<CAKzWOxWYXy5Sa2g+cCDEDcxWwE10OtrMaJzUyoLdFSDGtDfbyg@mail.gmail.com>
	<1313452622.11918.YahooMailNeo@web161012.mail.bf1.yahoo.com>
	<CAKzWOxVREFK=uzYzT95oXFT0-sURz7yeOwkuQMz6LKjQ6GvoXA@mail.gmail.com>
	<1313453126.47357.YahooMailNeo@web161010.mail.bf1.yahoo.com>
	<CAKzWOxWtxNoto1c6Chu5dZLPkt+ATcwCGpHOYtPpWydUrcCFfQ@mail.gmail.com>
	<CA+GmxbZ2RG+RS79F8yjXAR4eeFwFy3EMV0k4YiESwq3qMCbwkA@mail.gmail.com>
	<1313515585.77893.YahooMailNeo@web161019.mail.bf1.yahoo.com>
	<CAKbxfG9N1TUmT=HZQqhJfkLk_YiKc=XPqbBoa-9Rm=+Q97PZJw@mail.gmail.com>
	<1313520491.44910.YahooMailNeo@web161015.mail.bf1.yahoo.com>
	<CAKbxfG-3UDi59h3VcOm7+3+OXgaaX1fjNRLZiePrjVc6Ba10Cw@mail.gmail.com>
Message-ID: <1313522335.98043.YahooMailNeo@web161008.mail.bf1.yahoo.com>

I need to be able to play back early media to A leg, so if {ignore_early_media=false} there is no way to tell from originate_disposition that the bridge was answered?



________________________________
From: Anthony Minessale <anthony.minessale at gmail.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, August 16, 2011 11:56 AM
Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION

bridge_early_media implies ignore_early_media true so you can't use
them together but yes since there is no cause code for answered
because the call has not ended that would guarantee it was answered if
you exited originate with either ignore_early_media=true or
bridge_early_media=true

On Tue, Aug 16, 2011 at 1:48 PM, Sam <lakersman2006 at yahoo.com> wrote:
> "if you call somewhere that uses early media, the originate will end
> setting originate_disposition to SUCCESS meaning that a live channel
> was produced.? Now the bridge will begin between the A and B leg."
> Does that mean it is safe to assume when "originate_disposition" is SUCCESS,
> that the call has been answered if I set the following:
>
> {ignore_early_media=false}
> {bridge_early_media=true}
>
>
> ________________________________
> From: Anthony Minessale <anthony.minessale at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Sent: Tuesday, August 16, 2011 10:54 AM
> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>
> The originate disposition of A leg will always have the last known status
> from B
>
> originate, by default, returns when:
>
> 1) media is established on a specific outgoing leg, answered or not
> 2) all outgoing legs are terminated.
>
> if you call somewhere that uses early media, the originate will end
> setting originate_disposition to SUCCESS meaning that a live channel
> was produced.? Now the bridge will begin between the A and B leg.
>
> When the bridge ends if the B leg is hungup, its cause will be stored
> in "bridge_hangup_cause"
>
> if you want to keep the originate from ending when early media is
> established you can add {ignore_early_media=true} or
> {bridge_early_media=true} prepended to your dial string and the
> originate will never return untill all outbound legs are either hungup
> or one is answered.
>
>
> You should take some time to expand your mind to the different
> paradigm in FreeSWITCH where you may have as many as 10 outbound legs
> at once in a forked-dial situation and some of what you think is
> simple and obvious will quickly dissolve.
>
> Another thing you can do is set the variable "failed_xml_cdr_prefix"
> on the A leg.
> This prefix will be mixed with an incrementing variable for each
> outbound call leg and in the case of a failure the entire XML cdr will
> be set into a var on A leg.
>
> for instance if you set failed_xml_cdr_prefix=foo you would get foo_1
> foo_2 etc depending on the number of outbound call legs.
>
> Additionally you can set copy_xml_cdr on the A leg and when the bridge
> ends you will get a complete CDR for B in the "b_leg_cdr" variable on
> A
>
> Finally you should really go with the flow of how FreeSWITCH is
> engineered and try to keep your accounting logic in a separate place
> and monitor the XML-CDR, CDR-CSV or event_socket +
> CHANNEL_HANGUP_COMPLETE events to process this information.? There is
> much more to a call and what happens when it's transferred etc than
> what you can get in a single monolithic perspective of inside the
> channel.? Plus it's conter intuitive to put routing, application and
> accounting logic in the same place.
>
>
>
> On Tue, Aug 16, 2011 at 12:26 PM, Sam <lakersman2006 at yahoo.com> wrote:
>> Dmitry,
>> Yes, I agree. For an novice like me it was extremely confusing and
>> frustrating that I could not get the proper B leg disposition when A leg
>> is
>> answered by FS itself.
>> I hope there can be some methods or functions that will allow us to get
>> the
>> proper B Leg disposition.
>> ________________________________
>> From: Dmitry Sytchev <kbdfck at gmail.com>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Sent: Tuesday, August 16, 2011 2:30 AM
>> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>
>> Seems we need to clear things about B-leg disposition in wiki.
>> AFAIK there is no method to get correct disposition from B leg without
>> analyzing events in case when A-leg was answered by FS itself.
>> If A leg was not answered, we can use A-leg disposition for call
>> disposition.
>>
>> 2011/8/16 Michael Collins <msc at freeswitch.org>
>>
>>
>> On Mon, Aug 15, 2011 at 5:05 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>
>> I am using perl's $session->get_variable("originate_disposition");
>>
>> Are you looking at the b-leg's session?
>>
>>
>> Also, how come the "hangup_time" shows zero on answered calls?
>>
>> Because hangup_time refers to the point in time at which the call was hung
>> up. Since you are in the middle of a call (using the $session object) you
>> will never see the hangup_time because the object ceases to exist once the
>> call leg is disconnected.
>> I get the impression that you may be using the wrong tool for this
>> particular job, but I'm not sure without seeing it. If you don't mind
>> dropping it on pastebin we'll have a look and give you some suggestions.
>> -MC
>>
>> ________________________________
>> From: Michael Collins <msc at freeswitch.org>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Sent: Monday, August 15, 2011 4:58 PM
>> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>
>> And how are you checking the variable? Do you have an event socket open or
>> ... ?
>> -MC
>>
>> On Mon, Aug 15, 2011 at 4:57 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>
>> It is being generated with a bridge.
>>
>> ________________________________
>> From: Michael Collins <msc at freeswitch.org>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Sent: Monday, August 15, 2011 4:54 PM
>> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>
>> And how is it being generated? WIth a bridge or originate or ... ?
>> -MC
>>
>> On Mon, Aug 15, 2011 at 4:49 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>
>> I want to know the B-leg status of the call.
>>
>> ________________________________
>> From: Michael Collins <msc at freeswitch.org>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Sent: Monday, August 15, 2011 4:31 PM
>> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>
>> I think you may be wanting "endpoint_disposition" depending on exactly
>> what
>> you're looking at.
>> -MC
>>
>> On Mon, Aug 15, 2011 at 3:35 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>
>> For the ORIGINATE_DISPOSITION channel variable, does "SUCCESS" mean the
>> call
>> was ANSWERED?
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> --
>> Best regards,
>>
>> Dmitry Sytchev,
>> IT Engineer
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110816/8377250a/attachment-0001.html 

From vipkilla at gmail.com  Tue Aug 16 23:22:12 2011
From: vipkilla at gmail.com (vip killa)
Date: Tue, 16 Aug 2011 15:22:12 -0400
Subject: [Freeswitch-users] enterprise deployment not working
In-Reply-To: <CAC-LwPPSAei6EyeQBtrHoin0DtN73f2zfd6GU5JyQm0Z8_rXPw@mail.gmail.com>
References: <CAC-LwPPSAei6EyeQBtrHoin0DtN73f2zfd6GU5JyQm0Z8_rXPw@mail.gmail.com>
Message-ID: <CAC-LwPP+oNWNu-5=-Y2KQwGkn0qaaf7Bq59CcXTo_Ura+AZwBA@mail.gmail.com>

I did an ngrep trace and opensips is returning:

SIP/2.0 503 Service Unavailable
CSeq: 54 REGISTER

before it even tries the FS server...
perhaps this wiki page needs to be updated...


On Tue, Aug 16, 2011 at 2:03 PM, vip killa <vipkilla at gmail.com> wrote:

> I'm following tutorial at
> http://wiki.freeswitch.org/wiki/Enterprise_deployment_<http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS>
> OpenSIPS <http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS> and
> opensips is not forwarding the registration requests...
> Running opensips-1.6.4-2-tls,
> Here is what the debugging looks like when i try to register with the
> opensip box
> 10.20.30.17 is the opensips box.
> 10.20.30.18 is the FreeSWITCH box.
>
>
> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
> DBG:core:parse_msg: SIP Request:
> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
> DBG:core:parse_msg:  method:  <REGISTER>
> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
> DBG:core:parse_msg:  uri:     <sip:10.20.30.17>
> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
> DBG:core:parse_msg:  version: <SIP/2.0>
> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
> DBG:core:parse_headers: flags=2
> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
> DBG:core:get_hdr_field: cseq <CSeq>: <43> <REGISTER>
> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
> DBG:core:parse_via_param: found param type 232, <branch> =
> <z9hG4bKda029da0-e604-1910-80bc-002622a67db9>; state=6
> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
> DBG:core:parse_via_param: found param type 235, <rport> = <n/a>; state=17
> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
> DBG:core:parse_via: end of header reached, state=5
> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
> DBG:core:parse_headers: via found, flags=2
> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
> DBG:core:parse_headers: this is the first via
> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
> DBG:core:receive_msg: After parse_msg...
> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
> DBG:core:receive_msg: preparing to run routing scripts...
> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
> DBG:core:parse_headers: flags=100
> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:core:parse_to:
> end of header reached, state=10
> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:core:parse_to:
> display={}, ruri={sip:1000 at 10.20.30.17}
> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
> DBG:core:get_hdr_field: <To> [24]; uri=[sip:1000 at 10.20.30.17]
>  ]  16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
> DBG:core:get_hdr_field: to body [<sip:1000 at 10.20.30.17>
> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
> DBG:core:get_hdr_field: content_length=0
> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
> DBG:maxfwd:is_maxfwd_present: value = 70
> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:uri:has_totag:
> no totag
> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
> DBG:core:parse_to_param: tag=70ea9ca0-e604-1910-80bc-002622a67db9
> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:core:parse_to:
> end of header reached, state=29
> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:core:parse_to:
> display={}, ruri={sip:1000 at 10.20.30.17}
> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
> DBG:dispatcher:ds_select_dst: set [1]
> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
> DBG:dispatcher:ds_select_dst: alg hash [1], id [1]
> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
> DBG:core:parse_headers: flags=ffffffffffffffff
> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
> DBG:core:get_hdr_field: found end of header
> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
> DBG:core:check_ip_address: params 72.237.213.162, 72.237.213.162, 0
> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
> DBG:core:destroy_avp_list: destroying list (nil)
> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
> DBG:core:receive_msg: cleaning up
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110816/f63055ff/attachment.html 

From anthony.minessale at gmail.com  Tue Aug 16 23:24:00 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Aug 2011 14:24:00 -0500
Subject: [Freeswitch-users] ORIGINATE_DISPOSITION
In-Reply-To: <1313522335.98043.YahooMailNeo@web161008.mail.bf1.yahoo.com>
References: <1313447714.89555.YahooMailNeo@web161001.mail.bf1.yahoo.com>
	<CAKzWOxVwDaxsFJAr1dVP5nfmDuFXK2tptvai+oqtwZgph3u1kw@mail.gmail.com>
	<1313452154.28177.YahooMailNeo@web161001.mail.bf1.yahoo.com>
	<CAKzWOxWYXy5Sa2g+cCDEDcxWwE10OtrMaJzUyoLdFSDGtDfbyg@mail.gmail.com>
	<1313452622.11918.YahooMailNeo@web161012.mail.bf1.yahoo.com>
	<CAKzWOxVREFK=uzYzT95oXFT0-sURz7yeOwkuQMz6LKjQ6GvoXA@mail.gmail.com>
	<1313453126.47357.YahooMailNeo@web161010.mail.bf1.yahoo.com>
	<CAKzWOxWtxNoto1c6Chu5dZLPkt+ATcwCGpHOYtPpWydUrcCFfQ@mail.gmail.com>
	<CA+GmxbZ2RG+RS79F8yjXAR4eeFwFy3EMV0k4YiESwq3qMCbwkA@mail.gmail.com>
	<1313515585.77893.YahooMailNeo@web161019.mail.bf1.yahoo.com>
	<CAKbxfG9N1TUmT=HZQqhJfkLk_YiKc=XPqbBoa-9Rm=+Q97PZJw@mail.gmail.com>
	<1313520491.44910.YahooMailNeo@web161015.mail.bf1.yahoo.com>
	<CAKbxfG-3UDi59h3VcOm7+3+OXgaaX1fjNRLZiePrjVc6Ba10Cw@mail.gmail.com>
	<1313522335.98043.YahooMailNeo@web161008.mail.bf1.yahoo.com>
Message-ID: <CAKbxfG92amS7=q4ex-s7xUJpxHu3QPW6cfdGZ29kgOA0hDU3HA@mail.gmail.com>

that is what bridge_early_media=true means, so you can hear the early
media during originate but stil not return until the channel is
answered or hungup.

On Tue, Aug 16, 2011 at 2:18 PM, Sam <lakersman2006 at yahoo.com> wrote:
> I need to be able to play back early media to A leg, so if
> {ignore_early_media=false} there is no way to tell from
> originate_disposition that the bridge was answered?
>
> ________________________________
> From: Anthony Minessale <anthony.minessale at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Sent: Tuesday, August 16, 2011 11:56 AM
> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>
> bridge_early_media implies ignore_early_media true so you can't use
> them together but yes since there is no cause code for answered
> because the call has not ended that would guarantee it was answered if
> you exited originate with either ignore_early_media=true or
> bridge_early_media=true
>
> On Tue, Aug 16, 2011 at 1:48 PM, Sam <lakersman2006 at yahoo.com> wrote:
>> "if you call somewhere that uses early media, the originate will end
>> setting originate_disposition to SUCCESS meaning that a live channel
>> was produced.? Now the bridge will begin between the A and B leg."
>> Does that mean it is safe to assume when "originate_disposition" is
>> SUCCESS,
>> that the call has been answered if I set the following:
>>
>> {ignore_early_media=false}
>> {bridge_early_media=true}
>>
>>
>> ________________________________
>> From: Anthony Minessale <anthony.minessale at gmail.com>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Sent: Tuesday, August 16, 2011 10:54 AM
>> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>
>> The originate disposition of A leg will always have the last known status
>> from B
>>
>> originate, by default, returns when:
>>
>> 1) media is established on a specific outgoing leg, answered or not
>> 2) all outgoing legs are terminated.
>>
>> if you call somewhere that uses early media, the originate will end
>> setting originate_disposition to SUCCESS meaning that a live channel
>> was produced.? Now the bridge will begin between the A and B leg.
>>
>> When the bridge ends if the B leg is hungup, its cause will be stored
>> in "bridge_hangup_cause"
>>
>> if you want to keep the originate from ending when early media is
>> established you can add {ignore_early_media=true} or
>> {bridge_early_media=true} prepended to your dial string and the
>> originate will never return untill all outbound legs are either hungup
>> or one is answered.
>>
>>
>> You should take some time to expand your mind to the different
>> paradigm in FreeSWITCH where you may have as many as 10 outbound legs
>> at once in a forked-dial situation and some of what you think is
>> simple and obvious will quickly dissolve.
>>
>> Another thing you can do is set the variable "failed_xml_cdr_prefix"
>> on the A leg.
>> This prefix will be mixed with an incrementing variable for each
>> outbound call leg and in the case of a failure the entire XML cdr will
>> be set into a var on A leg.
>>
>> for instance if you set failed_xml_cdr_prefix=foo you would get foo_1
>> foo_2 etc depending on the number of outbound call legs.
>>
>> Additionally you can set copy_xml_cdr on the A leg and when the bridge
>> ends you will get a complete CDR for B in the "b_leg_cdr" variable on
>> A
>>
>> Finally you should really go with the flow of how FreeSWITCH is
>> engineered and try to keep your accounting logic in a separate place
>> and monitor the XML-CDR, CDR-CSV or event_socket +
>> CHANNEL_HANGUP_COMPLETE events to process this information.? There is
>> much more to a call and what happens when it's transferred etc than
>> what you can get in a single monolithic perspective of inside the
>> channel.? Plus it's conter intuitive to put routing, application and
>> accounting logic in the same place.
>>
>>
>>
>> On Tue, Aug 16, 2011 at 12:26 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>> Dmitry,
>>> Yes, I agree. For an novice like me it was extremely confusing and
>>> frustrating that I could not get the proper B leg disposition when A leg
>>> is
>>> answered by FS itself.
>>> I hope there can be some methods or functions that will allow us to get
>>> the
>>> proper B Leg disposition.
>>> ________________________________
>>> From: Dmitry Sytchev <kbdfck at gmail.com>
>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> Sent: Tuesday, August 16, 2011 2:30 AM
>>> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>>
>>> Seems we need to clear things about B-leg disposition in wiki.
>>> AFAIK there is no method to get correct disposition from B leg without
>>> analyzing events in case when A-leg was answered by FS itself.
>>> If A leg was not answered, we can use A-leg disposition for call
>>> disposition.
>>>
>>> 2011/8/16 Michael Collins <msc at freeswitch.org>
>>>
>>>
>>> On Mon, Aug 15, 2011 at 5:05 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>>
>>> I am using perl's $session->get_variable("originate_disposition");
>>>
>>> Are you looking at the b-leg's session?
>>>
>>>
>>> Also, how come the "hangup_time" shows zero on answered calls?
>>>
>>> Because hangup_time refers to the point in time at which the call was
>>> hung
>>> up. Since you are in the middle of a call (using the $session object) you
>>> will never see the hangup_time because the object ceases to exist once
>>> the
>>> call leg is disconnected.
>>> I get the impression that you may be using the wrong tool for this
>>> particular job, but I'm not sure without seeing it. If you don't mind
>>> dropping it on pastebin we'll have a look and give you some suggestions.
>>> -MC
>>>
>>> ________________________________
>>> From: Michael Collins <msc at freeswitch.org>
>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> Sent: Monday, August 15, 2011 4:58 PM
>>> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>>
>>> And how are you checking the variable? Do you have an event socket open
>>> or
>>> ... ?
>>> -MC
>>>
>>> On Mon, Aug 15, 2011 at 4:57 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>>
>>> It is being generated with a bridge.
>>>
>>> ________________________________
>>> From: Michael Collins <msc at freeswitch.org>
>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> Sent: Monday, August 15, 2011 4:54 PM
>>> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>>
>>> And how is it being generated? WIth a bridge or originate or ... ?
>>> -MC
>>>
>>> On Mon, Aug 15, 2011 at 4:49 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>>
>>> I want to know the B-leg status of the call.
>>>
>>> ________________________________
>>> From: Michael Collins <msc at freeswitch.org>
>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> Sent: Monday, August 15, 2011 4:31 PM
>>> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>>
>>> I think you may be wanting "endpoint_disposition" depending on exactly
>>> what
>>> you're looking at.
>>> -MC
>>>
>>> On Mon, Aug 15, 2011 at 3:35 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>>
>>> For the ORIGINATE_DISPOSITION channel variable, does "SUCCESS" mean the
>>> call
>>> was ANSWERED?
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> --
>>> Best regards,
>>>
>>> Dmitry Sytchev,
>>> IT Engineer
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> 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
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From marcdecorny at gmail.com  Tue Aug 16 23:30:23 2011
From: marcdecorny at gmail.com (Marc de Corny)
Date: Tue, 16 Aug 2011 20:30:23 +0100
Subject: [Freeswitch-users] Notify message to Gateway
Message-ID: <06E3326A-4909-4B64-8F24-E441211F6F71@gmail.com>

Hi all,

I have seen a lot of information on how to send NOTIFY messages to  
registered endpoints for ip phone resync.

I'm trying to send the same notify messages over a sip trunk to  
another platform where the endpoints are registered.

By putting the ip address of the remote server on the sip trunk in the  
host field of the send event FS did not send any messages.

Am i missing anything? Do  have to set the profile to external?

Any infornation is helpful.
Many thanks
Marc




From cesar.bermudez at gmail.com  Tue Aug 16 23:38:59 2011
From: cesar.bermudez at gmail.com (Cesar Bermudez)
Date: Tue, 16 Aug 2011 13:38:59 -0600
Subject: [Freeswitch-users] Billing solutions
Message-ID: <CADrcvV9wk2aQJA2GNMNDf4RN98G0Pd7Gjkt6-y_S3d10Nz5k9A@mail.gmail.com>

Hi FS users.

I want to know, what billing solutions are available for FS.
Open Source and not open source.
Features, etc.

Best regards
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110816/48d7235a/attachment-0001.html 

From avi at avimarcus.net  Tue Aug 16 23:41:13 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 16 Aug 2011 22:41:13 +0300
Subject: [Freeswitch-users] Goip GSM Gateway works great with FreeSwitch!
In-Reply-To: <1313496506598-6691087.post@n2.nabble.com>
References: <1313496506598-6691087.post@n2.nabble.com>
Message-ID: <CALNZuwnvSNaTD4LVwikb71YW4t2aa6=XWYyXaE17oWimrZ4_eQ@mail.gmail.com>

If you can find an appropriate place on the wiki for this, that would be a
better place to archive it for future usage.
-Avi


On Tue, Aug 16, 2011 at 3:08 PM, Dissident <jcgpoza at gmail.com> wrote:

>
> http://freeswitch-users.2379917.n2.nabble.com/file/n6691087/single-channel-gsm-gateway-goip-gateway.jpg
>
> Hi, I've configured this GSM gateway with Freeswitch and it works great.
> I'm
> posting this so that if somebody tries to configure this Gateway won't have
> to spend as much time as a I did :)
>
> This is the configuration:
>
> *Gateway*
> *freeswitch/conf/sip_profiles/external/goip.xml*
>
> *Tips to check that everything is OK
>  - Remember to remove the PIN number from your SIM card. Double check your
> SIM card with a Phone. It should not ask you for a  PIN. Disable it! Period
> :)
>  - *sofia status profile internal* (the Goip extension user must appear, I
> chose ext. 1000 for it...)
> - * sofia status* (to check that the Goip gateway is properly configured.
>  - On the Goip status page the Phone Status and your GSM Status must be
> LOGIN
>  - On the Goip status page GSM operator GSM signal must show something. If
> it doesn't it's because it has not logged properly with your GSM Sim card.
> Double check it with a Phone.
>  - If the Goip is logged properly with FS you can use your favorite SIP
> client to log directly into GOIP that is to say. if your GOIP IP is
> 192.168.2.2 you can log into 1000 at 192.168.2.2 *(again this is not your FS
> IP
> is the GOIP IP)* and make an outbound call to an ordinary cell phone. This
> is useful in case you have trouble with the outbound call.
>
> * Things to find out :??
> - First I used X-Lite, Windows XP and miniSip server 20 clients to test
> this
> gateway. It's kind of strange that with miniSip every time I call to the
> extension I chose for Goip a tone appears an then I can dial a cell phone
> number. FS says that this extension is not available, it shows up with the
> sofia status profile internal though.
>
> *I've used with Linux Ubuntu:
>  - FreeSWITCH Version 1.0.head (git-decfdbb 2011-08-11 14-15-26 -0500)
>  - Jitsy Java Sip Client
>  - 3 Android Phones with CSipSimple, Sipdroid and 3CXPhone Sip Client
>  - 2 linksys spa941
>  - 1 Goip GSM GateWay
>
> http://goip.com.ua/download/goip-series-manual.pdf
> http://goip.com.ua/download/goip-series-manual.pdf
>
> http://www.alibaba.com/trade/search?SearchText=goip+gsm+gateway&Country=&IndexArea=product_en&fsb=y&CatId=509
>
> http://www.alibaba.com/trade/search?SearchText=goip+gsm+gateway&Country=&IndexArea=product_en&fsb=y&CatId=509
>
>
> I hope you find this info useful, if you have any comments or requests or
> corrections... please let me know...
>
> best regards.
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Goip-GSM-Gateway-works-great-with-FreeSwitch-tp6691087p6691087.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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110816/7603b1f8/attachment.html 

From avi at avimarcus.net  Tue Aug 16 23:52:45 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 16 Aug 2011 22:52:45 +0300
Subject: [Freeswitch-users] Billing solutions
In-Reply-To: <CADrcvV9wk2aQJA2GNMNDf4RN98G0Pd7Gjkt6-y_S3d10Nz5k9A@mail.gmail.com>
References: <CADrcvV9wk2aQJA2GNMNDf4RN98G0Pd7Gjkt6-y_S3d10Nz5k9A@mail.gmail.com>
Message-ID: <CALNZuwnt1NXJkhLH4G0eMXKTR7dEq+neGdMXT9R6mab23yc=aA@mail.gmail.com>

Check http://wiki.freeswitch.org/wiki/Billing
-Avi

On Tue, Aug 16, 2011 at 10:38 PM, Cesar Bermudez
<cesar.bermudez at gmail.com>wrote:

> Hi FS users.
>
> I want to know, what billing solutions are available for FS.
> Open Source and not open source.
> Features, etc.
>
> Best regards
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110816/5f5fbcf9/attachment.html 

From lakersman2006 at yahoo.com  Tue Aug 16 23:57:07 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Tue, 16 Aug 2011 12:57:07 -0700 (PDT)
Subject: [Freeswitch-users] ORIGINATE_DISPOSITION
In-Reply-To: <CAKbxfG92amS7=q4ex-s7xUJpxHu3QPW6cfdGZ29kgOA0hDU3HA@mail.gmail.com>
References: <1313447714.89555.YahooMailNeo@web161001.mail.bf1.yahoo.com>
	<CAKzWOxVwDaxsFJAr1dVP5nfmDuFXK2tptvai+oqtwZgph3u1kw@mail.gmail.com>
	<1313452154.28177.YahooMailNeo@web161001.mail.bf1.yahoo.com>
	<CAKzWOxWYXy5Sa2g+cCDEDcxWwE10OtrMaJzUyoLdFSDGtDfbyg@mail.gmail.com>
	<1313452622.11918.YahooMailNeo@web161012.mail.bf1.yahoo.com>
	<CAKzWOxVREFK=uzYzT95oXFT0-sURz7yeOwkuQMz6LKjQ6GvoXA@mail.gmail.com>
	<1313453126.47357.YahooMailNeo@web161010.mail.bf1.yahoo.com>
	<CAKzWOxWtxNoto1c6Chu5dZLPkt+ATcwCGpHOYtPpWydUrcCFfQ@mail.gmail.com>
	<CA+GmxbZ2RG+RS79F8yjXAR4eeFwFy3EMV0k4YiESwq3qMCbwkA@mail.gmail.com>
	<1313515585.77893.YahooMailNeo@web161019.mail.bf1.yahoo.com>
	<CAKbxfG9N1TUmT=HZQqhJfkLk_YiKc=XPqbBoa-9Rm=+Q97PZJw@mail.gmail.com>
	<1313520491.44910.YahooMailNeo@web161015.mail.bf1.yahoo.com>
	<CAKbxfG-3UDi59h3VcOm7+3+OXgaaX1fjNRLZiePrjVc6Ba10Cw@mail.gmail.com>
	<1313522335.98043.YahooMailNeo@web161008.mail.bf1.yahoo.com>
	<CAKbxfG92amS7=q4ex-s7xUJpxHu3QPW6cfdGZ29kgOA0hDU3HA@mail.gmail.com>
Message-ID: <1313524627.43646.YahooMailNeo@web161009.mail.bf1.yahoo.com>

Thanks for the clarification. Another question, when the bridge call is answered and hung up, how would I get the call duration or the time the call is hung up? I saw that the channel variable "hangup_time" is 0 on an answered call.



________________________________
From: Anthony Minessale <anthony.minessale at gmail.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, August 16, 2011 12:24 PM
Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION

that is what bridge_early_media=true means, so you can hear the early
media during originate but stil not return until the channel is
answered or hungup.

On Tue, Aug 16, 2011 at 2:18 PM, Sam <lakersman2006 at yahoo.com> wrote:
> I need to be able to play back early media to A leg, so if
> {ignore_early_media=false} there is no way to tell from
> originate_disposition that the bridge was answered?
>
> ________________________________
> From: Anthony Minessale <anthony.minessale at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Sent: Tuesday, August 16, 2011 11:56 AM
> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>
> bridge_early_media implies ignore_early_media true so you can't use
> them together but yes since there is no cause code for answered
> because the call has not ended that would guarantee it was answered if
> you exited originate with either ignore_early_media=true or
> bridge_early_media=true
>
> On Tue, Aug 16, 2011 at 1:48 PM, Sam <lakersman2006 at yahoo.com> wrote:
>> "if you call somewhere that uses early media, the originate will end
>> setting originate_disposition to SUCCESS meaning that a live channel
>> was produced.? Now the bridge will begin between the A and B leg."
>> Does that mean it is safe to assume when "originate_disposition" is
>> SUCCESS,
>> that the call has been answered if I set the following:
>>
>> {ignore_early_media=false}
>> {bridge_early_media=true}
>>
>>
>> ________________________________
>> From: Anthony Minessale <anthony.minessale at gmail.com>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Sent: Tuesday, August 16, 2011 10:54 AM
>> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>
>> The originate disposition of A leg will always have the last known status
>> from B
>>
>> originate, by default, returns when:
>>
>> 1) media is established on a specific outgoing leg, answered or not
>> 2) all outgoing legs are terminated.
>>
>> if you call somewhere that uses early media, the originate will end
>> setting originate_disposition to SUCCESS meaning that a live channel
>> was produced.? Now the bridge will begin between the A and B leg.
>>
>> When the bridge ends if the B leg is hungup, its cause will be stored
>> in "bridge_hangup_cause"
>>
>> if you want to keep the originate from ending when early media is
>> established you can add {ignore_early_media=true} or
>> {bridge_early_media=true} prepended to your dial string and the
>> originate will never return untill all outbound legs are either hungup
>> or one is answered.
>>
>>
>> You should take some time to expand your mind to the different
>> paradigm in FreeSWITCH where you may have as many as 10 outbound legs
>> at once in a forked-dial situation and some of what you think is
>> simple and obvious will quickly dissolve.
>>
>> Another thing you can do is set the variable "failed_xml_cdr_prefix"
>> on the A leg.
>> This prefix will be mixed with an incrementing variable for each
>> outbound call leg and in the case of a failure the entire XML cdr will
>> be set into a var on A leg.
>>
>> for instance if you set failed_xml_cdr_prefix=foo you would get foo_1
>> foo_2 etc depending on the number of outbound call legs.
>>
>> Additionally you can set copy_xml_cdr on the A leg and when the bridge
>> ends you will get a complete CDR for B in the "b_leg_cdr" variable on
>> A
>>
>> Finally you should really go with the flow of how FreeSWITCH is
>> engineered and try to keep your accounting logic in a separate place
>> and monitor the XML-CDR, CDR-CSV or event_socket +
>> CHANNEL_HANGUP_COMPLETE events to process this information.? There is
>> much more to a call and what happens when it's transferred etc than
>> what you can get in a single monolithic perspective of inside the
>> channel.? Plus it's conter intuitive to put routing, application and
>> accounting logic in the same place.
>>
>>
>>
>> On Tue, Aug 16, 2011 at 12:26 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>> Dmitry,
>>> Yes, I agree. For an novice like me it was extremely confusing and
>>> frustrating that I could not get the proper B leg disposition when A leg
>>> is
>>> answered by FS itself.
>>> I hope there can be some methods or functions that will allow us to get
>>> the
>>> proper B Leg disposition.
>>> ________________________________
>>> From: Dmitry Sytchev <kbdfck at gmail.com>
>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> Sent: Tuesday, August 16, 2011 2:30 AM
>>> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>>
>>> Seems we need to clear things about B-leg disposition in wiki.
>>> AFAIK there is no method to get correct disposition from B leg without
>>> analyzing events in case when A-leg was answered by FS itself.
>>> If A leg was not answered, we can use A-leg disposition for call
>>> disposition.
>>>
>>> 2011/8/16 Michael Collins <msc at freeswitch.org>
>>>
>>>
>>> On Mon, Aug 15, 2011 at 5:05 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>>
>>> I am using perl's $session->get_variable("originate_disposition");
>>>
>>> Are you looking at the b-leg's session?
>>>
>>>
>>> Also, how come the "hangup_time" shows zero on answered calls?
>>>
>>> Because hangup_time refers to the point in time at which the call was
>>> hung
>>> up. Since you are in the middle of a call (using the $session object) you
>>> will never see the hangup_time because the object ceases to exist once
>>> the
>>> call leg is disconnected.
>>> I get the impression that you may be using the wrong tool for this
>>> particular job, but I'm not sure without seeing it. If you don't mind
>>> dropping it on pastebin we'll have a look and give you some suggestions.
>>> -MC
>>>
>>> ________________________________
>>> From: Michael Collins <msc at freeswitch.org>
>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> Sent: Monday, August 15, 2011 4:58 PM
>>> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>>
>>> And how are you checking the variable? Do you have an event socket open
>>> or
>>> ... ?
>>> -MC
>>>
>>> On Mon, Aug 15, 2011 at 4:57 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>>
>>> It is being generated with a bridge.
>>>
>>> ________________________________
>>> From: Michael Collins <msc at freeswitch.org>
>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> Sent: Monday, August 15, 2011 4:54 PM
>>> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>>
>>> And how is it being generated? WIth a bridge or originate or ... ?
>>> -MC
>>>
>>> On Mon, Aug 15, 2011 at 4:49 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>>
>>> I want to know the B-leg status of the call.
>>>
>>> ________________________________
>>> From: Michael Collins <msc at freeswitch.org>
>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> Sent: Monday, August 15, 2011 4:31 PM
>>> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>>
>>> I think you may be wanting "endpoint_disposition" depending on exactly
>>> what
>>> you're looking at.
>>> -MC
>>>
>>> On Mon, Aug 15, 2011 at 3:35 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>>
>>> For the ORIGINATE_DISPOSITION channel variable, does "SUCCESS" mean the
>>> call
>>> was ANSWERED?
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> --
>>> Best regards,
>>>
>>> Dmitry Sytchev,
>>> IT Engineer
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> 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
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110816/831f24fb/attachment-0001.html 

From nasida at live.ru  Wed Aug 17 00:06:03 2011
From: nasida at live.ru (Yuriy Nasida)
Date: Wed, 17 Aug 2011 00:06:03 +0400
Subject: [Freeswitch-users] Lua not playing wav files
In-Reply-To: <SNT127-W400A9FAAD362C48C2F3B5CD5290@phx.gbl>
References: <SNT127-W400A9FAAD362C48C2F3B5CD5290@phx.gbl>
Message-ID: <SNT127-W499B3A665B2B6809731A40D5290@phx.gbl>


The issue  consists in that I get dialplan automatically through xml_curl + php (by intralanman). 
I see it by means xml_curl_debug_on
<action application="lua" data="$1" inline="true"/>
How can I disable "inline="true"" ?


From: nasida at live.ru
To: freeswitch-users at lists.freeswitch.org
Date: Tue, 16 Aug 2011 21:46:35 +0400
Subject: [Freeswitch-users] Lua not playing wav files









Hi
Freeswitch-users,




My
simple lua script:




freeswitch.consoleLog("err","start
hello.lua\n")

session:answer();

message
= "ivr/ivr-enter_destination_telephone_number.wav"

session:execute("playback",
message)

session:hangup();




Script
looks fine I think, but FS doesn't play audio. If I use corresponding
XML dialplan all work fine.







logs
when I use lua:




2011-08-16
13:18:00.336003 [DEBUG] switch_core_state_machine.c:371
(sofia/external/79213777785 at 65.98.107.130:5080) State EXECUTE going
to sleep

2011-08-16
13:18:00.336003 [DEBUG] switch_core_state_machine.c:364
(sofia/external/79213777785 at 65.98.107.130:5080) State ROUTING

2011-08-16
13:18:00.336003 [DEBUG] mod_sofia.c:147
sofia/external/79213777785 at 65.98.107.130:5080 SOFIA ROUTING

2011-08-16
13:18:00.336003 [DEBUG] switch_core_state_machine.c:77
sofia/external/79213777785 at 65.98.107.130:5080 Standard ROUTING

2011-08-16
13:18:00.336003 [INFO] mod_dialplan_xml.c:331 Processing unknown
<79213777785>->inbound_type_uri in context public

2011-08-16
13:18:00.345157 [ERR] switch_cpp.cpp:1197 start hello.lua

2011-08-16
13:18:00.345157 [DEBUG] sofia_glue.c:4650 Audio Codec Compare
[PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000]

2011-08-16
13:18:00.345157 [DEBUG] sofia_glue.c:4650 Audio Codec Compare
[PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]

2011-08-16
13:18:00.345157 [DEBUG] sofia_glue.c:2773 Set Codec
sofia/external/79213777785 at 65.98.107.130:5080 PCMU/8000 20 ms 160
samples 64000 bits

2011-08-16
13:18:00.346166 [DEBUG] sofia_glue.c:4764 Set 2833 dtmf send/recv
payload to 101

2011-08-16
13:18:00.346166 [DEBUG] sofia_glue.c:3014 AUDIO RTP
[sofia/external/79213777785 at 65.98.107.130:5080] 65.98.107.130 port
26266 -> 212.232.72.134 port 49276 codec: 0 ms: 20

2011-08-16
13:18:00.346166 [DEBUG] switch_rtp.c:1623 Starting timer [soft] 160
bytes per 20ms

2011-08-16
13:18:00.347205 [DEBUG] sofia_glue.c:3276 Set 2833 dtmf send payload
to 101

2011-08-16
13:18:00.347205 [DEBUG] sofia_glue.c:3281 Set 2833 dtmf receive
payload to 101

2011-08-16
13:18:00.347205 [DEBUG] mod_sofia.c:681 Local SDP
sofia/external/79213777785 at 65.98.107.130:5080:

v=0

o=FreeSWITCH
1313488814 1313488815 IN IP4 65.98.107.130

s=FreeSWITCH

c=IN
IP4 65.98.107.130

t=0 0

m=audio
26266 RTP/AVP 0 101

a=rtpmap:0
PCMU/8000

a=rtpmap:101
telephone-event/8000

a=fmtp:101
0-16

a=silenceSupp:off
- - - -

a=ptime:20

a=sendrecv




2011-08-16
13:18:00.348214 [DEBUG] sofia.c:4761 Channel
sofia/external/79213777785 at 65.98.107.130:5080 entering state
[completed][200]

2011-08-16
13:18:00.348214 [DEBUG] switch_core_session.c:1939 Application
playback Requires media! pre_answering channel
sofia/external/79213777785 at 65.98.107.130:5080

2011-08-16
13:18:00.348214 [DEBUG] switch_cpp.cpp:618 CoreSession::hangup

2011-08-16
13:18:00.348214 [DEBUG] switch_cpp.cpp:988
sofia/external/79213777785 at 65.98.107.130:5080 destroy/unlink session
from object

2011-08-16
13:18:00.348214 [DEBUG] switch_core_state_machine.c:364
(sofia/external/79213777785 at 65.98.107.130:5080) State ROUTING going
to sleep"










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




logs
when I use XML dialplan:




2011-08-16
13:19:49.609765 [DEBUG] switch_core_state_machine.c:371
(sofia/external/79213777785 at 65.98.107.130:5080) State EXECUTE going
to sleep

2011-08-16
13:19:49.609765 [DEBUG] switch_core_state_machine.c:364
(sofia/external/79213777785 at 65.98.107.130:5080) State ROUTING

2011-08-16
13:19:49.609765 [DEBUG] mod_sofia.c:147
sofia/external/79213777785 at 65.98.107.130:5080 SOFIA ROUTING

2011-08-16
13:19:49.609765 [DEBUG] switch_core_state_machine.c:77
sofia/external/79213777785 at 65.98.107.130:5080 Standard ROUTING

2011-08-16
13:19:49.609765 [INFO] mod_dialplan_xml.c:331 Processing unknown
<79213777785>->inbound_type_uri in context public

2011-08-16
13:19:49.615894 [DEBUG] switch_core_state_machine.c:364
(sofia/external/79213777785 at 65.98.107.130:5080) State ROUTING going
to sleep

2011-08-16
13:19:49.615894 [DEBUG] switch_core_state_machine.c:371
(sofia/external/79213777785 at 65.98.107.130:5080) State EXECUTE

2011-08-16
13:19:49.615894 [DEBUG] mod_sofia.c:240
sofia/external/79213777785 at 65.98.107.130:5080 SOFIA EXECUTE

2011-08-16
13:19:49.615894 [DEBUG] switch_core_state_machine.c:157
sofia/external/79213777785 at 65.98.107.130:5080 Standard EXECUTE

2011-08-16
13:19:49.615894 [DEBUG] sofia_glue.c:4650 Audio Codec Compare
[PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000]

2011-08-16
13:19:49.615894 [DEBUG] sofia_glue.c:4650 Audio Codec Compare
[PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]

2011-08-16
13:19:49.616902 [DEBUG] sofia_glue.c:2773 Set Codec
sofia/external/79213777785 at 65.98.107.130:5080 PCMU/8000 20 ms 160
samples 64000 bits

2011-08-16
13:19:49.616902 [DEBUG] sofia_glue.c:4764 Set 2833 dtmf send/recv
payload to 101

2011-08-16
13:19:49.616902 [DEBUG] sofia_glue.c:3014 AUDIO RTP
[sofia/external/79213777785 at 65.98.107.130:5080] 65.98.107.130 port
23946 -> 212.232.72.134 port 49278 codec: 0 ms: 20

2011-08-16
13:19:49.616902 [DEBUG] switch_rtp.c:1623 Starting timer [soft] 160
bytes per 20ms

2011-08-16
13:19:49.617998 [DEBUG] sofia_glue.c:3276 Set 2833 dtmf send payload
to 101

2011-08-16
13:19:49.617998 [DEBUG] sofia_glue.c:3281 Set 2833 dtmf receive
payload to 101

2011-08-16
13:19:49.617998 [DEBUG] mod_sofia.c:681 Local SDP
sofia/external/79213777785 at 65.98.107.130:5080:

v=0

o=FreeSWITCH
1313491243 1313491244 IN IP4 65.98.107.130

s=FreeSWITCH

c=IN
IP4 65.98.107.130

t=0 0

m=audio
23946 RTP/AVP 0 101

a=rtpmap:0
PCMU/8000

a=rtpmap:101
telephone-event/8000

a=fmtp:101
0-16

a=silenceSupp:off
- - - -

a=ptime:20

a=sendrecv




2011-08-16
13:19:49.619008 [DEBUG] sofia.c:4761 Channel
sofia/external/79213777785 at 65.98.107.130:5080 entering state
[completed][200]

2011-08-16
13:19:49.631344 [DEBUG] switch_ivr_play_say.c:1278 Codec Activated
L16 at 8000hz 1 channels 20ms

2011-08-16
13:19:49.740067 [DEBUG] sofia.c:4761 Channel
sofia/external/79213777785 at 65.98.107.130:5080 entering state
[ready][200]

2011-08-16
13:19:52.879234 [DEBUG] switch_ivr_play_say.c:1648 done playing file

2011-08-16
13:19:52.880333 [NOTICE] switch_core_state_machine.c:189
sofia/external/79213777785 at 65.98.107.130:5080 has executed the last
dialplan instruction, hanging up.







I
have compared logs and saw that case without lua have some strings
unlike case with lua.




"2011-08-16
13:19:49.615894 [DEBUG] switch_core_state_machine.c:364
(sofia/external/79213777785 at 65.98.107.130:5080) State ROUTING going
to sleep

2011-08-16
13:19:49.615894 [DEBUG] switch_core_state_machine.c:371
(sofia/external/79213777785 at 65.98.107.130:5080) State EXECUTE

2011-08-16
13:19:49.615894 [DEBUG] mod_sofia.c:240
sofia/external/79213777785 at 65.98.107.130:5080 SOFIA EXECUTE

2011-08-16
13:19:49.615894 [DEBUG] switch_core_state_machine.c:157
sofia/external/79213777785 at 65.98.107.130:5080 Standard EXECUTE"






Many thanks to anyone who can 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 		 	   		  
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/02acaac3/attachment-0001.html 

From tacvbo at tacvbo.net  Wed Aug 17 00:07:19 2011
From: tacvbo at tacvbo.net (Octavio Ruiz)
Date: Tue, 16 Aug 2011 15:07:19 -0500
Subject: [Freeswitch-users] Building Freeswitch + FreeTDM + LibSng-ISDN RPMs
Message-ID: <CAFeQK0sejfBwWGPX0oqwJ48FT4oygjT7G27GT7tr_9A7kgpVyQ@mail.gmail.com>

I'm trying to build Freeswitch with full Sangoma's ( FreeTDM +
LibSng-ISDN ) support from SRPMs for Centos 5, but there are two not
satisfied dependencies for what looks like non existing RPMs, fact
that raises many questions on my head,

An extract from the freeswitch SPEC file:

-- from freeswitch.spec
Requires: libsng_isdn
BuildRequires: wanpipe
BuildRequires: libsng_isdn

even if looks quite easy to bypass this and get it to compile
satisfying those, either creating the missing libsng_isdn SPEC file
needed for building the package (libsng_isdn is only a bunch of
headers and an already-build shared object) and modifying the
wanpipe.rpmspec in order to change the naming behavior which doesn't
comply with the above (Wanpipe's resulting RPM is named
'wanpipe-<kernel-version>' which do not satisfy the solely string
'wanpipe') or either worst, in terms of cleanliness, creating dummy
RPM packages or removing dependencies on the  freeswitch.spec, I
wonder what is supposed to be the correct environment and proper
procedure for building Freeswitch and Wanrouter + SNG-ISDN following
what the SPEC file states.

-- from wanpipe.rpmspec
%define KERNEL_VERSION    %{?kern_ver}
%define WANPIPE_VER       wanpipe
%define name              %{WANPIPE_VER}

Am I missing something? Has anyone tested this before?  Someone else
is packaging Sangoma's Wanrouter on RPM with other naming scheme?  Are
they on the SPEC file merely to denote upcoming work?


From anthony.minessale at gmail.com  Wed Aug 17 00:26:19 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Aug 2011 15:26:19 -0500
Subject: [Freeswitch-users] ORIGINATE_DISPOSITION
In-Reply-To: <1313524627.43646.YahooMailNeo@web161009.mail.bf1.yahoo.com>
References: <1313447714.89555.YahooMailNeo@web161001.mail.bf1.yahoo.com>
	<CAKzWOxVwDaxsFJAr1dVP5nfmDuFXK2tptvai+oqtwZgph3u1kw@mail.gmail.com>
	<1313452154.28177.YahooMailNeo@web161001.mail.bf1.yahoo.com>
	<CAKzWOxWYXy5Sa2g+cCDEDcxWwE10OtrMaJzUyoLdFSDGtDfbyg@mail.gmail.com>
	<1313452622.11918.YahooMailNeo@web161012.mail.bf1.yahoo.com>
	<CAKzWOxVREFK=uzYzT95oXFT0-sURz7yeOwkuQMz6LKjQ6GvoXA@mail.gmail.com>
	<1313453126.47357.YahooMailNeo@web161010.mail.bf1.yahoo.com>
	<CAKzWOxWtxNoto1c6Chu5dZLPkt+ATcwCGpHOYtPpWydUrcCFfQ@mail.gmail.com>
	<CA+GmxbZ2RG+RS79F8yjXAR4eeFwFy3EMV0k4YiESwq3qMCbwkA@mail.gmail.com>
	<1313515585.77893.YahooMailNeo@web161019.mail.bf1.yahoo.com>
	<CAKbxfG9N1TUmT=HZQqhJfkLk_YiKc=XPqbBoa-9Rm=+Q97PZJw@mail.gmail.com>
	<1313520491.44910.YahooMailNeo@web161015.mail.bf1.yahoo.com>
	<CAKbxfG-3UDi59h3VcOm7+3+OXgaaX1fjNRLZiePrjVc6Ba10Cw@mail.gmail.com>
	<1313522335.98043.YahooMailNeo@web161008.mail.bf1.yahoo.com>
	<CAKbxfG92amS7=q4ex-s7xUJpxHu3QPW6cfdGZ29kgOA0hDU3HA@mail.gmail.com>
	<1313524627.43646.YahooMailNeo@web161009.mail.bf1.yahoo.com>
Message-ID: <CAKbxfG9B=5cC00XJ24=UTn-k97whgd262FFH8mCFUWPxNjTRow@mail.gmail.com>

again because in FreeSWITCH the legs are separate entity, if you
listen for hangup_complete events or log B-legs in your cdrs you will
get precise info about that.  if you really want you can add a
dialplan app to capture a timestamp when the bridge ends to save the
time.  This is of-couse only useful if the A leg does not also hangup
which is why it's best to extract this kind of data from the CDRs.


On Tue, Aug 16, 2011 at 2:57 PM, Sam <lakersman2006 at yahoo.com> wrote:
> Thanks for the clarification. Another question, when the bridge call is
> answered and hung up, how would I get the call duration or the time the call
> is hung up? I saw that the channel variable "hangup_time" is 0 on an
> answered call.
>
> ________________________________
> From: Anthony Minessale <anthony.minessale at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Sent: Tuesday, August 16, 2011 12:24 PM
> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>
> that is what bridge_early_media=true means, so you can hear the early
> media during originate but stil not return until the channel is
> answered or hungup.
>
> On Tue, Aug 16, 2011 at 2:18 PM, Sam <lakersman2006 at yahoo.com> wrote:
>> I need to be able to play back early media to A leg, so if
>> {ignore_early_media=false} there is no way to tell from
>> originate_disposition that the bridge was answered?
>>
>> ________________________________
>> From: Anthony Minessale <anthony.minessale at gmail.com>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Sent: Tuesday, August 16, 2011 11:56 AM
>> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>
>> bridge_early_media implies ignore_early_media true so you can't use
>> them together but yes since there is no cause code for answered
>> because the call has not ended that would guarantee it was answered if
>> you exited originate with either ignore_early_media=true or
>> bridge_early_media=true
>>
>> On Tue, Aug 16, 2011 at 1:48 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>> "if you call somewhere that uses early media, the originate will end
>>> setting originate_disposition to SUCCESS meaning that a live channel
>>> was produced.? Now the bridge will begin between the A and B leg."
>>> Does that mean it is safe to assume when "originate_disposition" is
>>> SUCCESS,
>>> that the call has been answered if I set the following:
>>>
>>> {ignore_early_media=false}
>>> {bridge_early_media=true}
>>>
>>>
>>> ________________________________
>>> From: Anthony Minessale <anthony.minessale at gmail.com>
>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> Sent: Tuesday, August 16, 2011 10:54 AM
>>> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>>
>>> The originate disposition of A leg will always have the last known status
>>> from B
>>>
>>> originate, by default, returns when:
>>>
>>> 1) media is established on a specific outgoing leg, answered or not
>>> 2) all outgoing legs are terminated.
>>>
>>> if you call somewhere that uses early media, the originate will end
>>> setting originate_disposition to SUCCESS meaning that a live channel
>>> was produced.? Now the bridge will begin between the A and B leg.
>>>
>>> When the bridge ends if the B leg is hungup, its cause will be stored
>>> in "bridge_hangup_cause"
>>>
>>> if you want to keep the originate from ending when early media is
>>> established you can add {ignore_early_media=true} or
>>> {bridge_early_media=true} prepended to your dial string and the
>>> originate will never return untill all outbound legs are either hungup
>>> or one is answered.
>>>
>>>
>>> You should take some time to expand your mind to the different
>>> paradigm in FreeSWITCH where you may have as many as 10 outbound legs
>>> at once in a forked-dial situation and some of what you think is
>>> simple and obvious will quickly dissolve.
>>>
>>> Another thing you can do is set the variable "failed_xml_cdr_prefix"
>>> on the A leg.
>>> This prefix will be mixed with an incrementing variable for each
>>> outbound call leg and in the case of a failure the entire XML cdr will
>>> be set into a var on A leg.
>>>
>>> for instance if you set failed_xml_cdr_prefix=foo you would get foo_1
>>> foo_2 etc depending on the number of outbound call legs.
>>>
>>> Additionally you can set copy_xml_cdr on the A leg and when the bridge
>>> ends you will get a complete CDR for B in the "b_leg_cdr" variable on
>>> A
>>>
>>> Finally you should really go with the flow of how FreeSWITCH is
>>> engineered and try to keep your accounting logic in a separate place
>>> and monitor the XML-CDR, CDR-CSV or event_socket +
>>> CHANNEL_HANGUP_COMPLETE events to process this information.? There is
>>> much more to a call and what happens when it's transferred etc than
>>> what you can get in a single monolithic perspective of inside the
>>> channel.? Plus it's conter intuitive to put routing, application and
>>> accounting logic in the same place.
>>>
>>>
>>>
>>> On Tue, Aug 16, 2011 at 12:26 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>>> Dmitry,
>>>> Yes, I agree. For an novice like me it was extremely confusing and
>>>> frustrating that I could not get the proper B leg disposition when A leg
>>>> is
>>>> answered by FS itself.
>>>> I hope there can be some methods or functions that will allow us to get
>>>> the
>>>> proper B Leg disposition.
>>>> ________________________________
>>>> From: Dmitry Sytchev <kbdfck at gmail.com>
>>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>> Sent: Tuesday, August 16, 2011 2:30 AM
>>>> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>>>
>>>> Seems we need to clear things about B-leg disposition in wiki.
>>>> AFAIK there is no method to get correct disposition from B leg without
>>>> analyzing events in case when A-leg was answered by FS itself.
>>>> If A leg was not answered, we can use A-leg disposition for call
>>>> disposition.
>>>>
>>>> 2011/8/16 Michael Collins <msc at freeswitch.org>
>>>>
>>>>
>>>> On Mon, Aug 15, 2011 at 5:05 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>>>
>>>> I am using perl's $session->get_variable("originate_disposition");
>>>>
>>>> Are you looking at the b-leg's session?
>>>>
>>>>
>>>> Also, how come the "hangup_time" shows zero on answered calls?
>>>>
>>>> Because hangup_time refers to the point in time at which the call was
>>>> hung
>>>> up. Since you are in the middle of a call (using the $session object)
>>>> you
>>>> will never see the hangup_time because the object ceases to exist once
>>>> the
>>>> call leg is disconnected.
>>>> I get the impression that you may be using the wrong tool for this
>>>> particular job, but I'm not sure without seeing it. If you don't mind
>>>> dropping it on pastebin we'll have a look and give you some suggestions.
>>>> -MC
>>>>
>>>> ________________________________
>>>> From: Michael Collins <msc at freeswitch.org>
>>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>> Sent: Monday, August 15, 2011 4:58 PM
>>>> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>>>
>>>> And how are you checking the variable? Do you have an event socket open
>>>> or
>>>> ... ?
>>>> -MC
>>>>
>>>> On Mon, Aug 15, 2011 at 4:57 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>>>
>>>> It is being generated with a bridge.
>>>>
>>>> ________________________________
>>>> From: Michael Collins <msc at freeswitch.org>
>>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>> Sent: Monday, August 15, 2011 4:54 PM
>>>> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>>>
>>>> And how is it being generated? WIth a bridge or originate or ... ?
>>>> -MC
>>>>
>>>> On Mon, Aug 15, 2011 at 4:49 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>>>
>>>> I want to know the B-leg status of the call.
>>>>
>>>> ________________________________
>>>> From: Michael Collins <msc at freeswitch.org>
>>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>> Sent: Monday, August 15, 2011 4:31 PM
>>>> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>>>
>>>> I think you may be wanting "endpoint_disposition" depending on exactly
>>>> what
>>>> you're looking at.
>>>> -MC
>>>>
>>>> On Mon, Aug 15, 2011 at 3:35 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>>>
>>>> For the ORIGINATE_DISPOSITION channel variable, does "SUCCESS" mean the
>>>> call
>>>> was ANSWERED?
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Best regards,
>>>>
>>>> Dmitry Sytchev,
>>>> IT Engineer
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> 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
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> 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
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From anthony.minessale at gmail.com  Wed Aug 17 00:52:38 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Aug 2011 15:52:38 -0500
Subject: [Freeswitch-users] any IVR example in C/C++?
In-Reply-To: <4E4A927A.4050102@hw.ac.uk>
References: <4E43EE36.60209@hw.ac.uk>
	<CAA4nhyAnC0HfxJJPGpOdiBGaG__gBf6Bp9KVi6spjDQgjLTZ5w@mail.gmail.com>
	<CALXCt0paCyLs3ojFVeWk9uO7VuuuEgXd_owR_10o0p5L-WStag@mail.gmail.com>
	<31462.1313279925@ccs.covici.com>
	<CAPerZE93NU+ccKdmGEHufb11rRv0S+L1m0=riKibtgUX2VLU1A@mail.gmail.com>
	<4E4A927A.4050102@hw.ac.uk>
Message-ID: <CAKbxfG8wK1rMKECft68Ws1KtnPuptoZjqF1OZHfqKqMG2uY1oQ@mail.gmail.com>

also if you look at switch_cpp.cpp its the c++ wrapper we use to
create all the embedded language modules so the documentation for the
embedded lua api's would also apply to that C++ wrapper if you chose
to use it directly.


On Tue, Aug 16, 2011 at 10:53 AM, xl127 <x.liu at hw.ac.uk> wrote:
> Thanks for everybody's points, helpful!
>
> Cheers,
> Xing
>
>
> On 14/08/11 04:04, Campbell Steven wrote:
>> Try here:
>>
>> http://fisheye.freeswitch.org/browse/freeswitch.git/src/mod/applications/mod_protovm
>>
>> Campbell
>>
>> On Sun, Aug 14, 2011 at 11:58 AM,<covici at ccs.covici.com> ?wrote:
>>> Where can I find Mock's voicemail -- I don't see it in contrib?
>>>
>>> Giovanni Maruzzelli<gmaruzz at gmail.com> ?wrote:
>>>
>>>> Also, you can check mod-voicemail.c and the new mod-voicemail made by
>>>> Moc, those are the only IVR written in C that I know about (they're
>>>> written in C because voicemail is considered a base feature, and been
>>>> written in C assure stability because people does not fiddle with
>>>> them)
>>>>
>>>> On 8/13/11, Moises Silva<moises.silva at gmail.com> ?wrote:
>>>>> On Thu, Aug 11, 2011 at 10:59 AM, xl127<x.liu at hw.ac.uk> ?wrote:
>>>>>> I am wondering how I could do this for a C/C++ application?
>>>>>> And in the scripts languages I can set a callback method, e.g.
>>>>>> ? ?session.setInputCallback(myInputCallback)
>>>>>> but I didn't find how to do this in C/C++.
>>>>> The default question here is, why do you need C/C++ for an IVR?
>>>>> FreeSWITCH allows you to use simpler/safer languages to build IVR's.
>>>>>
>>>>> You can certainly do it, but the reason you don't find examples is
>>>>> probably because most people understand there is no need for C/C++
>>>>> there. Having said that, you can take a look at the IVR/say/play API's
>>>>> in switch_ivr_play_say.c to find out how to provide a callback to the
>>>>> different API's thru the switch_input_args_t structure.
>>>>>
>>>>> Moises Silva
>>>>> Senior Software Engineer, Software Development Manager
>>>>> Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
>>>>> L3R 9R6 Canada
>>>>> t. 1 905 474 1990 x128 | e. moy at sangoma.com
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>> --
>>>> Sent from my mobile device
>>>>
>>>> Sincerely,
>>>>
>>>> Giovanni Maruzzelli
>>>> Cell : +39-347-2665618
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> --
>>> Your life is like a penny. ?You're going to lose it. ?The question is:
>>> How do
>>> you spend it?
>>>
>>> ? ? ? ? ?John Covici
>>> ? ? ? ? ?covici at ccs.covici.com
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Heriot-Watt University is a Scottish charity
> registered under charity number SC000278.
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From lakersman2006 at yahoo.com  Wed Aug 17 01:21:38 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Tue, 16 Aug 2011 14:21:38 -0700 (PDT)
Subject: [Freeswitch-users] ORIGINATE_DISPOSITION
In-Reply-To: <CAKbxfG9B=5cC00XJ24=UTn-k97whgd262FFH8mCFUWPxNjTRow@mail.gmail.com>
References: <1313447714.89555.YahooMailNeo@web161001.mail.bf1.yahoo.com>
	<CAKzWOxVwDaxsFJAr1dVP5nfmDuFXK2tptvai+oqtwZgph3u1kw@mail.gmail.com>
	<1313452154.28177.YahooMailNeo@web161001.mail.bf1.yahoo.com>
	<CAKzWOxWYXy5Sa2g+cCDEDcxWwE10OtrMaJzUyoLdFSDGtDfbyg@mail.gmail.com>
	<1313452622.11918.YahooMailNeo@web161012.mail.bf1.yahoo.com>
	<CAKzWOxVREFK=uzYzT95oXFT0-sURz7yeOwkuQMz6LKjQ6GvoXA@mail.gmail.com>
	<1313453126.47357.YahooMailNeo@web161010.mail.bf1.yahoo.com>
	<CAKzWOxWtxNoto1c6Chu5dZLPkt+ATcwCGpHOYtPpWydUrcCFfQ@mail.gmail.com>
	<CA+GmxbZ2RG+RS79F8yjXAR4eeFwFy3EMV0k4YiESwq3qMCbwkA@mail.gmail.com>
	<1313515585.77893.YahooMailNeo@web161019.mail.bf1.yahoo.com>
	<CAKbxfG9N1TUmT=HZQqhJfkLk_YiKc=XPqbBoa-9Rm=+Q97PZJw@mail.gmail.com>
	<1313520491.44910.YahooMailNeo@web161015.mail.bf1.yahoo.com>
	<CAKbxfG-3UDi59h3VcOm7+3+OXgaaX1fjNRLZiePrjVc6Ba10Cw@mail.gmail.com>
	<1313522335.98043.YahooMailNeo@web161008.mail.bf1.yahoo.com>
	<CAKbxfG92amS7=q4ex-s7xUJpxHu3QPW6cfdGZ29kgOA0hDU3HA@mail.gmail.com>
	<1313524627.43646.YahooMailNeo@web161009.mail.bf1.yahoo.com>
	<CAKbxfG9B=5cC00XJ24=UTn-k97whgd262FFH8mCFUWPxNjTRow@mail.gmail.com>
Message-ID: <1313529698.72008.YahooMailNeo@web161004.mail.bf1.yahoo.com>

I have tried to use the xml cdr to find the call duration, etc. but in the xml cdr, freeswitch does not return any of the duration fields, here is a pastebin of my xml output when using the perl $session->getXMLCDR(); and yes I have also set "process_cdr=b_only".

http://pastebin.freeswitch.org/17058



________________________________
From: Anthony Minessale <anthony.minessale at gmail.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, August 16, 2011 1:26 PM
Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION

again because in FreeSWITCH the legs are separate entity, if you
listen for hangup_complete events or log B-legs in your cdrs you will
get precise info about that.? if you really want you can add a
dialplan app to capture a timestamp when the bridge ends to save the
time.? This is of-couse only useful if the A leg does not also hangup
which is why it's best to extract this kind of data from the CDRs.


On Tue, Aug 16, 2011 at 2:57 PM, Sam <lakersman2006 at yahoo.com> wrote:
> Thanks for the clarification. Another question, when the bridge call is
> answered and hung up, how would I get the call duration or the time the call
> is hung up? I saw that the channel variable "hangup_time" is 0 on an
> answered call.
>
> ________________________________
> From: Anthony Minessale <anthony.minessale at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Sent: Tuesday, August 16, 2011 12:24 PM
> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>
> that is what bridge_early_media=true means, so you can hear the early
> media during originate but stil not return until the channel is
> answered or hungup.
>
> On Tue, Aug 16, 2011 at 2:18 PM, Sam <lakersman2006 at yahoo.com> wrote:
>> I need to be able to play back early media to A leg, so if
>> {ignore_early_media=false} there is no way to tell from
>> originate_disposition that the bridge was answered?
>>
>> ________________________________
>> From: Anthony Minessale <anthony.minessale at gmail.com>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Sent: Tuesday, August 16, 2011 11:56 AM
>> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>
>> bridge_early_media implies ignore_early_media true so you can't use
>> them together but yes since there is no cause code for answered
>> because the call has not ended that would guarantee it was answered if
>> you exited originate with either ignore_early_media=true or
>> bridge_early_media=true
>>
>> On Tue, Aug 16, 2011 at 1:48 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>> "if you call somewhere that uses early media, the originate will end
>>> setting originate_disposition to SUCCESS meaning that a live channel
>>> was produced.? Now the bridge will begin between the A and B leg."
>>> Does that mean it is safe to assume when "originate_disposition" is
>>> SUCCESS,
>>> that the call has been answered if I set the following:
>>>
>>> {ignore_early_media=false}
>>> {bridge_early_media=true}
>>>
>>>
>>> ________________________________
>>> From: Anthony Minessale <anthony.minessale at gmail.com>
>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> Sent: Tuesday, August 16, 2011 10:54 AM
>>> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>>
>>> The originate disposition of A leg will always have the last known status
>>> from B
>>>
>>> originate, by default, returns when:
>>>
>>> 1) media is established on a specific outgoing leg, answered or not
>>> 2) all outgoing legs are terminated.
>>>
>>> if you call somewhere that uses early media, the originate will end
>>> setting originate_disposition to SUCCESS meaning that a live channel
>>> was produced.? Now the bridge will begin between the A and B leg.
>>>
>>> When the bridge ends if the B leg is hungup, its cause will be stored
>>> in "bridge_hangup_cause"
>>>
>>> if you want to keep the originate from ending when early media is
>>> established you can add {ignore_early_media=true} or
>>> {bridge_early_media=true} prepended to your dial string and the
>>> originate will never return untill all outbound legs are either hungup
>>> or one is answered.
>>>
>>>
>>> You should take some time to expand your mind to the different
>>> paradigm in FreeSWITCH where you may have as many as 10 outbound legs
>>> at once in a forked-dial situation and some of what you think is
>>> simple and obvious will quickly dissolve.
>>>
>>> Another thing you can do is set the variable "failed_xml_cdr_prefix"
>>> on the A leg.
>>> This prefix will be mixed with an incrementing variable for each
>>> outbound call leg and in the case of a failure the entire XML cdr will
>>> be set into a var on A leg.
>>>
>>> for instance if you set failed_xml_cdr_prefix=foo you would get foo_1
>>> foo_2 etc depending on the number of outbound call legs.
>>>
>>> Additionally you can set copy_xml_cdr on the A leg and when the bridge
>>> ends you will get a complete CDR for B in the "b_leg_cdr" variable on
>>> A
>>>
>>> Finally you should really go with the flow of how FreeSWITCH is
>>> engineered and try to keep your accounting logic in a separate place
>>> and monitor the XML-CDR, CDR-CSV or event_socket +
>>> CHANNEL_HANGUP_COMPLETE events to process this information.? There is
>>> much more to a call and what happens when it's transferred etc than
>>> what you can get in a single monolithic perspective of inside the
>>> channel.? Plus it's conter intuitive to put routing, application and
>>> accounting logic in the same place.
>>>
>>>
>>>
>>> On Tue, Aug 16, 2011 at 12:26 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>>> Dmitry,
>>>> Yes, I agree. For an novice like me it was extremely confusing and
>>>> frustrating that I could not get the proper B leg disposition when A leg
>>>> is
>>>> answered by FS itself.
>>>> I hope there can be some methods or functions that will allow us to get
>>>> the
>>>> proper B Leg disposition.
>>>> ________________________________
>>>> From: Dmitry Sytchev <kbdfck at gmail.com>
>>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>> Sent: Tuesday, August 16, 2011 2:30 AM
>>>> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>>>
>>>> Seems we need to clear things about B-leg disposition in wiki.
>>>> AFAIK there is no method to get correct disposition from B leg without
>>>> analyzing events in case when A-leg was answered by FS itself.
>>>> If A leg was not answered, we can use A-leg disposition for call
>>>> disposition.
>>>>
>>>> 2011/8/16 Michael Collins <msc at freeswitch.org>
>>>>
>>>>
>>>> On Mon, Aug 15, 2011 at 5:05 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>>>
>>>> I am using perl's $session->get_variable("originate_disposition");
>>>>
>>>> Are you looking at the b-leg's session?
>>>>
>>>>
>>>> Also, how come the "hangup_time" shows zero on answered calls?
>>>>
>>>> Because hangup_time refers to the point in time at which the call was
>>>> hung
>>>> up. Since you are in the middle of a call (using the $session object)
>>>> you
>>>> will never see the hangup_time because the object ceases to exist once
>>>> the
>>>> call leg is disconnected.
>>>> I get the impression that you may be using the wrong tool for this
>>>> particular job, but I'm not sure without seeing it. If you don't mind
>>>> dropping it on pastebin we'll have a look and give you some suggestions.
>>>> -MC
>>>>
>>>> ________________________________
>>>> From: Michael Collins <msc at freeswitch.org>
>>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>> Sent: Monday, August 15, 2011 4:58 PM
>>>> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>>>
>>>> And how are you checking the variable? Do you have an event socket open
>>>> or
>>>> ... ?
>>>> -MC
>>>>
>>>> On Mon, Aug 15, 2011 at 4:57 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>>>
>>>> It is being generated with a bridge.
>>>>
>>>> ________________________________
>>>> From: Michael Collins <msc at freeswitch.org>
>>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>> Sent: Monday, August 15, 2011 4:54 PM
>>>> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>>>
>>>> And how is it being generated? WIth a bridge or originate or ... ?
>>>> -MC
>>>>
>>>> On Mon, Aug 15, 2011 at 4:49 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>>>
>>>> I want to know the B-leg status of the call.
>>>>
>>>> ________________________________
>>>> From: Michael Collins <msc at freeswitch.org>
>>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>> Sent: Monday, August 15, 2011 4:31 PM
>>>> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>>>
>>>> I think you may be wanting "endpoint_disposition" depending on exactly
>>>> what
>>>> you're looking at.
>>>> -MC
>>>>
>>>> On Mon, Aug 15, 2011 at 3:35 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>>>
>>>> For the ORIGINATE_DISPOSITION channel variable, does "SUCCESS" mean the
>>>> call
>>>> was ANSWERED?
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Best regards,
>>>>
>>>> Dmitry Sytchev,
>>>> IT Engineer
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> 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
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> 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
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110816/8462c722/attachment-0001.html 

From anthony.minessale at gmail.com  Wed Aug 17 01:41:02 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Aug 2011 16:41:02 -0500
Subject: [Freeswitch-users] ORIGINATE_DISPOSITION
In-Reply-To: <1313529698.72008.YahooMailNeo@web161004.mail.bf1.yahoo.com>
References: <1313447714.89555.YahooMailNeo@web161001.mail.bf1.yahoo.com>
	<CAKzWOxVwDaxsFJAr1dVP5nfmDuFXK2tptvai+oqtwZgph3u1kw@mail.gmail.com>
	<1313452154.28177.YahooMailNeo@web161001.mail.bf1.yahoo.com>
	<CAKzWOxWYXy5Sa2g+cCDEDcxWwE10OtrMaJzUyoLdFSDGtDfbyg@mail.gmail.com>
	<1313452622.11918.YahooMailNeo@web161012.mail.bf1.yahoo.com>
	<CAKzWOxVREFK=uzYzT95oXFT0-sURz7yeOwkuQMz6LKjQ6GvoXA@mail.gmail.com>
	<1313453126.47357.YahooMailNeo@web161010.mail.bf1.yahoo.com>
	<CAKzWOxWtxNoto1c6Chu5dZLPkt+ATcwCGpHOYtPpWydUrcCFfQ@mail.gmail.com>
	<CA+GmxbZ2RG+RS79F8yjXAR4eeFwFy3EMV0k4YiESwq3qMCbwkA@mail.gmail.com>
	<1313515585.77893.YahooMailNeo@web161019.mail.bf1.yahoo.com>
	<CAKbxfG9N1TUmT=HZQqhJfkLk_YiKc=XPqbBoa-9Rm=+Q97PZJw@mail.gmail.com>
	<1313520491.44910.YahooMailNeo@web161015.mail.bf1.yahoo.com>
	<CAKbxfG-3UDi59h3VcOm7+3+OXgaaX1fjNRLZiePrjVc6Ba10Cw@mail.gmail.com>
	<1313522335.98043.YahooMailNeo@web161008.mail.bf1.yahoo.com>
	<CAKbxfG92amS7=q4ex-s7xUJpxHu3QPW6cfdGZ29kgOA0hDU3HA@mail.gmail.com>
	<1313524627.43646.YahooMailNeo@web161009.mail.bf1.yahoo.com>
	<CAKbxfG9B=5cC00XJ24=UTn-k97whgd262FFH8mCFUWPxNjTRow@mail.gmail.com>
	<1313529698.72008.YahooMailNeo@web161004.mail.bf1.yahoo.com>
Message-ID: <CAKbxfG_XY+UvkVtAr7udVRC6XS=TwPeo3K8_uUGVaBmNb+rfsA@mail.gmail.com>

I think you are still missing the point if you are calling the get
XMLCDR on the A leg.


The xml cdr is dumped to the disk by FS, the B-LEG is an entirely
different channel from the one you are in when you call those
functions.  I think I put in a lengthy post this morning explaining a
few ways to get B's CDR from A but you cannot get A's CDR from A while
the call is still active because if the call was not active anymore
your script will be exited and there is no way to get the time.

you just use mod_xml_cdr and/or the event_socket +
channel_hangup_complete event (there is even a param to get an xml_cdr
in that event if you want both)


On Tue, Aug 16, 2011 at 4:21 PM, Sam <lakersman2006 at yahoo.com> wrote:
> I have tried to use the xml cdr to find the call duration, etc. but in the
> xml cdr, freeswitch does not return any of the duration fields, here is a
> pastebin of my xml output when using the perl $session->getXMLCDR(); and yes
> I have also set "process_cdr=b_only".
> http://pastebin.freeswitch.org/17058
>
> ________________________________
> From: Anthony Minessale <anthony.minessale at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Sent: Tuesday, August 16, 2011 1:26 PM
> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>
> again because in FreeSWITCH the legs are separate entity, if you
> listen for hangup_complete events or log B-legs in your cdrs you will
> get precise info about that.? if you really want you can add a
> dialplan app to capture a timestamp when the bridge ends to save the
> time.? This is of-couse only useful if the A leg does not also hangup
> which is why it's best to extract this kind of data from the CDRs.
>
>
> On Tue, Aug 16, 2011 at 2:57 PM, Sam <lakersman2006 at yahoo.com> wrote:
>> Thanks for the clarification. Another question, when the bridge call is
>> answered and hung up, how would I get the call duration or the time the
>> call
>> is hung up? I saw that the channel variable "hangup_time" is 0 on an
>> answered call.
>>
>> ________________________________
>> From: Anthony Minessale <anthony.minessale at gmail.com>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Sent: Tuesday, August 16, 2011 12:24 PM
>> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>
>> that is what bridge_early_media=true means, so you can hear the early
>> media during originate but stil not return until the channel is
>> answered or hungup.
>>
>> On Tue, Aug 16, 2011 at 2:18 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>> I need to be able to play back early media to A leg, so if
>>> {ignore_early_media=false} there is no way to tell from
>>> originate_disposition that the bridge was answered?
>>>
>>> ________________________________
>>> From: Anthony Minessale <anthony.minessale at gmail.com>
>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> Sent: Tuesday, August 16, 2011 11:56 AM
>>> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>>
>>> bridge_early_media implies ignore_early_media true so you can't use
>>> them together but yes since there is no cause code for answered
>>> because the call has not ended that would guarantee it was answered if
>>> you exited originate with either ignore_early_media=true or
>>> bridge_early_media=true
>>>
>>> On Tue, Aug 16, 2011 at 1:48 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>>> "if you call somewhere that uses early media, the originate will end
>>>> setting originate_disposition to SUCCESS meaning that a live channel
>>>> was produced.? Now the bridge will begin between the A and B leg."
>>>> Does that mean it is safe to assume when "originate_disposition" is
>>>> SUCCESS,
>>>> that the call has been answered if I set the following:
>>>>
>>>> {ignore_early_media=false}
>>>> {bridge_early_media=true}
>>>>
>>>>
>>>> ________________________________
>>>> From: Anthony Minessale <anthony.minessale at gmail.com>
>>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>> Sent: Tuesday, August 16, 2011 10:54 AM
>>>> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>>>
>>>> The originate disposition of A leg will always have the last known
>>>> status
>>>> from B
>>>>
>>>> originate, by default, returns when:
>>>>
>>>> 1) media is established on a specific outgoing leg, answered or not
>>>> 2) all outgoing legs are terminated.
>>>>
>>>> if you call somewhere that uses early media, the originate will end
>>>> setting originate_disposition to SUCCESS meaning that a live channel
>>>> was produced.? Now the bridge will begin between the A and B leg.
>>>>
>>>> When the bridge ends if the B leg is hungup, its cause will be stored
>>>> in "bridge_hangup_cause"
>>>>
>>>> if you want to keep the originate from ending when early media is
>>>> established you can add {ignore_early_media=true} or
>>>> {bridge_early_media=true} prepended to your dial string and the
>>>> originate will never return untill all outbound legs are either hungup
>>>> or one is answered.
>>>>
>>>>
>>>> You should take some time to expand your mind to the different
>>>> paradigm in FreeSWITCH where you may have as many as 10 outbound legs
>>>> at once in a forked-dial situation and some of what you think is
>>>> simple and obvious will quickly dissolve.
>>>>
>>>> Another thing you can do is set the variable "failed_xml_cdr_prefix"
>>>> on the A leg.
>>>> This prefix will be mixed with an incrementing variable for each
>>>> outbound call leg and in the case of a failure the entire XML cdr will
>>>> be set into a var on A leg.
>>>>
>>>> for instance if you set failed_xml_cdr_prefix=foo you would get foo_1
>>>> foo_2 etc depending on the number of outbound call legs.
>>>>
>>>> Additionally you can set copy_xml_cdr on the A leg and when the bridge
>>>> ends you will get a complete CDR for B in the "b_leg_cdr" variable on
>>>> A
>>>>
>>>> Finally you should really go with the flow of how FreeSWITCH is
>>>> engineered and try to keep your accounting logic in a separate place
>>>> and monitor the XML-CDR, CDR-CSV or event_socket +
>>>> CHANNEL_HANGUP_COMPLETE events to process this information.? There is
>>>> much more to a call and what happens when it's transferred etc than
>>>> what you can get in a single monolithic perspective of inside the
>>>> channel.? Plus it's conter intuitive to put routing, application and
>>>> accounting logic in the same place.
>>>>
>>>>
>>>>
>>>> On Tue, Aug 16, 2011 at 12:26 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>>>> Dmitry,
>>>>> Yes, I agree. For an novice like me it was extremely confusing and
>>>>> frustrating that I could not get the proper B leg disposition when A
>>>>> leg
>>>>> is
>>>>> answered by FS itself.
>>>>> I hope there can be some methods or functions that will allow us to get
>>>>> the
>>>>> proper B Leg disposition.
>>>>> ________________________________
>>>>> From: Dmitry Sytchev <kbdfck at gmail.com>
>>>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>>> Sent: Tuesday, August 16, 2011 2:30 AM
>>>>> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>>>>
>>>>> Seems we need to clear things about B-leg disposition in wiki.
>>>>> AFAIK there is no method to get correct disposition from B leg without
>>>>> analyzing events in case when A-leg was answered by FS itself.
>>>>> If A leg was not answered, we can use A-leg disposition for call
>>>>> disposition.
>>>>>
>>>>> 2011/8/16 Michael Collins <msc at freeswitch.org>
>>>>>
>>>>>
>>>>> On Mon, Aug 15, 2011 at 5:05 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>>>>
>>>>> I am using perl's $session->get_variable("originate_disposition");
>>>>>
>>>>> Are you looking at the b-leg's session?
>>>>>
>>>>>
>>>>> Also, how come the "hangup_time" shows zero on answered calls?
>>>>>
>>>>> Because hangup_time refers to the point in time at which the call was
>>>>> hung
>>>>> up. Since you are in the middle of a call (using the $session object)
>>>>> you
>>>>> will never see the hangup_time because the object ceases to exist once
>>>>> the
>>>>> call leg is disconnected.
>>>>> I get the impression that you may be using the wrong tool for this
>>>>> particular job, but I'm not sure without seeing it. If you don't mind
>>>>> dropping it on pastebin we'll have a look and give you some
>>>>> suggestions.
>>>>> -MC
>>>>>
>>>>> ________________________________
>>>>> From: Michael Collins <msc at freeswitch.org>
>>>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>>> Sent: Monday, August 15, 2011 4:58 PM
>>>>> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>>>>
>>>>> And how are you checking the variable? Do you have an event socket open
>>>>> or
>>>>> ... ?
>>>>> -MC
>>>>>
>>>>> On Mon, Aug 15, 2011 at 4:57 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>>>>
>>>>> It is being generated with a bridge.
>>>>>
>>>>> ________________________________
>>>>> From: Michael Collins <msc at freeswitch.org>
>>>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>>> Sent: Monday, August 15, 2011 4:54 PM
>>>>> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>>>>
>>>>> And how is it being generated? WIth a bridge or originate or ... ?
>>>>> -MC
>>>>>
>>>>> On Mon, Aug 15, 2011 at 4:49 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>>>>
>>>>> I want to know the B-leg status of the call.
>>>>>
>>>>> ________________________________
>>>>> From: Michael Collins <msc at freeswitch.org>
>>>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>>> Sent: Monday, August 15, 2011 4:31 PM
>>>>> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>>>>
>>>>> I think you may be wanting "endpoint_disposition" depending on exactly
>>>>> what
>>>>> you're looking at.
>>>>> -MC
>>>>>
>>>>> On Mon, Aug 15, 2011 at 3:35 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>>>>
>>>>> For the ORIGINATE_DISPOSITION channel variable, does "SUCCESS" mean the
>>>>> call
>>>>> was ANSWERED?
>>>>>
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Best regards,
>>>>>
>>>>> Dmitry Sytchev,
>>>>> IT Engineer
>>>>>
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> 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
>>>> googletalk:conf+888 at conference.freeswitch.org
>>>> pstn:+19193869900
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> 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
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> 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
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From mgg at giagnocavo.net  Wed Aug 17 01:48:43 2011
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Tue, 16 Aug 2011 17:48:43 -0400
Subject: [Freeswitch-users] ORIGINATE_DISPOSITION
In-Reply-To: <1313529698.72008.YahooMailNeo@web161004.mail.bf1.yahoo.com>
References: <1313447714.89555.YahooMailNeo@web161001.mail.bf1.yahoo.com>
	<CAKzWOxVwDaxsFJAr1dVP5nfmDuFXK2tptvai+oqtwZgph3u1kw@mail.gmail.com>
	<1313452154.28177.YahooMailNeo@web161001.mail.bf1.yahoo.com>
	<CAKzWOxWYXy5Sa2g+cCDEDcxWwE10OtrMaJzUyoLdFSDGtDfbyg@mail.gmail.com>
	<1313452622.11918.YahooMailNeo@web161012.mail.bf1.yahoo.com>
	<CAKzWOxVREFK=uzYzT95oXFT0-sURz7yeOwkuQMz6LKjQ6GvoXA@mail.gmail.com>
	<1313453126.47357.YahooMailNeo@web161010.mail.bf1.yahoo.com>
	<CAKzWOxWtxNoto1c6Chu5dZLPkt+ATcwCGpHOYtPpWydUrcCFfQ@mail.gmail.com>
	<CA+GmxbZ2RG+RS79F8yjXAR4eeFwFy3EMV0k4YiESwq3qMCbwkA@mail.gmail.com>
	<1313515585.77893.YahooMailNeo@web161019.mail.bf1.yahoo.com>
	<CAKbxfG9N1TUmT=HZQqhJfkLk_YiKc=XPqbBoa-9Rm=+Q97PZJw@mail.gmail.com>
	<1313520491.44910.YahooMailNeo@web161015.mail.bf1.yahoo.com>
	<CAKbxfG-3UDi59h3VcOm7+3+OXgaaX1fjNRLZiePrjVc6Ba10Cw@mail.gmail.com>
	<1313522335.98043.YahooMailNeo@web161008.mail.bf1.yahoo.com>
	<CAKbxfG92amS7=q4ex-s7xUJpxHu3QPW6cfdGZ29kgOA0hDU3HA@mail.gmail.com>
	<1313524627.43646.YahooMailNeo@web161009.mail.bf1.yahoo.com>
	<CAKbxfG9B=5cC00XJ24=UTn-k97whgd262FFH8mCFUWPxNjTRow@mail.gmail.com>
	<1313529698.72008.YahooMailNeo@web161004.mail.bf1.yahoo.com>
Message-ID: <03351FCC6082174C8534AB714B8258A5E416BC2E@mse17be1.mse17.exchange.ms>

I'm gonna jump in with some non-specific advice that might save you a lot of hassle. Go back to the drawing board and work really hard on creating a design that does not involve getting CDR information while processing any of the call legs. Set custom variables, and pick up the a_leg CDR file, and get the b-legs from there. Anthony and I spent many hours going back and forth on this (well, more like he kindly indulged my insistence on doing it my way) and after a long time and lots of obscure edge cases making things unpleasant, I rewrote it and haven't had a problem since.

The FS code might have some of the bugs fixed, but it's not a recommend nor very well tested scenario to try to get CDR info while a leg is still going, so just save yourself the pain.

-Michael

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sam
Sent: Tuesday, August 16, 2011 3:22 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION

I have tried to use the xml cdr to find the call duration, etc. but in the xml cdr, freeswitch does not return any of the duration fields, here is a pastebin of my xml output when using the perl $session->getXMLCDR(); and yes I have also set "process_cdr=b_only".

http://pastebin.freeswitch.org/17058

________________________________
From: Anthony Minessale <anthony.minessale at gmail.com<mailto:anthony.minessale at gmail.com>>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
Sent: Tuesday, August 16, 2011 1:26 PM
Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION

again because in FreeSWITCH the legs are separate entity, if you
listen for hangup_complete events or log B-legs in your cdrs you will
get precise info about that.  if you really want you can add a
dialplan app to capture a timestamp when the bridge ends to save the
time.  This is of-couse only useful if the A leg does not also hangup
which is why it's best to extract this kind of data from the CDRs.


On Tue, Aug 16, 2011 at 2:57 PM, Sam <lakersman2006 at yahoo.com<mailto:lakersman2006 at yahoo.com>> wrote:
> Thanks for the clarification. Another question, when the bridge call is
> answered and hung up, how would I get the call duration or the time the call
> is hung up? I saw that the channel variable "hangup_time" is 0 on an
> answered call.
>
> ________________________________
> From: Anthony Minessale <anthony.minessale at gmail.com<mailto:anthony.minessale at gmail.com>>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
> Sent: Tuesday, August 16, 2011 12:24 PM
> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>
> that is what bridge_early_media=true means, so you can hear the early
> media during originate but stil not return until the channel is
> answered or hungup.
>
> On Tue, Aug 16, 2011 at 2:18 PM, Sam <lakersman2006 at yahoo.com<mailto:lakersman2006 at yahoo.com>> wrote:
>> I need to be able to play back early media to A leg, so if
>> {ignore_early_media=false} there is no way to tell from
>> originate_disposition that the bridge was answered?
>>
>> ________________________________
>> From: Anthony Minessale <anthony.minessale at gmail.com<mailto:anthony.minessale at gmail.com>>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
>> Sent: Tuesday, August 16, 2011 11:56 AM
>> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>
>> bridge_early_media implies ignore_early_media true so you can't use
>> them together but yes since there is no cause code for answered
>> because the call has not ended that would guarantee it was answered if
>> you exited originate with either ignore_early_media=true or
>> bridge_early_media=true
>>
>> On Tue, Aug 16, 2011 at 1:48 PM, Sam <lakersman2006 at yahoo.com<mailto:lakersman2006 at yahoo.com>> wrote:
>>> "if you call somewhere that uses early media, the originate will end
>>> setting originate_disposition to SUCCESS meaning that a live channel
>>> was produced.  Now the bridge will begin between the A and B leg."
>>> Does that mean it is safe to assume when "originate_disposition" is
>>> SUCCESS,
>>> that the call has been answered if I set the following:
>>>
>>> {ignore_early_media=false}
>>> {bridge_early_media=true}
>>>
>>>
>>> ________________________________
>>> From: Anthony Minessale <anthony.minessale at gmail.com<mailto:anthony.minessale at gmail.com>>
>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
>>> Sent: Tuesday, August 16, 2011 10:54 AM
>>> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>>
>>> The originate disposition of A leg will always have the last known status
>>> from B
>>>
>>> originate, by default, returns when:
>>>
>>> 1) media is established on a specific outgoing leg, answered or not
>>> 2) all outgoing legs are terminated.
>>>
>>> if you call somewhere that uses early media, the originate will end
>>> setting originate_disposition to SUCCESS meaning that a live channel
>>> was produced.  Now the bridge will begin between the A and B leg.
>>>
>>> When the bridge ends if the B leg is hungup, its cause will be stored
>>> in "bridge_hangup_cause"
>>>
>>> if you want to keep the originate from ending when early media is
>>> established you can add {ignore_early_media=true} or
>>> {bridge_early_media=true} prepended to your dial string and the
>>> originate will never return untill all outbound legs are either hungup
>>> or one is answered.
>>>
>>>
>>> You should take some time to expand your mind to the different
>>> paradigm in FreeSWITCH where you may have as many as 10 outbound legs
>>> at once in a forked-dial situation and some of what you think is
>>> simple and obvious will quickly dissolve.
>>>
>>> Another thing you can do is set the variable "failed_xml_cdr_prefix"
>>> on the A leg.
>>> This prefix will be mixed with an incrementing variable for each
>>> outbound call leg and in the case of a failure the entire XML cdr will
>>> be set into a var on A leg.
>>>
>>> for instance if you set failed_xml_cdr_prefix=foo you would get foo_1
>>> foo_2 etc depending on the number of outbound call legs.
>>>
>>> Additionally you can set copy_xml_cdr on the A leg and when the bridge
>>> ends you will get a complete CDR for B in the "b_leg_cdr" variable on
>>> A
>>>
>>> Finally you should really go with the flow of how FreeSWITCH is
>>> engineered and try to keep your accounting logic in a separate place
>>> and monitor the XML-CDR, CDR-CSV or event_socket +
>>> CHANNEL_HANGUP_COMPLETE events to process this information.  There is
>>> much more to a call and what happens when it's transferred etc than
>>> what you can get in a single monolithic perspective of inside the
>>> channel.  Plus it's conter intuitive to put routing, application and
>>> accounting logic in the same place.
>>>
>>>
>>>
>>> On Tue, Aug 16, 2011 at 12:26 PM, Sam <lakersman2006 at yahoo.com<mailto:lakersman2006 at yahoo.com>> wrote:
>>>> Dmitry,
>>>> Yes, I agree. For an novice like me it was extremely confusing and
>>>> frustrating that I could not get the proper B leg disposition when A leg
>>>> is
>>>> answered by FS itself.
>>>> I hope there can be some methods or functions that will allow us to get
>>>> the
>>>> proper B Leg disposition.
>>>> ________________________________
>>>> From: Dmitry Sytchev <kbdfck at gmail.com<mailto:kbdfck at gmail.com>>
>>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
>>>> Sent: Tuesday, August 16, 2011 2:30 AM
>>>> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>>>
>>>> Seems we need to clear things about B-leg disposition in wiki.
>>>> AFAIK there is no method to get correct disposition from B leg without
>>>> analyzing events in case when A-leg was answered by FS itself.
>>>> If A leg was not answered, we can use A-leg disposition for call
>>>> disposition.
>>>>
>>>> 2011/8/16 Michael Collins <msc at freeswitch.org<mailto:msc at freeswitch.org>>
>>>>
>>>>
>>>> On Mon, Aug 15, 2011 at 5:05 PM, Sam <lakersman2006 at yahoo.com<mailto:lakersman2006 at yahoo.com>> wrote:
>>>>
>>>> I am using perl's $session->get_variable("originate_disposition");
>>>>
>>>> Are you looking at the b-leg's session?
>>>>
>>>>
>>>> Also, how come the "hangup_time" shows zero on answered calls?
>>>>
>>>> Because hangup_time refers to the point in time at which the call was
>>>> hung
>>>> up. Since you are in the middle of a call (using the $session object)
>>>> you
>>>> will never see the hangup_time because the object ceases to exist once
>>>> the
>>>> call leg is disconnected.
>>>> I get the impression that you may be using the wrong tool for this
>>>> particular job, but I'm not sure without seeing it. If you don't mind
>>>> dropping it on pastebin we'll have a look and give you some suggestions.
>>>> -MC
>>>>
>>>> ________________________________
>>>> From: Michael Collins <msc at freeswitch.org<mailto:msc at freeswitch.org>>
>>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
>>>> Sent: Monday, August 15, 2011 4:58 PM
>>>> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>>>
>>>> And how are you checking the variable? Do you have an event socket open
>>>> or
>>>> ... ?
>>>> -MC
>>>>
>>>> On Mon, Aug 15, 2011 at 4:57 PM, Sam <lakersman2006 at yahoo.com<mailto:lakersman2006 at yahoo.com>> wrote:
>>>>
>>>> It is being generated with a bridge.
>>>>
>>>> ________________________________
>>>> From: Michael Collins <msc at freeswitch.org<mailto:msc at freeswitch.org>>
>>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
>>>> Sent: Monday, August 15, 2011 4:54 PM
>>>> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>>>
>>>> And how is it being generated? WIth a bridge or originate or ... ?
>>>> -MC
>>>>
>>>> On Mon, Aug 15, 2011 at 4:49 PM, Sam <lakersman2006 at yahoo.com<mailto:lakersman2006 at yahoo.com>> wrote:
>>>>
>>>> I want to know the B-leg status of the call.
>>>>
>>>> ________________________________
>>>> From: Michael Collins <msc at freeswitch.org<mailto:msc at freeswitch.org>>
>>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>>
>>>> Sent: Monday, August 15, 2011 4:31 PM
>>>> Subject: Re: [Freeswitch-users] ORIGINATE_DISPOSITION
>>>>
>>>> I think you may be wanting "endpoint_disposition" depending on exactly
>>>> what
>>>> you're looking at.
>>>> -MC
>>>>
>>>> On Mon, Aug 15, 2011 at 3:35 PM, Sam <lakersman2006 at yahoo.com<mailto:lakersman2006 at yahoo.com>> wrote:
>>>>
>>>> For the ORIGINATE_DISPOSITION channel variable, does "SUCCESS" mean the
>>>> call
>>>> was ANSWERED?
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Best regards,
>>>>
>>>> Dmitry Sytchev,
>>>> IT Engineer
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> 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:anthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<mailto:anthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org<mailto:888 at conference.freeswitch.org>
>>> googletalk:conf+888 at conference.freeswitch.org<mailto:888 at conference.freeswitch.org>
>>> pstn:+19193869900
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> 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:anthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<mailto:anthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org<mailto:888 at conference.freeswitch.org>
>> googletalk:conf+888 at conference.freeswitch.org<mailto:888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> Anthony Minessale II
>
> 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:anthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<mailto:anthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org<mailto:888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<mailto:888 at conference.freeswitch.org>
> pstn:+19193869900
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



--
Anthony Minessale II

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:anthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<mailto:anthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org<mailto:888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<mailto:888 at conference.freeswitch.org>
pstn:+19193869900


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

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110816/2a3670f8/attachment-0001.html 

From db3l.net at gmail.com  Wed Aug 17 01:55:43 2011
From: db3l.net at gmail.com (David Bolen)
Date: Tue, 16 Aug 2011 17:55:43 -0400
Subject: [Freeswitch-users] DTMF bleed through in conference
References: <m2liuuz3y9.fsf@valheru.db3l.homeip.net>
	<CAKzWOxU_oaWg1_c=92W30g-pZyWkjWWW2YV5bh_hG-r1mZdVZA@mail.gmail.com>
	<CAKDfjgf2bYTDPsFaoM3ubvu=NGoPxN-tSJR0aCo7NZyRcmAqFw@mail.gmail.com>
Message-ID: <m2hb5hyqao.fsf@valheru.db3l.homeip.net>

Kristian Kielhofner <kris at kriskinc.com> writes:

> Agreed.  Sounds like an analog device not clamping inband DTMF -> RFC
> 2833 events properly (or quickly enough).
>
> Grab a pcap and look at the audio.  See if you can see DTMF in the
> audio stream itself.

Thanks.  I do believe the problem is while tones are using RFC2833.  I
did some separate in-band testing (manually configuring the PAP2T for
in-band and using start_dtmf on the switch) and in that case full
tones go through to the conference while also being recognized (though
in that case by switch_ivr_async, not switch_rcp).  What I hear in the
2833 case is a more truncated/clipped "blip" of sound.

I just ran a specific test with two conference participants - blink on
a Mac (software sipphone), and an analog phone through a PAP2T.  Both
connections direct to the FreeSWITCH server, no PSTN gateway involved.
I used blink as the DTMF generator.

I pressed "4" 4 times in blink, 3 of which (cases 1,2 and 4) were
audible blips to the other phone.  A summary of the blink outbound rtp
is at the bottom.  It seems to have the c101 types properly for the
presses.  Not sure if the occasional repeated sequence number is an
issue.  The pcap for the outbound leg is entirely c0 rtp types.

I used wireshark to generate audio files from the inbound blink leg
and the outbound PAP2T leg of the call and verified that the inbound
leg from blink to FreeSwitch was silent, while you can clearly hear
the brief blips of tones on the outbound leg.  So whatever is
happening (even if not FreeSWITCH but networking or some other host
issue) doesn't appear to be the end device(s).

-- David

Inbound packets (sipphone -> FreeSWITCH):
  17:23:24.954474 udp/rtp 160 c0  6611 197118243 1533652571
  17:23:24.973732 udp/rtp 160 c0  6612 197118403 1533652571
  17:23:25.053559 udp/rtp 4 c101 * 6613 197118723 1533652571
  17:23:25.053588 udp/rtp 4 c101 * 6613 197118723 1533652571
  17:23:25.053594 udp/rtp 4 c101 * 6613 197118723 1533652571
  17:23:25.098159 udp/rtp 4 c101 * 6614 197118723 1533652571
  17:23:25.098389 udp/rtp 4 c101 * 6615 197118723 1533652571
  17:23:25.114263 udp/rtp 4 c101 * 6616 197118723 1533652571
  17:23:25.114377 udp/rtp 4 c101 * 6617 197118723 1533652571
  17:23:25.114392 udp/rtp 4 c101 * 6617 197118723 1533652571
  17:23:25.114402 udp/rtp 4 c101 * 6617 197118723 1533652571
  17:23:25.127360 udp/rtp 160 c0  6618 197119763 1533652571
  17:23:25.142563 udp/rtp 160 c0  6619 197119923 1533652571
  17:23:25.157877 udp/rtp 160 c0  6620 197120083 1533652571
  17:23:25.241723 udp/rtp 4 c101 * 6621 197120403 1533652571
  17:23:25.241751 udp/rtp 4 c101 * 6621 197120403 1533652571
  17:23:25.241766 udp/rtp 4 c101 * 6621 197120403 1533652571
  17:23:25.260182 udp/rtp 4 c101 * 6622 197120403 1533652571
  17:23:25.313686 udp/rtp 4 c101 * 6623 197120403 1533652571
  17:23:25.313708 udp/rtp 4 c101 * 6624 197120403 1533652571
  17:23:25.313721 udp/rtp 4 c101 * 6624 197120403 1533652571
  17:23:25.313728 udp/rtp 4 c101 * 6624 197120403 1533652571
  17:23:25.320545 udp/rtp 160 c0  6625 197121363 1533652571
  17:23:25.325668 udp/rtp 160 c0  6626 197121523 1533652571
  17:23:25.355376 udp/rtp 160 c0  6627 197121683 1533652571
  17:23:25.360447 udp/rtp 160 c0  6628 197121843 1533652571
  17:23:25.385616 udp/rtp 160 c0  6629 197122003 1533652571
  17:23:25.486738 udp/rtp 4 c101 * 6630 197122243 1533652571
  17:23:25.486777 udp/rtp 4 c101 * 6630 197122243 1533652571
  17:23:25.486826 udp/rtp 4 c101 * 6630 197122243 1533652571
  17:23:25.486921 udp/rtp 4 c101 * 6631 197122243 1533652571
  17:23:25.492201 udp/rtp 4 c101 * 6632 197122243 1533652571
  17:23:25.534484 udp/rtp 4 c101 * 6633 197122243 1533652571
  17:23:25.534505 udp/rtp 4 c101 * 6634 197122243 1533652571
  17:23:25.534514 udp/rtp 4 c101 * 6634 197122243 1533652571
  17:23:25.534519 udp/rtp 4 c101 * 6634 197122243 1533652571
  17:23:25.540002 udp/rtp 160 c0  6635 197123283 1533652571
  17:23:25.540155 udp/rtp 160 c0  6636 197123443 1533652571
  17:23:25.580041 udp/rtp 160 c0  6637 197123603 1533652571
  17:23:25.640075 udp/rtp 4 c101 * 6638 197123923 1533652571
  17:23:25.640106 udp/rtp 4 c101 * 6638 197123923 1533652571
  17:23:25.646655 udp/rtp 4 c101 * 6638 197123923 1533652571
  17:23:25.705011 udp/rtp 4 c101 * 6639 197123923 1533652571
  17:23:25.710230 udp/rtp 4 c101 * 6640 197123923 1533652571
  17:23:25.710259 udp/rtp 4 c101 * 6641 197123923 1533652571
  17:23:25.710286 udp/rtp 4 c101 * 6641 197123923 1533652571
  17:23:25.710301 udp/rtp 4 c101 * 6641 197123923 1533652571
  17:23:25.710313 udp/rtp 160 c0  6642 197124883 1533652571
  17:23:25.723594 udp/rtp 160 c0  6643 197125043 1533652571
  17:23:25.745100 udp/rtp 160 c0  6644 197125203 1533652571
  17:23:25.762744 udp/rtp 160 c0  6645 197125363 1533652571



From avi at avimarcus.net  Wed Aug 17 02:01:13 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 17 Aug 2011 01:01:13 +0300
Subject: [Freeswitch-users] DTMF bleed through in conference
In-Reply-To: <m2hb5hyqao.fsf@valheru.db3l.homeip.net>
References: <m2liuuz3y9.fsf@valheru.db3l.homeip.net>
	<CAKzWOxU_oaWg1_c=92W30g-pZyWkjWWW2YV5bh_hG-r1mZdVZA@mail.gmail.com>
	<CAKDfjgf2bYTDPsFaoM3ubvu=NGoPxN-tSJR0aCo7NZyRcmAqFw@mail.gmail.com>
	<m2hb5hyqao.fsf@valheru.db3l.homeip.net>
Message-ID: <CALNZuwm=zsnmMwTfQbaruAZANg-FsEA5s9wnK_48iq3fcvPFaw@mail.gmail.com>

Could this be a keypad sound that gets picked up by the microphone?
-Avi


On Wed, Aug 17, 2011 at 12:55 AM, David Bolen <db3l.net at gmail.com> wrote:

> Kristian Kielhofner <kris at kriskinc.com> writes:
>
> > Agreed.  Sounds like an analog device not clamping inband DTMF -> RFC
> > 2833 events properly (or quickly enough).
> >
> > Grab a pcap and look at the audio.  See if you can see DTMF in the
> > audio stream itself.
>
> Thanks.  I do believe the problem is while tones are using RFC2833.  I
> did some separate in-band testing (manually configuring the PAP2T for
> in-band and using start_dtmf on the switch) and in that case full
> tones go through to the conference while also being recognized (though
> in that case by switch_ivr_async, not switch_rcp).  What I hear in the
> 2833 case is a more truncated/clipped "blip" of sound.
>
> I just ran a specific test with two conference participants - blink on
> a Mac (software sipphone), and an analog phone through a PAP2T.  Both
> connections direct to the FreeSWITCH server, no PSTN gateway involved.
> I used blink as the DTMF generator.
>
> I pressed "4" 4 times in blink, 3 of which (cases 1,2 and 4) were
> audible blips to the other phone.  A summary of the blink outbound rtp
> is at the bottom.  It seems to have the c101 types properly for the
> presses.  Not sure if the occasional repeated sequence number is an
> issue.  The pcap for the outbound leg is entirely c0 rtp types.
>
> I used wireshark to generate audio files from the inbound blink leg
> and the outbound PAP2T leg of the call and verified that the inbound
> leg from blink to FreeSwitch was silent, while you can clearly hear
> the brief blips of tones on the outbound leg.  So whatever is
> happening (even if not FreeSWITCH but networking or some other host
> issue) doesn't appear to be the end device(s).
>
> -- David
>
> Inbound packets (sipphone -> FreeSWITCH):
>  17:23:24.954474 udp/rtp 160 c0  6611 197118243 1533652571
>  17:23:24.973732 udp/rtp 160 c0  6612 197118403 1533652571
>  17:23:25.053559 udp/rtp 4 c101 * 6613 197118723 1533652571
>  17:23:25.053588 udp/rtp 4 c101 * 6613 197118723 1533652571
>  17:23:25.053594 udp/rtp 4 c101 * 6613 197118723 1533652571
>  17:23:25.098159 udp/rtp 4 c101 * 6614 197118723 1533652571
>  17:23:25.098389 udp/rtp 4 c101 * 6615 197118723 1533652571
>  17:23:25.114263 udp/rtp 4 c101 * 6616 197118723 1533652571
>  17:23:25.114377 udp/rtp 4 c101 * 6617 197118723 1533652571
>  17:23:25.114392 udp/rtp 4 c101 * 6617 197118723 1533652571
>  17:23:25.114402 udp/rtp 4 c101 * 6617 197118723 1533652571
>  17:23:25.127360 udp/rtp 160 c0  6618 197119763 1533652571
>  17:23:25.142563 udp/rtp 160 c0  6619 197119923 1533652571
>  17:23:25.157877 udp/rtp 160 c0  6620 197120083 1533652571
>  17:23:25.241723 udp/rtp 4 c101 * 6621 197120403 1533652571
>  17:23:25.241751 udp/rtp 4 c101 * 6621 197120403 1533652571
>  17:23:25.241766 udp/rtp 4 c101 * 6621 197120403 1533652571
>  17:23:25.260182 udp/rtp 4 c101 * 6622 197120403 1533652571
>  17:23:25.313686 udp/rtp 4 c101 * 6623 197120403 1533652571
>  17:23:25.313708 udp/rtp 4 c101 * 6624 197120403 1533652571
>  17:23:25.313721 udp/rtp 4 c101 * 6624 197120403 1533652571
>  17:23:25.313728 udp/rtp 4 c101 * 6624 197120403 1533652571
>  17:23:25.320545 udp/rtp 160 c0  6625 197121363 1533652571
>  17:23:25.325668 udp/rtp 160 c0  6626 197121523 1533652571
>  17:23:25.355376 udp/rtp 160 c0  6627 197121683 1533652571
>  17:23:25.360447 udp/rtp 160 c0  6628 197121843 1533652571
>  17:23:25.385616 udp/rtp 160 c0  6629 197122003 1533652571
>  17:23:25.486738 udp/rtp 4 c101 * 6630 197122243 1533652571
>  17:23:25.486777 udp/rtp 4 c101 * 6630 197122243 1533652571
>  17:23:25.486826 udp/rtp 4 c101 * 6630 197122243 1533652571
>  17:23:25.486921 udp/rtp 4 c101 * 6631 197122243 1533652571
>  17:23:25.492201 udp/rtp 4 c101 * 6632 197122243 1533652571
>  17:23:25.534484 udp/rtp 4 c101 * 6633 197122243 1533652571
>  17:23:25.534505 udp/rtp 4 c101 * 6634 197122243 1533652571
>  17:23:25.534514 udp/rtp 4 c101 * 6634 197122243 1533652571
>  17:23:25.534519 udp/rtp 4 c101 * 6634 197122243 1533652571
>  17:23:25.540002 udp/rtp 160 c0  6635 197123283 1533652571
>  17:23:25.540155 udp/rtp 160 c0  6636 197123443 1533652571
>  17:23:25.580041 udp/rtp 160 c0  6637 197123603 1533652571
>  17:23:25.640075 udp/rtp 4 c101 * 6638 197123923 1533652571
>  17:23:25.640106 udp/rtp 4 c101 * 6638 197123923 1533652571
>  17:23:25.646655 udp/rtp 4 c101 * 6638 197123923 1533652571
>  17:23:25.705011 udp/rtp 4 c101 * 6639 197123923 1533652571
>  17:23:25.710230 udp/rtp 4 c101 * 6640 197123923 1533652571
>  17:23:25.710259 udp/rtp 4 c101 * 6641 197123923 1533652571
>  17:23:25.710286 udp/rtp 4 c101 * 6641 197123923 1533652571
>  17:23:25.710301 udp/rtp 4 c101 * 6641 197123923 1533652571
>  17:23:25.710313 udp/rtp 160 c0  6642 197124883 1533652571
>  17:23:25.723594 udp/rtp 160 c0  6643 197125043 1533652571
>  17:23:25.745100 udp/rtp 160 c0  6644 197125203 1533652571
>  17:23:25.762744 udp/rtp 160 c0  6645 197125363 1533652571
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/6fcd70e2/attachment.html 

From michal.bielicki at seventhsignal.de  Wed Aug 17 02:22:35 2011
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Wed, 17 Aug 2011 00:22:35 +0200
Subject: [Freeswitch-users] Building Freeswitch + FreeTDM + LibSng-ISDN
	RPMs
In-Reply-To: <CAFeQK0sejfBwWGPX0oqwJ48FT4oygjT7G27GT7tr_9A7kgpVyQ@mail.gmail.com>
References: <CAFeQK0sejfBwWGPX0oqwJ48FT4oygjT7G27GT7tr_9A7kgpVyQ@mail.gmail.com>
Message-ID: <F2633BA2-5F5D-4983-A50E-04CD2D523372@seventhsignal.de>

Just use my rpms :)
repo.seventhsignal.de for the sangoma stuff
repo.freeswitch.de for freeswitch rpms on centos 5 and centos 6

there are also SRPMS for all the stuff there so you can look and analyse the spec files I did for the sangoma stuff.

Am 16.08.2011 um 22:07 schrieb Octavio Ruiz:

> I'm trying to build Freeswitch with full Sangoma's ( FreeTDM +
> LibSng-ISDN ) support from SRPMs for Centos 5, but there are two not
> satisfied dependencies for what looks like non existing RPMs, fact
> that raises many questions on my head,
> 
> An extract from the freeswitch SPEC file:
> 
> -- from freeswitch.spec
> Requires: libsng_isdn
> BuildRequires: wanpipe
> BuildRequires: libsng_isdn
> 
> even if looks quite easy to bypass this and get it to compile
> satisfying those, either creating the missing libsng_isdn SPEC file
> needed for building the package (libsng_isdn is only a bunch of
> headers and an already-build shared object) and modifying the
> wanpipe.rpmspec in order to change the naming behavior which doesn't
> comply with the above (Wanpipe's resulting RPM is named
> 'wanpipe-<kernel-version>' which do not satisfy the solely string
> 'wanpipe') or either worst, in terms of cleanliness, creating dummy
> RPM packages or removing dependencies on the  freeswitch.spec, I
> wonder what is supposed to be the correct environment and proper
> procedure for building Freeswitch and Wanrouter + SNG-ISDN following
> what the SPEC file states.
> 
> -- from wanpipe.rpmspec
> %define KERNEL_VERSION    %{?kern_ver}
> %define WANPIPE_VER       wanpipe
> %define name              %{WANPIPE_VER}
> 
> Am I missing something? Has anyone tested this before?  Someone else
> is packaging Sangoma's Wanrouter on RPM with other naming scheme?  Are
> they on the SPEC file merely to denote upcoming work?
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Michal Bielicki
Gesch?ftsf?hrer / CEO

Seventh Signal Ltd. & Co. KG
Weigandufer 45, B?ro 115, D-12059 Berlin
Voice: +49 30 60988730

Amtsgericht Charlottenburg HRA 44413 B
Ust.-ID: DE266981999
Gesch?ftsf?hrer: Michal Bielicki
Pers?nlich Haftende Gesellschafterin:
Seventh Signal Ltd, 69 Great Hampton St. Birmingham, 
B18 6EW, GB, Company Nr.: 06889439
WWW.: http://www.seventhsignal.de


----


From tculjaga at gmail.com  Wed Aug 17 02:47:55 2011
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 17 Aug 2011 00:47:55 +0200
Subject: [Freeswitch-users] FS performance using ESL
In-Reply-To: <CAKbxfG__pZe9yhB7QY2ovTvLg1QgBx3AyTxCP6GRznkinQMpXg@mail.gmail.com>
References: <CAJoKKdU9VPzOS-C3jHCjG6Y0V-Jo_j7hqYwt6-Zh7ohb1pq3xQ@mail.gmail.com>
	<CAKbxfG-SWBzu0dBv_5B0v7710q7vLeeduT9L5V_cnceYHjrGJQ@mail.gmail.com>
	<CAJoKKdUcA=9CvSdRPhaoMopK+f4K2W1DO0=0b5mck_KEjOc9pQ@mail.gmail.com>
	<CAJoKKdXR64qNfCYj0uyYFda4YLavwd_NFn3bH6Td+2eSGo7Y1w@mail.gmail.com>
	<CAKbxfG_SnLS2c6CnepKcy=ea7CY=JnM5Lyy_Rv8ahUYpqJdooA@mail.gmail.com>
	<CAJoKKdX_sPBeWGv2xq7NOX4gd2Q1QQvWpm0wiSLv+67PvaGFVQ@mail.gmail.com>
	<CAKbxfG__pZe9yhB7QY2ovTvLg1QgBx3AyTxCP6GRznkinQMpXg@mail.gmail.com>
Message-ID: <CAJoKKdUxTFvbhshrYmC9Jj59zpnCM1AQ8zqTbQty-awEJhUZOA@mail.gmail.com>

On Tue, Aug 16, 2011 at 8:21 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> supplying full or not will not change performance it just controls
> weather or not each socket has full control to do other event socket
> commands besides ones that relate to the specific channel.
>
> myevents is probably a touch more efficient than filtering on
> unique-id but its negligible.
>
> The best way to gain performance is to limit the number of events you
> subscribe to, to the bare necessity.
>


understand, thanks for your response.

T.



>
>
>
> On Tue, Aug 16, 2011 at 7:44 AM, Tihomir Culjaga <tculjaga at gmail.com>
> wrote:
> >
> >
> > On Mon, Aug 15, 2011 at 6:30 PM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> You must have something setup strangely cos it would definitely reduce
> >> your overall cps to use ESL but not down to 2 CPS.
> >>
> >> Did you look over the server stats like top etc and look for any
> >> misconfiguration?
> >>
> > Hello Anthony, thanks for your response....
> >
> > yay, i found the cause .... testserver and FS were running on the same
> > server. The server had just 1GB of RAM and of course ... by forking
> > testserver (on 8 CPS) took all the remaining RAM ending to write into
> swap
> > ... this triggered a domino effect on the entire server becoming less and
> > less responsive as testserver started to run from swap!!!... yay ..
> really
> > bad... didn't see it happen until i started nmon ... top/htop didn't make
> it
> > in time to show this issue..
> >
> > anyhow, i moved testserver to another machine reaching 35 CPS ... really
> > nice indeed.
> >
> >
> > now, having a referent point (testserver) im trying to reach that 35 CPS
> > with a java application.
> >
> > yes, i notices few issues :=)
> >
> > please check: http://pastebin.freeswitch.org/17052
> >
> >
> >
> ------------------------------------snipp-----------------------------------
> >
> > Control: full
> >
> >
> > // here i subscribe to all events ... well not good idea but its a start
> > events plain all
> >
> > Content-Type: command/reply
> > Reply-Text: +OK event listener enabled plain
> >
> > //and here i do a filter per uuid
> > filter Unique-ID f7a7b97b-df96-41f3-a6a3-fdf24350a45c
> >
> > Content-Type: command/reply
> > Reply-Text: +OK filter added.
> > [Unique-ID]=[f7a7b97b-df96-41f3-a6a3-fdf24350a45c]
> >
> > linger
> >
> > Content-Type: command/reply
> > Reply-Text: +OK will linger
> >
> >
> > // here i send answer in sync mode ( i could change it into async)
> > sendmsg
> > call-command: execute
> > execute-app-name: answer
> > event-lock: true
> >
> > Content-Type: command/reply
> > Reply-Text: +OK
> >
> > Content-Length: 1805
> > Content-Type: text/event-plain
> >
> >
> >
> --------------------------------------------------------------------------------
> >
> >
> >
> >
> > so my questions:
> >
> > if i use <action application="socket" data="192.168.254.99:8084 async
> > full"/> and if i subscribe to "myevents" i don't need to set a filter on
> > uuid and i could gain performance.
> > if i use <action application="socket" data="192.168.254.99:8084async"/>  i
> > will be getting events for the call in question only... so no special
> > filters needed and i could limit the number of events im subscribing
> >
> >
> > what is a better approach in a matter of performance ?
> > What do i loose/gain by using async full vs async mode ?
> >
> >
> >
> > Thanks for your answer,
> > Tihomir.
> >
> >
> >
> >
> >
> >
> >
> >
> >>
> >> On Thu, Aug 11, 2011 at 6:56 PM, Tihomir Culjaga <tculjaga at gmail.com>
> >> wrote:
> >> > is there any other method than esl to controll calls on FS from an
> >> > eternal
> >> > application?
> >> > will mod_curl or mod_xml_curl get better performance?
> >> >
> >> > T.
> >> >
> >> > On Fri, Aug 12, 2011 at 1:33 AM, Tihomir Culjaga <tculjaga at gmail.com>
> >> > wrote:
> >> >>
> >> >> Hi Anthony, thanks for your response ...
> >> >>
> >> >>
> >> >> this is what i have:
> >> >>
> >> >>         esl_filter(&handle, "unique-id",
> >> >> esl_event_get_header(handle.info_event, "caller-unique-id"));
> >> >>         esl_events(&handle, ESL_EVENT_TYPE_PLAIN, "CHANNEL_DATA
> >> >> CHANNEL_EXECUTE_COMPLETE CHANNEL_HANGUP");
> >> >>
> >> >> what do you suggest i put there ?
> >> >>
> >> >>
> >> >> is the inbound method less costly ?
> >> >>
> >> >>
> >> >>
> >> >>
> >> >> I modified testserver.c just a bit...
> >> >>
> >> >> #include <sys/types.h>  /* include this before any other sys headers
> */
> >> >> #include <sys/wait.h>   /* header for waitpid() and various macros */
> >> >> #include <signal.h>     /* header for signal functions */
> >> >> #include <stdio.h>      /* header for fprintf() */
> >> >> #include <unistd.h>     /* header for fork() */
> >> >> #include <stdlib.h>
> >> >> #include <esl.h>
> >> >>
> >> >> void sig_chld(int);     /* prototype for our SIGCHLD handler */
> >> >>
> >> >> static void mycallback(esl_socket_t server_sock, esl_socket_t
> >> >> client_sock,
> >> >> struct sockaddr_in *addr)
> >> >> {
> >> >>         esl_handle_t handle = {{0}};
> >> >>         int done = 0;
> >> >>         esl_status_t status;
> >> >>         time_t exp = 0;
> >> >>
> >> >>         if (fork() != 0) {
> >> >>                 close(client_sock);
> >> >>                 return;
> >> >>         }
> >> >>
> >> >>         esl_attach_handle(&handle, client_sock, addr);
> >> >>
> >> >>         esl_log(ESL_LOG_INFO, "Connected! %d\n", handle.sock);
> >> >>
> >> >>         esl_filter(&handle, "unique-id",
> >> >> esl_event_get_header(handle.info_event, "caller-unique-id"));
> >> >>         esl_events(&handle, ESL_EVENT_TYPE_PLAIN, "CHANNEL_DATA
> >> >> CHANNEL_EXECUTE_COMPLETE CHANNEL_HANGUP");
> >> >>
> >> >>         esl_send_recv(&handle, "linger");
> >> >>
> >> >>         esl_execute(&handle, "answer", NULL, NULL);
> >> >>         //esl_execute(&handle, "conference", "3000 at default", NULL);
> >> >>         esl_execute(&handle, "playback",
> >> >> "/home/tculjaga/myWavFile.wav",
> >> >> NULL);
> >> >>         //esl_execute(&handle, "sleep", "1000", NULL);
> >> >>         //esl_execute(&handle, "hangup", NULL, NULL);
> >> >>
> >> >>         while((status = esl_recv_timed(&handle, 1000)) != ESL_FAIL) {
> >> >>                 if (done) {
> >> >>                         if (time(NULL) >= exp) {
> >> >>                                 break;
> >> >>                         }
> >> >>                 } else if (status == ESL_SUCCESS) {
> >> >>                         const char *type =
> >> >> esl_event_get_header(handle.last_event, "content-type");
> >> >>                         if (type && !strcasecmp(type,
> >> >> "text/disconnect-notice")) {
> >> >>                                 const char *dispo =
> >> >> esl_event_get_header(handle.last_event, "content-disposition");
> >> >>                                 esl_log(ESL_LOG_INFO, "Got a
> >> >> disconnection
> >> >> notice dispostion: [%s]\n", dispo ? dispo : "");
> >> >>                                 if (!strcmp(dispo, "linger")) {
> >> >>                                         done = 1;
> >> >>                                         esl_log(ESL_LOG_INFO,
> "Waiting
> >> >> 5
> >> >> seconds for any remaining events.\n");
> >> >>                                         exp = time(NULL) + 5;
> >> >>                                 }
> >> >>                         }
> >> >>                 }
> >> >>         }
> >> >>
> >> >>         esl_log(ESL_LOG_INFO, "Disconnected! %d\n", handle.sock);
> >> >>         esl_disconnect(&handle);
> >> >>
> >> >>         close(client_sock);
> >> >>
> >> >>         _exit(0);
> >> >> }
> >> >>
> >> >> /*
> >> >>  * The signal handler function -- only gets called when a SIGCHLD
> >> >>  * is received, ie when a child terminates
> >> >>  */
> >> >> void sig_chld(int signo)
> >> >> {
> >> >>     int status;
> >> >>
> >> >>     /* Wait for any child without blocking */
> >> >>     if (waitpid(-1, &status, WNOHANG) < 0)
> >> >>     {
> >> >>         /*
> >> >>          * calling standard I/O functions like fprintf() in a
> >> >>          * signal handler is not recommended, but probably OK
> >> >>          * in toy programs like this one.
> >> >>          */
> >> >>         fprintf(stderr, "waitpid failed\n");
> >> >>         return;
> >> >>     }
> >> >> }
> >> >>
> >> >> int main(void)
> >> >> {
> >> >>         struct sigaction act;
> >> >>
> >> >>         /* Assign sig_chld as our SIGCHLD handler */
> >> >>         act.sa_handler = sig_chld;
> >> >>
> >> >>         /* We don't want to block any other signals in this example
> */
> >> >>         sigemptyset(&act.sa_mask);
> >> >>
> >> >>         /*
> >> >>          * We're only interested in children that have terminated,
> not
> >> >> ones
> >> >>          * which have been stopped (eg user pressing control-Z at
> >> >> terminal)
> >> >>          */
> >> >>         act.sa_flags = SA_NOCLDSTOP;
> >> >>
> >> >>         /*
> >> >>          * Make these values effective. If we were writing a real
> >> >>          * application, we would probably save the old value instead
> of
> >> >>          * passing NULL.
> >> >>          */
> >> >> /*      if (sigaction(SIGCHLD, &act, NULL) < 0)
> >> >>         {
> >> >>                 fprintf(stderr, "sigaction failed\n");
> >> >>                 return 1;
> >> >>         }
> >> >> */
> >> >>         signal(SIGCHLD, SIG_IGN);
> >> >>
> >> >>         esl_global_set_default_logger(0);
> >> >>         esl_listen("localhost", 8088, mycallback);
> >> >>
> >> >>         return 0;
> >> >> }
> >> >>
> >> >>
> >> >>
> >> >>
> >> >> On Thu, Aug 11, 2011 at 9:59 PM, Anthony Minessale
> >> >> <anthony.minessale at gmail.com> wrote:
> >> >>>
> >> >>> try removing the filter and event subscriptions
> >> >>> it's costly to consume all of the events especially at 75cps.
> >> >>>
> >> >>>
> >> >>> On Thu, Aug 11, 2011 at 5:23 AM, Tihomir Culjaga <
> tculjaga at gmail.com>
> >> >>> wrote:
> >> >>> > hello,
> >> >>> >
> >> >>> > im wondering how much performance do we loose when using ESL
> instead
> >> >>> > of
> >> >>> > running it via dialplan?
> >> >>> >
> >> >>> >
> >> >>> > without ESL with a fine tuned FS and a short dialplan ( answer,
> >> >>> > playback
> >> >>> > like 20 seconds file, hangup ) im able to service 75 CPS. On the
> >> >>> > same
> >> >>> > FS,
> >> >>> > when i use ESL to answer the call, playback the same file and
> >> >>> > hangup,
> >> >>> > im not
> >> >>> > able to run more than 2 CPS... this is a huge impact and i really
> >> >>> > can't
> >> >>> > believe it.
> >> >>> >
> >> >>> > I'm using event-socket outbound e.g.:
> >> >>> >
> >> >>> > <action application="socket" data="127.0.0.1:8088 async full"/>
> >> >>> >
> >> >>> > my extension looks like:
> >> >>> >
> >> >>> > <extension name="ESL_C">
> >> >>> >   <condition field="destination_number" expression="^(6666)$">
> >> >>> >     <action application="socket" data="127.0.0.1:8088 async
> full"/>
> >> >>> >     <action application="sleep" data="1000"/>
> >> >>> >     <action application="hangup"/>
> >> >>> >   </condition>
> >> >>> > </extension>
> >> >>> >
> >> >>> >
> >> >>> > im using testserver from lib/esl/ and i just removed the
> conference
> >> >>> > command
> >> >>> > and added the playback one.... also i moved the esl_debug lvl to 0
> >> >>> >
> >> >>> >
> >> >>> > anyhow, FS cannot run more than 2 CPS compared to 75 CPS when the
> >> >>> > playback
> >> >>> > is done from the dialplan.
> >> >>> >
> >> >>> >
> >> >>> > Please, can someone give me a clue on what is going on?
> >> >>> > Maybe im doing something wrong?
> >> >>> > how to get maximum FS performance using ESL ?
> >> >>> >
> >> >>> >
> >> >>> >
> >> >>> > Regards,
> >> >>> > Tihomir.
> >> >>> >
> >> >>> >
> >> >>> > _______________________________________________
> >> >>> > Join us at ClueCon 2011, Aug 9-11, Chicago
> >> >>> > http://www.cluecon.com 877-7-4ACLUE
> >> >>> >
> >> >>> > FreeSWITCH-users mailing list
> >> >>> > FreeSWITCH-users at lists.freeswitch.org
> >> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>> >
> >> >>> >
> >> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>> > http://www.freeswitch.org
> >> >>> >
> >> >>> >
> >> >>>
> >> >>>
> >> >>>
> >> >>> --
> >> >>> Anthony Minessale II
> >> >>>
> >> >>> 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
> >> >>> googletalk:conf+888 at conference.freeswitch.org
> >> >>> pstn:+19193869900
> >> >>>
> >> >>> _______________________________________________
> >> >>> Join us at ClueCon 2011, Aug 9-11, Chicago
> >> >>> http://www.cluecon.com 877-7-4ACLUE
> >> >>>
> >> >>> FreeSWITCH-users mailing list
> >> >>> FreeSWITCH-users at lists.freeswitch.org
> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>>
> >> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>> http://www.freeswitch.org
> >> >>
> >> >
> >> >
> >> > _______________________________________________
> >> > Join us at ClueCon 2011, Aug 9-11, Chicago
> >> > http://www.cluecon.com 877-7-4ACLUE
> >> >
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> 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
> >> googletalk:conf+888 at conference.freeswitch.org
> >> pstn:+19193869900
> >>
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/e4e2eaa4/attachment-0001.html 

From db3l.net at gmail.com  Wed Aug 17 02:55:52 2011
From: db3l.net at gmail.com (David Bolen)
Date: Tue, 16 Aug 2011 18:55:52 -0400
Subject: [Freeswitch-users] DTMF bleed through in conference
References: <m2liuuz3y9.fsf@valheru.db3l.homeip.net>
	<CAKzWOxU_oaWg1_c=92W30g-pZyWkjWWW2YV5bh_hG-r1mZdVZA@mail.gmail.com>
	<CAKDfjgf2bYTDPsFaoM3ubvu=NGoPxN-tSJR0aCo7NZyRcmAqFw@mail.gmail.com>
	<m2hb5hyqao.fsf@valheru.db3l.homeip.net>
	<CALNZuwm=zsnmMwTfQbaruAZANg-FsEA5s9wnK_48iq3fcvPFaw@mail.gmail.com>
Message-ID: <m2d3g5ynif.fsf@valheru.db3l.homeip.net>

Avi Marcus <avi at avimarcus.net> writes:

> Could this be a keypad sound that gets picked up by the microphone?

Yeah, I considered that (wouldn't be the first time I've confused
things while running experiments with a bunch of microphones and
outputs next to each other), but in past tests, I've muted all input
devices in the conference, and still hear it.  Not sure I remembered
to do that in the test in my last message, so I just re-ran it, with
both blink and the PAP2T analog phone muted, and it still occurred.

In taking it one step further though, if the leg using DTMF is muted
at the conference level (either via the cli or invoking mute through
the DTMF bindings), then while DTMF is still recognized, nothing makes
it to other participants.  Which I guess either means something really
is in-band (which doesn't explain my rtp pcap being silent, plus if
in-band I'd expect to hear the full tone, since blink, which no analog
DTMF detector should be all or nothing) or that flag is indirectly
covering up the issue by avoiding whatever processing is putting the
glitch on the other participant leg. I'll probably try adding some
debugging to mod_conference at a few points to see if I can at least
identify where it thinks the audio may be coming from (or at least
confirm it thinks there is audio on the muted channel).

-- David



From peter.olsson at visionutveckling.se  Mon Aug 15 22:35:47 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 15 Aug 2011 20:35:47 +0200
Subject: [Freeswitch-users] Mod event socket error
In-Reply-To: <CALV5c5NZXjzyy6e4kDv7+5X1gZHgwaXNRx_PY5r63Mu7Ts=6DQ@mail.gmail.com>
References: <CALV5c5NZXjzyy6e4kDv7+5X1gZHgwaXNRx_PY5r63Mu7Ts=6DQ@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59EBABB8C8@cooper>

It means you're reading too slow from the socket, so FS will try to push more events then you will handle.

PS! If this mail arrives a couple of days too late, it's because of some local routing problems from my ISP, so the question is probably already answered by then :)

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Leonardo P. Bidinoto [leonardo.bidinoto at voicetechnology.com.br]
Skickat: den 15 augusti 2011 20:21
Till: FreeSwitch Users List
?mne: [Freeswitch-users] Mod event socket error

Hi Guys,

Im getting a lot of errors in my test machines, like these:

2011-08-15 10:12:39.433325 [CRIT] mod_event_socket.c:378 Lost 244 events!
2011-08-15 10:12:47.689947 [CRIT] mod_event_socket.c:378 Lost 32 events!
2011-08-15 10:12:48.770116 [CRIT] mod_event_socket.c:378 Lost 3 events!
2011-08-15 10:12:50.771600 [CRIT] mod_event_socket.c:378 Lost 54 events!
2011-08-15 10:12:50.791728 [CRIT] mod_event_socket.c:378 Lost 9 events!
2011-08-15 10:12:51.012380 [CRIT] mod_event_socket.c:378 Lost 39 events!
2011-08-15 10:12:51.012380 [CRIT] mod_event_socket.c:378 Lost 1 events!

What this kind of error means?

--
Leonardo Pires Bidinoto
Voice Technology
www.voicetechnology.com.br<http://www.voicetechnology.com.br>
!DSPAM:4e49644532765854471514!


From anthony.minessale at gmail.com  Wed Aug 17 04:11:20 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Aug 2011 19:11:20 -0500
Subject: [Freeswitch-users] DTMF bleed through in conference
In-Reply-To: <m2d3g5ynif.fsf@valheru.db3l.homeip.net>
References: <m2liuuz3y9.fsf@valheru.db3l.homeip.net>
	<CAKzWOxU_oaWg1_c=92W30g-pZyWkjWWW2YV5bh_hG-r1mZdVZA@mail.gmail.com>
	<CAKDfjgf2bYTDPsFaoM3ubvu=NGoPxN-tSJR0aCo7NZyRcmAqFw@mail.gmail.com>
	<m2hb5hyqao.fsf@valheru.db3l.homeip.net>
	<CALNZuwm=zsnmMwTfQbaruAZANg-FsEA5s9wnK_48iq3fcvPFaw@mail.gmail.com>
	<m2d3g5ynif.fsf@valheru.db3l.homeip.net>
Message-ID: <CAKbxfG8zWwtd5=fw_TeQeY6RMDvs0a9dPSk-2BBO=M_LUtxoLw@mail.gmail.com>

even when you have 100% soft phones you hear the blip?
Do you hear the blip on every phone?

Something in the mix must be doing generated tones or you would not
hear them.  FS would not just invent them.


On Tue, Aug 16, 2011 at 5:55 PM, David Bolen <db3l.net at gmail.com> wrote:
> Avi Marcus <avi at avimarcus.net> writes:
>
>> Could this be a keypad sound that gets picked up by the microphone?
>
> Yeah, I considered that (wouldn't be the first time I've confused
> things while running experiments with a bunch of microphones and
> outputs next to each other), but in past tests, I've muted all input
> devices in the conference, and still hear it. ?Not sure I remembered
> to do that in the test in my last message, so I just re-ran it, with
> both blink and the PAP2T analog phone muted, and it still occurred.
>
> In taking it one step further though, if the leg using DTMF is muted
> at the conference level (either via the cli or invoking mute through
> the DTMF bindings), then while DTMF is still recognized, nothing makes
> it to other participants. ?Which I guess either means something really
> is in-band (which doesn't explain my rtp pcap being silent, plus if
> in-band I'd expect to hear the full tone, since blink, which no analog
> DTMF detector should be all or nothing) or that flag is indirectly
> covering up the issue by avoiding whatever processing is putting the
> glitch on the other participant leg. I'll probably try adding some
> debugging to mod_conference at a few points to see if I can at least
> identify where it thinks the audio may be coming from (or at least
> confirm it thinks there is audio on the muted channel).
>
> -- David
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From anthony.minessale at gmail.com  Wed Aug 17 04:29:26 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 16 Aug 2011 19:29:26 -0500
Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
In-Reply-To: <1313446639.81086.YahooMailNeo@web161008.mail.bf1.yahoo.com>
References: <4E4164C0.8030507@tiendalinux.com>
	<1312937649.7702.YahooMailNeo@web161011.mail.bf1.yahoo.com>
	<CAKbxfG_802p+G1a9HtLEh_i-K1C06fpZgyR4_NP=jZJR68YK8g@mail.gmail.com>
	<1313446639.81086.YahooMailNeo@web161008.mail.bf1.yahoo.com>
Message-ID: <CAKbxfG8jVqBMWODa72nS692buKjeX_LrAT05zz1HeQqg5MM3Yg@mail.gmail.com>

You should never answer a call before bridging it anyway, it breaks all of
the accounting.
It would make sense to find out why the provider is doing that and get it
fixed.


On Mon, Aug 15, 2011 at 5:17 PM, Sam <lakersman2006 at yahoo.com> wrote:

> Anthony,
>
> My gripe was not about simply having a DIALSTATUS variable in Freeswitch
> which copies what is from "originate_disposition" what I wanted is to be
> able to get the status of the B-Leg because right now when early media is
> played (which i wanted)  "originate_disposition" shows "ANSWER" which I
> think is caused by me explitly called the "answer" app in my dialplan before
> the bridge app, this is because my DID provider requires an answer/sip 200
> or else it will keep re-sending the sip invite, therefore causing freeswitch
> to keep creating new channels. All I want is to be able to get the proper
> sip/hangup/dial statuses of the B-leg.
>
> ------------------------------
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Wednesday, August 10, 2011 8:52 AM
> *Subject:* Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
> =D
>
> ok, sure.  If that's your only complaint.... see
> commit 9d98d49f0556fb79656c8403f285ae0a615439d3
>
>
>
> Some caveats
>
> 1) There is actually less specific, more generalized data in this
> DIALSTATUS variable than what we already report, when you're ready to move
> on see the originate_disposition variable:  It's kind of like going from
> reporting the precise geo-location of a cafe in Paris to generalizing it to
> "EUROPE"
>
> We follow the Q.850 standard for call cause codes and follow the SIP RFC to
> map sip response codes to/from the Q.850 equivalent.  Also each module has
> its own version "sip_hangup_disposition" for sip so you have both the real
> sip response code AND the official Q.850 equiv variables set on each call.
>
>
> 2) We don't have a torture feature so we never return that code.
>
>
> 3) Since our originate can return before a call is answered I added "EARLY"
> which means the originate succeeded but its still not answered.
>
> 4) For any others that do not map directly to FreeSWITCH, I just installed
> a copy of originate_disposition for good measure.
>
> P.S
>
> This email took longer to compose than the patch did while sitting in the
> middle of a crowded room so you probably could have simply parsed the
> originate originate_disposition yourself but if it helps people get over
> being stuck in a paradigm, it's worth it for me to write........
>
>
> On Tue, Aug 9, 2011 at 7:54 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> I find that Asterisk's AGI is much easier to use, they allow you to
> retrieve the dial status much easier than freeswitch's api's. Come on
> freeswitch, if you want to be better than asterisk, you should make it
> easier to get the dialstatus, etc. At this point asterisk is still defacto.
>
> ------------------------------
> *From:* Nestor A Diaz <nestor at tiendalinux.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Tuesday, August 9, 2011 9:48 AM
> *Subject:* [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
> Hi Guys.
>
> I am starting to use FreeSWITCH, i am an asterisk user since the 1.0.7
> release appears as a package on the debian distribution, at the beginning i
> was amazed by the fact i can build a PBX for my own business and i did,
> later i began to install this system for my customers and sooner i meet the
> problems, however being the software open source i always find a way to fix
> things using patchs from others, sometimes i felt how my life was at risk
> when the system stops working and that usually happens when i have to use
> queues and dealing with digium hardware.
>
> Fixing those problems either by applying patches or by changing the
> hardware where the digium cards were supposed to be installed helps me, but
> that was to much stress for me and seeking for a balance that will let me
> invest more time on services, configuration and hoping to have better
> hardware options brings me to freeswitch.
>
> I agree with freeswitch philosophy that instead of having thousands of
> modules that don't work fine i prefer just a few that works the way it
> should be, a rock solid system for a corporate pbx and a call center is what
> i want.
>
> So here i am trying to begin the conversion, and i hope the information we
> can transcript in this list will help others that want to try another
> alternative to asterisk.
>
> First of all i think the saner for a migration is to have the two systems
> running either on the same machine or different and use the stable features
> of each one.
>
> So could you please freeswitch users help me with this rosetta stone
> migration guide in order to post it to voip-info.org or freeswitch wiki (i
> list only the ones i currently use ):
>
>
>   *Technology* *Asterisk* *Freeswitch*  PSTN Connectivity (Digium /
> Sangoma) dahdi freetdm  IAX2 mod_iax ?? none stable yet.
> Use Asterisk to forward traffic via SIP.
> Enable Hardware HPET for IAX2 trunk if card not available for Asterisk  Bluetooth
> Channel chan_mobile ??
> Use asterisk via SIP
>   Skype Skypeforasterisk (no longer for sale) mod_skypeopen  CDR
> Stadistics Arternic cdr-stats ??  Queue Statistics  Asteriskguru
> queue-stats ??  Web Management Freepbx ??  IVR AGI / AMI Event Socket  Codec
> G.729 Transcodind Cards
> G.729 licenses
> Free G.729 (Intel IPP) Transcodind Cards
> G.729 licenses
> fsg729 Intel IPP(any experience with it ? )  Fax Handling Iaxmodem with
> Hylafax ??
> Iaxmodem via asterisk to FS via SIP ?
>   SIP chan_sip sofia  ACD app_queue mod_callcenter
>
> Thank you all
>
>
> --
> Nestor A. Diaz
> Ingeniero de Sistemas
> Tel. +57 1-485-3020 x 211
> Cel. +57 316-227-3593
> Tel. SIP: sip:211 at tiendalinux.com
> Email/MSN: nestor at tiendalinux.com
> http://www.tiendalinux.com/
> Bogota, Colombia
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110816/492208dd/attachment-0001.html 

From peter.olsson at visionutveckling.se  Mon Aug 15 20:35:44 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 15 Aug 2011 18:35:44 +0200
Subject: [Freeswitch-users] Help with dual IP gateways
In-Reply-To: <005501cc5b67$be0ace00$3a206a00$@yahoo.com>
References: <005501cc5b67$be0ace00$3a206a00$@yahoo.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59EBABB8C5@cooper>

The easiest way to accomplish this is to have two NIC's in the FS-server. One NIC connected to the Internet - to the provider that you want to route VoIP over. And the other NIC connected to the local LAN. Instead of configuring the "normal" default GW in the network, you use the default GW for the other provider.

This way you will be able to place calls, and it will be routed on the other connection - actually all Internet traffic on the FS-box will use the VoIP-specific Internet provider.

/Peter

________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Lars Zeb [larclap at yahoo.com]
Skickat: den 15 augusti 2011 18:23
Till: 'FreeSWITCH Users Help'
?mne: [Freeswitch-users] Help with dual IP gateways

Currently my LAN is connected to the internet via DSL. The FreeSWITCH box is
on this subnet. To save money, I am moving the data portion of my LAN to a
new ISP and I want to segregate the VOIP to another ISP. I am tired of
having a bad VOIP connection during lengthy downloads.

My VOIP and FreeSWITCH skills are minimal. I have used FreeSWITCH for over a
year in a home/business environment. The only reason it is working is with
the help of this list.

My knowledge of IP is similar. I do not know how to setup a LAN with two
gateways with all nodes seeing one another. I do want to be able to call out
via FreeSWITCH from a softphone on the data portion of the new LAN.

A friend suggested I need a dual ported WAN firewall/router with load
balancing to enable all the nodes to be on the same subnet. Can anyone help
me with suggestions? Is there a consultant I can hire to help with this?

Thanks, Lars



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

!DSPAM:4e49491232761283432527!



From db3l.net at gmail.com  Wed Aug 17 04:49:54 2011
From: db3l.net at gmail.com (David Bolen)
Date: Tue, 16 Aug 2011 20:49:54 -0400
Subject: [Freeswitch-users] DTMF bleed through in conference
References: <m2liuuz3y9.fsf@valheru.db3l.homeip.net>
	<CAKzWOxU_oaWg1_c=92W30g-pZyWkjWWW2YV5bh_hG-r1mZdVZA@mail.gmail.com>
	<CAKDfjgf2bYTDPsFaoM3ubvu=NGoPxN-tSJR0aCo7NZyRcmAqFw@mail.gmail.com>
	<m2hb5hyqao.fsf@valheru.db3l.homeip.net>
	<CALNZuwm=zsnmMwTfQbaruAZANg-FsEA5s9wnK_48iq3fcvPFaw@mail.gmail.com>
	<m2d3g5ynif.fsf@valheru.db3l.homeip.net>
	<CAKbxfG8zWwtd5=fw_TeQeY6RMDvs0a9dPSk-2BBO=M_LUtxoLw@mail.gmail.com>
Message-ID: <m262lwzwst.fsf@valheru.db3l.homeip.net>

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

> even when you have 100% soft phones you hear the blip?
> Do you hear the blip on every phone?

Of the two phones in my recent tests, the DTMF originator was the soft
phone, the other participant was an analog handset through the PAP2T.
Both were muted (locally on the analog phone, and in the software for
the soft phone). I can try a test with just multiple soft phones
though.

I seem to get the exact same thing regardless of DTMF source (soft
phone, analog through PAP2T, analog through PSTN independent of
gateway) or other participants on the call.  So if the tone was
in-band, it would have to be multiple similar failures of all those
sources.

And yes, though I haven't run that as recently, if there are >2
participants I'm pretty sure all others other than the DMTF source
hear the blip.

> Something in the mix must be doing generated tones or you would not
> hear them.  FS would not just invent them.

That certainly seems to be the simplest/most plausible answer, I just
don't know how it could be happening with the softphone as DTMF
source since there should be no in-band tone.

I did see where mod_conference can have a member flag to replicate
DTMF to other participants, but I don't have that on, so I don't think
that's it.  It would probably be a full tone too, not the brief burst.

I'm sure it's got to be explainable, but so far I guess I just haven't
looked in the right place.

-- David



From lakersman2006 at yahoo.com  Wed Aug 17 05:19:53 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Tue, 16 Aug 2011 18:19:53 -0700 (PDT)
Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
In-Reply-To: <CAKbxfG8jVqBMWODa72nS692buKjeX_LrAT05zz1HeQqg5MM3Yg@mail.gmail.com>
References: <4E4164C0.8030507@tiendalinux.com>
	<1312937649.7702.YahooMailNeo@web161011.mail.bf1.yahoo.com>
	<CAKbxfG_802p+G1a9HtLEh_i-K1C06fpZgyR4_NP=jZJR68YK8g@mail.gmail.com>
	<1313446639.81086.YahooMailNeo@web161008.mail.bf1.yahoo.com>
	<CAKbxfG8jVqBMWODa72nS692buKjeX_LrAT05zz1HeQqg5MM3Yg@mail.gmail.com>
Message-ID: <1313543993.57876.YahooMailNeo@web161017.mail.bf1.yahoo.com>

How come in some of the examples I see it calling answer()?

http://wiki.freeswitch.org/wiki/Perl_Console_IVR_Example



________________________________
From: Anthony Minessale <anthony.minessale at gmail.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, August 16, 2011 5:29 PM
Subject: Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide


You should never answer a call before bridging it anyway, it breaks all of the accounting.
It would make sense to find out why the provider is doing that and get it fixed.



On Mon, Aug 15, 2011 at 5:17 PM, Sam <lakersman2006 at yahoo.com> wrote:

Anthony,
>
>My gripe was not about simply having a DIALSTATUS variable in Freeswitch which copies what is from "originate_disposition" what I wanted is to be able to get the status of the B-Leg because right now when early media is played (which i wanted)? "originate_disposition" shows "ANSWER" which I think is caused by me explitly called the "answer" app in my dialplan before the bridge app, this is because my DID provider requires an answer/sip 200 or else it will keep re-sending the sip invite, therefore causing freeswitch to keep creating new channels. All I want is to be able to get the proper sip/hangup/dial statuses of the B-leg.
>
>
>
>
>________________________________
> From: Anthony Minessale <anthony.minessale at gmail.com>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>Sent: Wednesday, August 10, 2011 8:52 AM
>Subject: Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
>
>
>=D?
>
>
>ok, sure. ?If that's your only complaint.... see commit?9d98d49f0556fb79656c8403f285ae0a615439d3
>
>
>
>Some caveats
>
>
>1) There is actually less?specific, more generalized data in this DIALSTATUS variable than what we already report, when you're ready to move on see the originate_disposition variable: ?It's kind of like going from reporting the precise geo-location of a cafe in Paris to generalizing it to "EUROPE"?
>
>
>We follow the Q.850 standard for call cause codes and follow the SIP RFC to map sip response codes to/from the Q.850?equivalent. ?Also each module has its own version "sip_hangup_disposition" for sip so you have both the real sip response code AND the official Q.850 equiv variables set on each call.
>
>
>
>
>2) We don't have a torture feature so we never return that code.
>
>
>
>
>3) Since our originate can return before a call is answered I added "EARLY" which means the originate succeeded but its still not answered.
>
>
>4) For any others that do not map directly to FreeSWITCH, I just installed a copy of originate_disposition for good measure.
>
>P.S?
>
>
>This email took longer to compose than the patch did while sitting in the middle of a crowded room so you probably could have simply parsed the originate originate_disposition yourself but if it helps people get over being stuck in a?paradigm, it's worth it for me to write........
>?
>
>
>On Tue, Aug 9, 2011 at 7:54 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
>I find that Asterisk's AGI is much easier to use, they allow you to retrieve the dial status much easier than freeswitch's api's. Come on freeswitch, if you want to be better than asterisk, you should make it easier to get the dialstatus, etc. At this point asterisk is still defacto.
>>
>>
>>
>>
>>________________________________
>> From: Nestor A Diaz <nestor at tiendalinux.com>
>>To: freeswitch-users at lists.freeswitch.org
>>Sent: Tuesday, August 9, 2011 9:48 AM
>>Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>>
>>
>>
>>Hi Guys.
>>
>>I am starting to use FreeSWITCH, i am an asterisk user since the
1.0.7
release appears as a package on the debian distribution, at the
beginning i was amazed by the fact i can build a PBX for my own
business
and i did, later i began to install this system for my customers and
sooner i meet the problems, however being the software open source i
always find a way to fix things using patchs from others, sometimes i
felt how my life was at risk when the system stops working and that
usually happens when i have to use queues and dealing with digium
hardware.
>>
>>Fixing those problems either by applying patches or by changing
the
hardware where the digium cards were supposed to be installed helps me,
but that was to much stress for me and seeking for a balance that will
let me invest more time on services, configuration and hoping to have
better hardware
options brings me to freeswitch.
>>
>>I agree with freeswitch philosophy that instead of having
thousands of
modules
that don't work fine i prefer just a few that works the way it should
be, a rock solid system for a corporate pbx and a call center is what i
want.
>>
>>So here i am trying to begin the conversion, and i hope the
information
we can transcript in this list will help others that want to try
another
alternative to asterisk.
>>
>>First of all i think the saner for a migration is to have the two
systems
running either on the same machine or different and use the stable
features of each one.
>>
>>So could you please freeswitch users help me with this rosetta
stone migration guide in order to post it to voip-info.org or
freeswitch wiki (i list only the ones i currently use ):
>>
>>
>>
>>Technology Asterisk Freeswitch 
>>PSTN Connectivity (Digium /
Sangoma) dahdi freetdm 
>>IAX2 mod_iax ?? none stable yet.
>>Use Asterisk to forward traffic via SIP.
>>Enable Hardware HPET for IAX2 trunk if card not available for Asterisk  
>>Bluetooth Channel chan_mobile ??
>>Use asterisk via SIP
>> 
>>Skype Skypeforasterisk (no longer for sale) mod_skypeopen 
>>CDR Stadistics Arternic cdr-stats  ?? 
>>Queue Statistics  Asteriskguru queue-stats ?? 
>>Web Management Freepbx ?? 
>>IVR AGI / AMI Event Socket 
>>Codec G.729 Transcodind Cards
>>G.729 licenses
>>Free G.729 (Intel IPP) Transcodind Cards
>>G.729 licenses
>>fsg729 Intel IPP(any experience with it ? ) 
>>Fax Handling Iaxmodem with Hylafax ??
>>Iaxmodem via asterisk to FS via SIP ?
>> 
>>SIP chan_sip sofia 
>>ACD app_queue mod_callcenter 
>>
>>Thank you all
>>
>>
>>-- 
>>Nestor A. Diaz
>>Ingeniero de Sistemas
>>Tel. +57 1-485-3020 x 211
>>Cel. +57 316-227-3593
>>Tel. SIP: sip:211 at tiendalinux.com
>>Email/MSN: nestor at tiendalinux.com
>>http://www.tiendalinux.com/
>>Bogota, Colombia 
>> 
>>
>>
>>_______________________________________________
>>Join us at ClueCon 2011, Aug 9-11, Chicago
>>http://www.cluecon.com 877-7-4ACLUE
>>
>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>>
>>
>>_______________________________________________
>>Join us at ClueCon 2011, Aug 9-11, Chicago
>>http://www.cluecon.com 877-7-4ACLUE
>>
>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>>
>
>
>-- 
>Anthony Minessale II
>
>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
>googletalk:conf+888 at conference.freeswitch.org
>pstn:+19193869900
>
>_______________________________________________
>Join us at ClueCon 2011, Aug 9-11, Chicago
>http://www.cluecon.com 877-7-4ACLUE
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>
>
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110816/5909a248/attachment-0001.html 

From mario_fs at mgtech.com  Wed Aug 17 06:01:36 2011
From: mario_fs at mgtech.com (Mario G)
Date: Tue, 16 Aug 2011 19:01:36 -0700
Subject: [Freeswitch-users] Help with dual IP gateways
In-Reply-To: <005501cc5b67$be0ace00$3a206a00$@yahoo.com>
References: <005501cc5b67$be0ace00$3a206a00$@yahoo.com>
Message-ID: <A6DCFC10-7528-4364-8D4C-9DC4ACD7AD92@mgtech.com>

I worked a year on this and was planning to put it on the wiki. Does this help: I have 2 DSLs, 1 static, 1 dynamic, FS is on a Mac mini with only 1 nic. The router does dual wan/load balancing. And by golly.... FS can auto switch between DSL lines when one has a problem. Wondering if this info is worth my time putting on the wiki, I did the OS X page and rewrote other stuff and it took weeks. But if there is interest I will do it.

On Aug 15, 2011, at 9:23 AM, Lars Zeb wrote:

> Currently my LAN is connected to the internet via DSL. The FreeSWITCH box is
> on this subnet. To save money, I am moving the data portion of my LAN to a
> new ISP and I want to segregate the VOIP to another ISP. I am tired of
> having a bad VOIP connection during lengthy downloads.
> 
> My VOIP and FreeSWITCH skills are minimal. I have used FreeSWITCH for over a
> year in a home/business environment. The only reason it is working is with
> the help of this list.
> 
> My knowledge of IP is similar. I do not know how to setup a LAN with two
> gateways with all nodes seeing one another. I do want to be able to call out
> via FreeSWITCH from a softphone on the data portion of the new LAN. 
> 
> A friend suggested I need a dual ported WAN firewall/router with load
> balancing to enable all the nodes to be on the same subnet. Can anyone help
> me with suggestions? Is there a consultant I can hire to help with this?
> 
> Thanks, Lars
> 
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From gohar.ahmed at vopium.com  Wed Aug 17 08:54:07 2011
From: gohar.ahmed at vopium.com (Gohar Ahmed)
Date: Wed, 17 Aug 2011 09:54:07 +0500
Subject: [Freeswitch-users] SIP proxy collect DTMF using FS
In-Reply-To: <CAJUJwthVmvx3vvHwpO+ZOS0tDaLREp4H3OfX39hBJQ0cJOvp_Q@mail.gmail.com>
References: <CAJUJwtjqMyvVX1eteYBWFf41Mz+fZjnRZA_0BR2DZs0R5zh_tA@mail.gmail.com>	<0D92C5B0-CD84-47B1-A17C-A2B083B760E2@freeswitch.org>
	<CAJUJwthVmvx3vvHwpO+ZOS0tDaLREp4H3OfX39hBJQ0cJOvp_Q@mail.gmail.com>
Message-ID: <02b801cc5c99$b3e9fdf0$1bbdf9d0$@ahmed@vopium.com>

Hey, 

 

Don't know if this will do the job or not but do take a look at deflect
application: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect 

This will remove the FS from the call path and tell the originator to go
somewhere else. 

 

I hope this helps, maybe some other gurus here suggest you anything better
than this.

 

Regards,

Gohar A.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sam
Govind
Sent: Monday, August 15, 2011 10:58 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] SIP proxy collect DTMF using FS

 

Thanks Brian for showing concern. I'm always open for ideas. What I'm trying
to achieve is collect DTMF from user and then have my SIP Proxy verify if a
particular caller is allowed to dial that destination(input as DTMF).

 

Obviously I could've done the same checks at FS, BUT I'm required to let SIP
Proxy verify instead. FreeSWITCH is only required to get input, release the
call and if Proxy allows the call only then call be routed to any other
FreeSWITCH (Pool of FS LoadBalanced).

 

This is supposed to simplify the operations of FS , decrease the load volume
on FS, and increase the call capacity.

 

If you've any better Ideas do share.

 

On Sun, Aug 14, 2011 at 8:39 AM, Brian West <brian at freeswitch.org> wrote:

What are you trying to do exactly because it sounds like you have selected
the most painful way to accomplish it?

 

/b

 

On Aug 9, 2011, at 2:11 AM, Sam Govind wrote:





Hi guys,

I'm looking to establish a scenario like this, any idea how to do it, if its
possible.

1- SIP proxy send call to FS where DTMF will be collected. (I'm thinking of
using PlayAndGetDigits)
2- DTMF collected be sent back to SIP proxy while FS ends the call
3- Call at SIP proxy end keeps running for some other processing.

basically I just need FS to collect DTMF and send those back to SIP Proxy.

Any ideas are welcome.

Thanks.

 


_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

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

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/09ff8eed/attachment.html 

From lakersman2006 at yahoo.com  Wed Aug 17 09:14:40 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Tue, 16 Aug 2011 22:14:40 -0700 (PDT)
Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
In-Reply-To: <CAKbxfG8jVqBMWODa72nS692buKjeX_LrAT05zz1HeQqg5MM3Yg@mail.gmail.com>
References: <4E4164C0.8030507@tiendalinux.com>
	<1312937649.7702.YahooMailNeo@web161011.mail.bf1.yahoo.com>
	<CAKbxfG_802p+G1a9HtLEh_i-K1C06fpZgyR4_NP=jZJR68YK8g@mail.gmail.com>
	<1313446639.81086.YahooMailNeo@web161008.mail.bf1.yahoo.com>
	<CAKbxfG8jVqBMWODa72nS692buKjeX_LrAT05zz1HeQqg5MM3Yg@mail.gmail.com>
Message-ID: <1313558080.89178.YahooMailNeo@web161010.mail.bf1.yahoo.com>

The DID provider I am using is from iCall, and I was searching through their website and noticed that they mentioned a quote with your name on it http://carriers.icall.com/open-source/
so it appears you have had experience with them.



________________________________
From: Anthony Minessale <anthony.minessale at gmail.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, August 16, 2011 5:29 PM
Subject: Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide


You should never answer a call before bridging it anyway, it breaks all of the accounting.
It would make sense to find out why the provider is doing that and get it fixed.



On Mon, Aug 15, 2011 at 5:17 PM, Sam <lakersman2006 at yahoo.com> wrote:

Anthony,
>
>My gripe was not about simply having a DIALSTATUS variable in Freeswitch which copies what is from "originate_disposition" what I wanted is to be able to get the status of the B-Leg because right now when early media is played (which i wanted)? "originate_disposition" shows "ANSWER" which I think is caused by me explitly called the "answer" app in my dialplan before the bridge app, this is because my DID provider requires an answer/sip 200 or else it will keep re-sending the sip invite, therefore causing freeswitch to keep creating new channels. All I want is to be able to get the proper sip/hangup/dial statuses of the B-leg.
>
>
>
>
>________________________________
> From: Anthony Minessale <anthony.minessale at gmail.com>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>Sent: Wednesday, August 10, 2011 8:52 AM
>Subject: Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
>
>
>=D?
>
>
>ok, sure. ?If that's your only complaint.... see commit?9d98d49f0556fb79656c8403f285ae0a615439d3
>
>
>
>Some caveats
>
>
>1) There is actually less?specific, more generalized data in this DIALSTATUS variable than what we already report, when you're ready to move on see the originate_disposition variable: ?It's kind of like going from reporting the precise geo-location of a cafe in Paris to generalizing it to "EUROPE"?
>
>
>We follow the Q.850 standard for call cause codes and follow the SIP RFC to map sip response codes to/from the Q.850?equivalent. ?Also each module has its own version "sip_hangup_disposition" for sip so you have both the real sip response code AND the official Q.850 equiv variables set on each call.
>
>
>
>
>2) We don't have a torture feature so we never return that code.
>
>
>
>
>3) Since our originate can return before a call is answered I added "EARLY" which means the originate succeeded but its still not answered.
>
>
>4) For any others that do not map directly to FreeSWITCH, I just installed a copy of originate_disposition for good measure.
>
>P.S?
>
>
>This email took longer to compose than the patch did while sitting in the middle of a crowded room so you probably could have simply parsed the originate originate_disposition yourself but if it helps people get over being stuck in a?paradigm, it's worth it for me to write........
>?
>
>
>On Tue, Aug 9, 2011 at 7:54 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
>I find that Asterisk's AGI is much easier to use, they allow you to retrieve the dial status much easier than freeswitch's api's. Come on freeswitch, if you want to be better than asterisk, you should make it easier to get the dialstatus, etc. At this point asterisk is still defacto.
>>
>>
>>
>>
>>________________________________
>> From: Nestor A Diaz <nestor at tiendalinux.com>
>>To: freeswitch-users at lists.freeswitch.org
>>Sent: Tuesday, August 9, 2011 9:48 AM
>>Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>>
>>
>>
>>Hi Guys.
>>
>>I am starting to use FreeSWITCH, i am an asterisk user since the
1.0.7
release appears as a package on the debian distribution, at the
beginning i was amazed by the fact i can build a PBX for my own
business
and i did, later i began to install this system for my customers and
sooner i meet the problems, however being the software open source i
always find a way to fix things using patchs from others, sometimes i
felt how my life was at risk when the system stops working and that
usually happens when i have to use queues and dealing with digium
hardware.
>>
>>Fixing those problems either by applying patches or by changing
the
hardware where the digium cards were supposed to be installed helps me,
but that was to much stress for me and seeking for a balance that will
let me invest more time on services, configuration and hoping to have
better hardware
options brings me to freeswitch.
>>
>>I agree with freeswitch philosophy that instead of having
thousands of
modules
that don't work fine i prefer just a few that works the way it should
be, a rock solid system for a corporate pbx and a call center is what i
want.
>>
>>So here i am trying to begin the conversion, and i hope the
information
we can transcript in this list will help others that want to try
another
alternative to asterisk.
>>
>>First of all i think the saner for a migration is to have the two
systems
running either on the same machine or different and use the stable
features of each one.
>>
>>So could you please freeswitch users help me with this rosetta
stone migration guide in order to post it to voip-info.org or
freeswitch wiki (i list only the ones i currently use ):
>>
>>
>>
>>Technology Asterisk Freeswitch 
>>PSTN Connectivity (Digium /
Sangoma) dahdi freetdm 
>>IAX2 mod_iax ?? none stable yet.
>>Use Asterisk to forward traffic via SIP.
>>Enable Hardware HPET for IAX2 trunk if card not available for Asterisk  
>>Bluetooth Channel chan_mobile ??
>>Use asterisk via SIP
>> 
>>Skype Skypeforasterisk (no longer for sale) mod_skypeopen 
>>CDR Stadistics Arternic cdr-stats  ?? 
>>Queue Statistics  Asteriskguru queue-stats ?? 
>>Web Management Freepbx ?? 
>>IVR AGI / AMI Event Socket 
>>Codec G.729 Transcodind Cards
>>G.729 licenses
>>Free G.729 (Intel IPP) Transcodind Cards
>>G.729 licenses
>>fsg729 Intel IPP(any experience with it ? ) 
>>Fax Handling Iaxmodem with Hylafax ??
>>Iaxmodem via asterisk to FS via SIP ?
>> 
>>SIP chan_sip sofia 
>>ACD app_queue mod_callcenter 
>>
>>Thank you all
>>
>>
>>-- 
>>Nestor A. Diaz
>>Ingeniero de Sistemas
>>Tel. +57 1-485-3020 x 211
>>Cel. +57 316-227-3593
>>Tel. SIP: sip:211 at tiendalinux.com
>>Email/MSN: nestor at tiendalinux.com
>>http://www.tiendalinux.com/
>>Bogota, Colombia 
>> 
>>
>>
>>_______________________________________________
>>Join us at ClueCon 2011, Aug 9-11, Chicago
>>http://www.cluecon.com 877-7-4ACLUE
>>
>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>>
>>
>>_______________________________________________
>>Join us at ClueCon 2011, Aug 9-11, Chicago
>>http://www.cluecon.com 877-7-4ACLUE
>>
>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>>
>
>
>-- 
>Anthony Minessale II
>
>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
>googletalk:conf+888 at conference.freeswitch.org
>pstn:+19193869900
>
>_______________________________________________
>Join us at ClueCon 2011, Aug 9-11, Chicago
>http://www.cluecon.com 877-7-4ACLUE
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>
>
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110816/bf72d86a/attachment-0001.html 

From lakersman2006 at yahoo.com  Wed Aug 17 11:33:46 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Wed, 17 Aug 2011 00:33:46 -0700 (PDT)
Subject: [Freeswitch-users] Simulating Asterisk dial(x:y:z)
Message-ID: <1313566426.80683.YahooMailNeo@web161019.mail.bf1.yahoo.com>

I want to simulate the Asterisk dial(x:y:z) app where (x=total call time, y=warning time, z=warning interval) in Freeswitch. I know it can be done with sched_hangup and sched_broadcast, my question is how can I call a series of "api_on_answer" comands to do it?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/e4401fbe/attachment.html 

From chrisbware at interfree.it  Wed Aug 17 12:39:23 2011
From: chrisbware at interfree.it (chrisbware at interfree.it)
Date: 17 Aug 2011 08:39:23 -0000
Subject: [Freeswitch-users] Help with dual IP gateways
Message-ID: <20110817083923.31480.qmail@community35.interfree.it>

An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/55af976d/attachment.html 

From chrisbware at interfree.it  Wed Aug 17 13:12:44 2011
From: chrisbware at interfree.it (chrisbware at interfree.it)
Date: 17 Aug 2011 09:12:44 -0000
Subject: [Freeswitch-users] Lua not playing wav files
Message-ID: <20110817091244.5582.qmail@community37.interfree.it>

An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/f0ae8e4e/attachment-0001.html 

From nasida at live.ru  Wed Aug 17 13:53:52 2011
From: nasida at live.ru (Yuriy Nasida)
Date: Wed, 17 Aug 2011 13:53:52 +0400
Subject: [Freeswitch-users] Lua not playing wav files
In-Reply-To: <20110817091244.5582.qmail@community37.interfree.it>
References: <20110817091244.5582.qmail@community37.interfree.it>
Message-ID: <SNT127-W13CB7ED823A696374E24B9D5280@phx.gbl>


I was try it. There is no differents. 

The issue  consists in that I get dialplan automatically through xml_curl + php (by intralanman). I see it by means xml_curl_debug_on:
<action application="lua" data="$1" inline="true"/>How can I disable "inline="true"" ?
In the mysql table created under the README from intralanman I see  field "type".  I can put "action" or "anti-action" only. How can I control "inline" parameter?
Many thanks to anyone who can help. 

Date: Wed, 17 Aug 2011 09:12:44 +0000
From: chrisbware at interfree.it
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Lua not playing wav files


 
try:
 
 
 
session:streamFile(message)
 
instead of
 
session:execute("playback", message)
 
 
 
-----Messaggio originale-----
Da: Yuriy Nasida <nasida at live.ru>
Inviato il: 16 Ago 2011 - 10:46
A: <freeswitch-users at lists.freeswitch.org>


 
 Hi Freeswitch-users,
  
 My simple lua script:
  
 freeswitch.consoleLog("err","start hello.lua\n")
 session:answer();
 message = "ivr/ivr-enter_destination_telephone_number.wav"
 session:execute("playback", message)
 session:hangup();
  
 Script looks fine I think, but FS doesn't play audio. If I use corresponding XML dialplan all work fine.
  
  
 logs when I use lua:
  
 2011-08-16 13:18:00.336003 [DEBUG] switch_core_state_machine.c:371 (sofia/external/79213777785 at 65.98.107.130:5080) State EXECUTE going to sleep
 2011-08-16 13:18:00.336003 [DEBUG] switch_core_state_machine.c:364 (sofia/external/79213777785 at 65.98.107.130:5080) State ROUTING
 2011-08-16 13:18:00.336003 [DEBUG] mod_sofia.c:147 sofia/external/79213777785 at 65.98.107.130:5080 SOFIA ROUTING
 2011-08-16 13:18:00.336003 [DEBUG] switch_core_state_machine.c:77 sofia/external/79213777785 at 65.98.107.130:5080 Standard ROUTING
 2011-08-16 13:18:00.336003 [INFO] mod_dialplan_xml.c:331 Processing unknown <79213777785>->inbound_type_uri in context public
 2011-08-16 13:18:00.345157 [ERR] switch_cpp.cpp:1197 start hello.lua
 2011-08-16 13:18:00.345157 [DEBUG] sofia_glue.c:4650 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000]
 2011-08-16 13:18:00.345157 [DEBUG] sofia_glue.c:4650 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
 2011-08-16 13:18:00.345157 [DEBUG] sofia_glue.c:2773 Set Codec sofia/external/79213777785 at 65.98.107.130:5080 PCMU/8000 20 ms 160 samples 64000 bits
 2011-08-16 13:18:00.346166 [DEBUG] sofia_glue.c:4764 Set 2833 dtmf send/recv payload to 101
 2011-08-16 13:18:00.346166 [DEBUG] sofia_glue.c:3014 AUDIO RTP [sofia/external/79213777785 at 65.98.107.130:5080] 65.98.107.130 port 26266 -> 212.232.72.134 port 49276 codec: 0 ms: 20
 2011-08-16 13:18:00.346166 [DEBUG] switch_rtp.c:1623 Starting timer [soft] 160 bytes per 20ms
 2011-08-16 13:18:00.347205 [DEBUG] sofia_glue.c:3276 Set 2833 dtmf send payload to 101
 2011-08-16 13:18:00.347205 [DEBUG] sofia_glue.c:3281 Set 2833 dtmf receive payload to 101
 2011-08-16 13:18:00.347205 [DEBUG] mod_sofia.c:681 Local SDP sofia/external/79213777785 at 65.98.107.130:5080:
 v=0
 o=FreeSWITCH 1313488814 1313488815 IN IP4 65.98.107.130
 s=FreeSWITCH
 c=IN IP4 65.98.107.130
 t=0 0
 m=audio 26266 RTP/AVP 0 101
 a=rtpmap:0 PCMU/8000
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-16
 a=silenceSupp:off - - - -
 a=ptime:20
 a=sendrecv
  
 2011-08-16 13:18:00.348214 [DEBUG] sofia.c:4761 Channel sofia/external/79213777785 at 65.98.107.130:5080 entering state [completed][200]
 2011-08-16 13:18:00.348214 [DEBUG] switch_core_session.c:1939 Application playback Requires media! pre_answering channel sofia/external/79213777785 at 65.98.107.130:5080
 2011-08-16 13:18:00.348214 [DEBUG] switch_cpp.cpp:618 CoreSession::hangup
 2011-08-16 13:18:00.348214 [DEBUG] switch_cpp.cpp:988 sofia/external/79213777785 at 65.98.107.130:5080 destroy/unlink session from object
 2011-08-16 13:18:00.348214 [DEBUG] switch_core_state_machine.c:364 (sofia/external/79213777785 at 65.98.107.130:5080) State ROUTING going to sleep"
  
  
  
 ========================================
  
 logs when I use XML dialplan:
  
 2011-08-16 13:19:49.609765 [DEBUG] switch_core_state_machine.c:371 (sofia/external/79213777785 at 65.98.107.130:5080) State EXECUTE going to sleep
 2011-08-16 13:19:49.609765 [DEBUG] switch_core_state_machine.c:364 (sofia/external/79213777785 at 65.98.107.130:5080) State ROUTING
 2011-08-16 13:19:49.609765 [DEBUG] mod_sofia.c:147 sofia/external/79213777785 at 65.98.107.130:5080 SOFIA ROUTING
 2011-08-16 13:19:49.609765 [DEBUG] switch_core_state_machine.c:77 sofia/external/79213777785 at 65.98.107.130:5080 Standard ROUTING
 2011-08-16 13:19:49.609765 [INFO] mod_dialplan_xml.c:331 Processing unknown <79213777785>->inbound_type_uri in context public
 2011-08-16 13:19:49.615894 [DEBUG] switch_core_state_machine.c:364 (sofia/external/79213777785 at 65.98.107.130:5080) State ROUTING going to sleep
 2011-08-16 13:19:49.615894 [DEBUG] switch_core_state_machine.c:371 (sofia/external/79213777785 at 65.98.107.130:5080) State EXECUTE
 2011-08-16 13:19:49.615894 [DEBUG] mod_sofia.c:240 sofia/external/79213777785 at 65.98.107.130:5080 SOFIA EXECUTE
 2011-08-16 13:19:49.615894 [DEBUG] switch_core_state_machine.c:157 sofia/external/79213777785 at 65.98.107.130:5080 Standard EXECUTE
 2011-08-16 13:19:49.615894 [DEBUG] sofia_glue.c:4650 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000]
 2011-08-16 13:19:49.615894 [DEBUG] sofia_glue.c:4650 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
 2011-08-16 13:19:49.616902 [DEBUG] sofia_glue.c:2773 Set Codec sofia/external/79213777785 at 65.98.107.130:5080 PCMU/8000 20 ms 160 samples 64000 bits
 2011-08-16 13:19:49.616902 [DEBUG] sofia_glue.c:4764 Set 2833 dtmf send/recv payload to 101
 2011-08-16 13:19:49.616902 [DEBUG] sofia_glue.c:3014 AUDIO RTP [sofia/external/79213777785 at 65.98.107.130:5080] 65.98.107.130 port 23946 -> 212.232.72.134 port 49278 codec: 0 ms: 20
 2011-08-16 13:19:49.616902 [DEBUG] switch_rtp.c:1623 Starting timer [soft] 160 bytes per 20ms
 2011-08-16 13:19:49.617998 [DEBUG] sofia_glue.c:3276 Set 2833 dtmf send payload to 101
 2011-08-16 13:19:49.617998 [DEBUG] sofia_glue.c:3281 Set 2833 dtmf receive payload to 101
 2011-08-16 13:19:49.617998 [DEBUG] mod_sofia.c:681 Local SDP sofia/external/79213777785 at 65.98.107.130:5080:
 v=0
 o=FreeSWITCH 1313491243 1313491244 IN IP4 65.98.107.130
 s=FreeSWITCH
 c=IN IP4 65.98.107.130
 t=0 0
 m=audio 23946 RTP/AVP 0 101
 a=rtpmap:0 PCMU/8000
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-16
 a=silenceSupp:off - - - -
 a=ptime:20
 a=sendrecv
  
 2011-08-16 13:19:49.619008 [DEBUG] sofia.c:4761 Channel sofia/external/79213777785 at 65.98.107.130:5080 entering state [completed][200]
 2011-08-16 13:19:49.631344 [DEBUG] switch_ivr_play_say.c:1278 Codec Activated L16 at 8000hz 1 channels 20ms
 2011-08-16 13:19:49.740067 [DEBUG] sofia.c:4761 Channel sofia/external/79213777785 at 65.98.107.130:5080 entering state [ready][200]
 2011-08-16 13:19:52.879234 [DEBUG] switch_ivr_play_say.c:1648 done playing file
 2011-08-16 13:19:52.880333 [NOTICE] switch_core_state_machine.c:189 sofia/external/79213777785 at 65.98.107.130:5080 has executed the last dialplan instruction, hanging up.
  
  
 I have compared logs and saw that case without lua have some strings unlike case with lua.
  
 "2011-08-16 13:19:49.615894 [DEBUG] switch_core_state_machine.c:364 (sofia/external/79213777785 at 65.98.107.130:5080) State ROUTING going to sleep
 2011-08-16 13:19:49.615894 [DEBUG] switch_core_state_machine.c:371 (sofia/external/79213777785 at 65.98.107.130:5080) State EXECUTE
 2011-08-16 13:19:49.615894 [DEBUG] mod_sofia.c:240 sofia/external/79213777785 at 65.98.107.130:5080 SOFIA EXECUTE
 2011-08-16 13:19:49.615894 [DEBUG] switch_core_state_machine.c:157 sofia/external/79213777785 at 65.98.107.130:5080 Standard EXECUTE"
  
  
 Many thanks to anyone who can help. 
  



-------------------------------------------------------------------------------
Valore legale alle tue mail
InterfreePEC - la tua Posta Elettronica Certificata
http://pec.interfree.it
-------------------------------------------------------------------------------



FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org 		 	   		  
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/8f3ccefe/attachment-0001.html 

From x.liu at hw.ac.uk  Wed Aug 17 14:09:22 2011
From: x.liu at hw.ac.uk (xl127)
Date: Wed, 17 Aug 2011 11:09:22 +0100
Subject: [Freeswitch-users] mod_spidermonkey loading error
In-Reply-To: <20110817091244.5582.qmail@community37.interfree.it>
References: <20110817091244.5582.qmail@community37.interfree.it>
Message-ID: <4E4B9352.4010006@hw.ac.uk>

Hi,

I can run the FreeSwitch on CentOS 5 without any problem.
When I run it on Fedora 14 (tried latest git version and latest snapshot 
version), I got following error

011-08-16 18:19:52.928857 [CRIT] switch_loadable_module.c:929 Error 
Loading module /usr/local/freeswitch/mod/mod_spidermonkey.so
**/usr/lib/libldap-2.4.so.2: undefined symbol: PR_GetDirectorySeparator**

I'm stuck here for a while, googled around but didn't figure out a solution.

Any suggestions?

Thanks,

Xing



-- 
Heriot-Watt University is a Scottish charity
registered under charity number SC000278.



From boris at tagnet.ru  Wed Aug 17 14:22:34 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Wed, 17 Aug 2011 16:22:34 +0600
Subject: [Freeswitch-users] DTMF when speaking
Message-ID: <4E4B966A.9070709@tagnet.ru>

Hello!

     Sometimes (very rarely) when I speaking I can hear DTMF signals. So 
the question is - may the FS (proxy_media=false, bypass_media=false) 
generate this DTMFs if no start_dtfm application is used? FS version is 
git-3c74f39 2011-07-22 23-06-06 +0200. Clients devices are different: 
Planet, Linksys, Audiocodes and even Cisco 5350.

-- 
Regards,
Boris




From peter.olsson at visionutveckling.se  Wed Aug 17 14:26:19 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 17 Aug 2011 12:26:19 +0200
Subject: [Freeswitch-users] Lua not playing wav files
In-Reply-To: <SNT127-W13CB7ED823A696374E24B9D5280@phx.gbl>
References: <20110817091244.5582.qmail@community37.interfree.it>
	<SNT127-W13CB7ED823A696374E24B9D5280@phx.gbl>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59EBCDE9F4@cooper>

The problem is that lua script is called before media is up, so you will need to get rid of that inline=true - I don't know how to do that though, since it's a dynamically generated dialplan.

/Peter


Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Yuriy Nasida
Skickat: den 17 augusti 2011 11:54
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Lua not playing wav files

I was try it. There is no differents.


The issue  consists in that I get dialplan automatically through xml_curl + php (by intralanman).
I see it by means xml_curl_debug_on:

<action application="lua" data="$1" inline="true"/>

How can I disable "inline="true"" ?

In the mysql table created under the README from intralanman I see  field "type".  I can put "action" or "anti-action" only. How can I control "inline" parameter?

Many thanks to anyone who can help.
________________________________
Date: Wed, 17 Aug 2011 09:12:44 +0000
From: chrisbware at interfree.it
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Lua not playing wav files


try:



session:streamFile(message)

instead of

session:execute("playback", message)



-----Messaggio originale-----
Da: Yuriy Nasida <nasida at live.ru>
Inviato il: 16 Ago 2011 - 10:46
A: <freeswitch-users at lists.freeswitch.org>


Hi Freeswitch-users,

My simple lua script:

freeswitch.consoleLog("err","start hello.lua\n")
session:answer();
message = "ivr/ivr-enter_destination_telephone_number.wav"
session:execute("playback", message)
session:hangup();

Script looks fine I think, but FS doesn't play audio. If I use corresponding XML dialplan all work fine.


logs when I use lua:

2011-08-16 13:18:00.336003 [DEBUG] switch_core_state_machine.c:371 (sofia/external/79213777785 at 65.98.107.130:5080) State EXECUTE going to sleep
2011-08-16 13:18:00.336003 [DEBUG] switch_core_state_machine.c:364 (sofia/external/79213777785 at 65.98.107.130:5080) State ROUTING
2011-08-16 13:18:00.336003 [DEBUG] mod_sofia.c:147 sofia/external/79213777785 at 65.98.107.130:5080 SOFIA ROUTING
2011-08-16 13:18:00.336003 [DEBUG] switch_core_state_machine.c:77 sofia/external/79213777785 at 65.98.107.130:5080 Standard ROUTING
2011-08-16 13:18:00.336003 [INFO] mod_dialplan_xml.c:331 Processing unknown <79213777785>->inbound_type_uri in context public
2011-08-16 13:18:00.345157 [ERR] switch_cpp.cpp:1197 start hello.lua
2011-08-16 13:18:00.345157 [DEBUG] sofia_glue.c:4650 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000]
2011-08-16 13:18:00.345157 [DEBUG] sofia_glue.c:4650 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
2011-08-16 13:18:00.345157 [DEBUG] sofia_glue.c:2773 Set Codec sofia/external/79213777785 at 65.98.107.130:5080 PCMU/8000 20 ms 160 samples 64000 bits
2011-08-16 13:18:00.346166 [DEBUG] sofia_glue.c:4764 Set 2833 dtmf send/recv payload to 101
2011-08-16 13:18:00.346166 [DEBUG] sofia_glue.c:3014 AUDIO RTP [sofia/external/79213777785 at 65.98.107.130:5080] 65.98.107.130 port 26266 -> 212.232.72.134 port 49276 codec: 0 ms: 20
2011-08-16 13:18:00.346166 [DEBUG] switch_rtp.c:1623 Starting timer [soft] 160 bytes per 20ms
2011-08-16 13:18:00.347205 [DEBUG] sofia_glue.c:3276 Set 2833 dtmf send payload to 101
2011-08-16 13:18:00.347205 [DEBUG] sofia_glue.c:3281 Set 2833 dtmf receive payload to 101
2011-08-16 13:18:00.347205 [DEBUG] mod_sofia.c:681 Local SDP sofia/external/79213777785 at 65.98.107.130:5080:
v=0
o=FreeSWITCH 1313488814 1313488815 IN IP4 65.98.107.130
s=FreeSWITCH
c=IN IP4 65.98.107.130
t=0 0
m=audio 26266 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2011-08-16 13:18:00.348214 [DEBUG] sofia.c:4761 Channel sofia/external/79213777785 at 65.98.107.130:5080 entering state [completed][200]
2011-08-16 13:18:00.348214 [DEBUG] switch_core_session.c:1939 Application playback Requires media! pre_answering channel sofia/external/79213777785 at 65.98.107.130:5080
2011-08-16 13:18:00.348214 [DEBUG] switch_cpp.cpp:618 CoreSession::hangup
2011-08-16 13:18:00.348214 [DEBUG] switch_cpp.cpp:988 sofia/external/79213777785 at 65.98.107.130:5080 destroy/unlink session from object
2011-08-16 13:18:00.348214 [DEBUG] switch_core_state_machine.c:364 (sofia/external/79213777785 at 65.98.107.130:5080) State ROUTING going to sleep"



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

logs when I use XML dialplan:

2011-08-16 13:19:49.609765 [DEBUG] switch_core_state_machine.c:371 (sofia/external/79213777785 at 65.98.107.130:5080) State EXECUTE going to sleep
2011-08-16 13:19:49.609765 [DEBUG] switch_core_state_machine.c:364 (sofia/external/79213777785 at 65.98.107.130:5080) State ROUTING
2011-08-16 13:19:49.609765 [DEBUG] mod_sofia.c:147 sofia/external/79213777785 at 65.98.107.130:5080 SOFIA ROUTING
2011-08-16 13:19:49.609765 [DEBUG] switch_core_state_machine.c:77 sofia/external/79213777785 at 65.98.107.130:5080 Standard ROUTING
2011-08-16 13:19:49.609765 [INFO] mod_dialplan_xml.c:331 Processing unknown <79213777785>->inbound_type_uri in context public
2011-08-16 13:19:49.615894 [DEBUG] switch_core_state_machine.c:364 (sofia/external/79213777785 at 65.98.107.130:5080) State ROUTING going to sleep
2011-08-16 13:19:49.615894 [DEBUG] switch_core_state_machine.c:371 (sofia/external/79213777785 at 65.98.107.130:5080) State EXECUTE
2011-08-16 13:19:49.615894 [DEBUG] mod_sofia.c:240 sofia/external/79213777785 at 65.98.107.130:5080 SOFIA EXECUTE
2011-08-16 13:19:49.615894 [DEBUG] switch_core_state_machine.c:157 sofia/external/79213777785 at 65.98.107.130:5080 Standard EXECUTE
2011-08-16 13:19:49.615894 [DEBUG] sofia_glue.c:4650 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000]
2011-08-16 13:19:49.615894 [DEBUG] sofia_glue.c:4650 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
2011-08-16 13:19:49.616902 [DEBUG] sofia_glue.c:2773 Set Codec sofia/external/79213777785 at 65.98.107.130:5080 PCMU/8000 20 ms 160 samples 64000 bits
2011-08-16 13:19:49.616902 [DEBUG] sofia_glue.c:4764 Set 2833 dtmf send/recv payload to 101
2011-08-16 13:19:49.616902 [DEBUG] sofia_glue.c:3014 AUDIO RTP [sofia/external/79213777785 at 65.98.107.130:5080] 65.98.107.130 port 23946 -> 212.232.72.134 port 49278 codec: 0 ms: 20
2011-08-16 13:19:49.616902 [DEBUG] switch_rtp.c:1623 Starting timer [soft] 160 bytes per 20ms
2011-08-16 13:19:49.617998 [DEBUG] sofia_glue.c:3276 Set 2833 dtmf send payload to 101
2011-08-16 13:19:49.617998 [DEBUG] sofia_glue.c:3281 Set 2833 dtmf receive payload to 101
2011-08-16 13:19:49.617998 [DEBUG] mod_sofia.c:681 Local SDP sofia/external/79213777785 at 65.98.107.130:5080:
v=0
o=FreeSWITCH 1313491243 1313491244 IN IP4 65.98.107.130
s=FreeSWITCH
c=IN IP4 65.98.107.130
t=0 0
m=audio 23946 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2011-08-16 13:19:49.619008 [DEBUG] sofia.c:4761 Channel sofia/external/79213777785 at 65.98.107.130:5080 entering state [completed][200]
2011-08-16 13:19:49.631344 [DEBUG] switch_ivr_play_say.c:1278 Codec Activated L16 at 8000hz 1 channels 20ms
2011-08-16 13:19:49.740067 [DEBUG] sofia.c:4761 Channel sofia/external/79213777785 at 65.98.107.130:5080 entering state [ready][200]
2011-08-16 13:19:52.879234 [DEBUG] switch_ivr_play_say.c:1648 done playing file
2011-08-16 13:19:52.880333 [NOTICE] switch_core_state_machine.c:189 sofia/external/79213777785 at 65.98.107.130:5080 has executed the last dialplan instruction, hanging up.


I have compared logs and saw that case without lua have some strings unlike case with lua.

"2011-08-16 13:19:49.615894 [DEBUG] switch_core_state_machine.c:364 (sofia/external/79213777785 at 65.98.107.130:5080) State ROUTING going to sleep
2011-08-16 13:19:49.615894 [DEBUG] switch_core_state_machine.c:371 (sofia/external/79213777785 at 65.98.107.130:5080) State EXECUTE
2011-08-16 13:19:49.615894 [DEBUG] mod_sofia.c:240 sofia/external/79213777785 at 65.98.107.130:5080 SOFIA EXECUTE
2011-08-16 13:19:49.615894 [DEBUG] switch_core_state_machine.c:157 sofia/external/79213777785 at 65.98.107.130:5080 Standard EXECUTE"


Many thanks to anyone who can help.


-------------------------------------------------------------------------------
Valore legale alle tue mail
InterfreePEC - la tua Posta Elettronica Certificata
http://pec.interfree.it
-------------------------------------------------------------------------------

FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
!DSPAM:4e4b918432761631268876!
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/d04b9e5f/attachment-0001.html 

From peter.olsson at visionutveckling.se  Wed Aug 17 14:32:09 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 17 Aug 2011 12:32:09 +0200
Subject: [Freeswitch-users] DTMF when speaking
In-Reply-To: <4E4B966A.9070709@tagnet.ru>
References: <4E4B966A.9070709@tagnet.ru>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59EBCDE9F7@cooper>

This is probably generated from one of the endpoints, some phones seems to detect some voice as DTMF, and converts it to that.

/Peter


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Boris Kovalenko
Skickat: den 17 augusti 2011 12:23
Till: FreeSWITCH Users Help
?mne: [Freeswitch-users] DTMF when speaking

Hello!

     Sometimes (very rarely) when I speaking I can hear DTMF signals. So 
the question is - may the FS (proxy_media=false, bypass_media=false) 
generate this DTMFs if no start_dtfm application is used? FS version is 
git-3c74f39 2011-07-22 23-06-06 +0200. Clients devices are different: 
Planet, Linksys, Audiocodes and even Cisco 5350.

-- 
Regards,
Boris




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

!DSPAM:4e4b96e232761613586755!



From boris at tagnet.ru  Wed Aug 17 14:39:38 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Wed, 17 Aug 2011 16:39:38 +0600
Subject: [Freeswitch-users] DTMF when speaking
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C59EBCDE9F7@cooper>
References: <4E4B966A.9070709@tagnet.ru>
	<549CFEF87AEDE841A38E9D15EAB4C04C59EBCDE9F7@cooper>
Message-ID: <4E4B9A6A.1050603@tagnet.ru>

I also think so... but... very different devices I use and can't find 
exactly one
> This is probably generated from one of the endpoints, some phones seems to detect some voice as DTMF, and converts it to that.
>
> /Peter
>
>
> -----Ursprungligt meddelande-----
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Boris Kovalenko
> Skickat: den 17 augusti 2011 12:23
> Till: FreeSWITCH Users Help
> ?mne: [Freeswitch-users] DTMF when speaking
>
> Hello!
>
>       Sometimes (very rarely) when I speaking I can hear DTMF signals. So
> the question is - may the FS (proxy_media=false, bypass_media=false)
> generate this DTMFs if no start_dtfm application is used? FS version is
> git-3c74f39 2011-07-22 23-06-06 +0200. Clients devices are different:
> Planet, Linksys, Audiocodes and even Cisco 5350.
>


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   ???. +7 (3435) 230001
   ???? +7 (3435) 230005



From jcgpoza at gmail.com  Wed Aug 17 15:54:23 2011
From: jcgpoza at gmail.com (Dissident)
Date: Wed, 17 Aug 2011 04:54:23 -0700 (PDT)
Subject: [Freeswitch-users] Goip GSM Gateway works great with FreeSwitch!
In-Reply-To: <CALNZuwnvSNaTD4LVwikb71YW4t2aa6=XWYyXaE17oWimrZ4_eQ@mail.gmail.com>
References: <1313496506598-6691087.post@n2.nabble.com>
	<CALNZuwnvSNaTD4LVwikb71YW4t2aa6=XWYyXaE17oWimrZ4_eQ@mail.gmail.com>
Message-ID: <1313582063738-6695132.post@n2.nabble.com>

Hello Marcus, 

here is the link http://wiki.freeswitch.org/wiki/Goip_FreeSwitch_HowTo 

Yes, I think it should be linked from the Interop List.

Best regards.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Goip-GSM-Gateway-works-great-with-FreeSwitch-tp6691087p6695132.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From nasida at live.ru  Wed Aug 17 16:18:03 2011
From: nasida at live.ru (Yuriy Nasida)
Date: Wed, 17 Aug 2011 16:18:03 +0400
Subject: [Freeswitch-users] Lua not playing wav files
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C59EBCDE9F4@cooper>
References: <20110817091244.5582.qmail@community37.interfree.it>,
	<SNT127-W13CB7ED823A696374E24B9D5280@phx.gbl>,
	<549CFEF87AEDE841A38E9D15EAB4C04C59EBCDE9F4@cooper>
Message-ID: <SNT127-W14951F0011911E7A832DFBD5280@phx.gbl>


I see. Anyway thank you.
--Yuriy

From: peter.olsson at visionutveckling.se
To: freeswitch-users at lists.freeswitch.org
Date: Wed, 17 Aug 2011 12:26:19 +0200
Subject: Re: [Freeswitch-users] Lua not playing wav files




The problem is that lua script is called before media is up, so you will need to get rid of that inline=true ? I don?t know how to do that though, since it?s a dynamically generated dialplan. /Peter  Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Yuriy Nasida
Skickat: den 17 augusti 2011 11:54
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Lua not playing wav files I was try it. There is no differents.   The issue  consists in that I get dialplan automatically through xml_curl + php (by intralanman). I see it by means xml_curl_debug_on: <action application="lua" data="$1" inline="true"/> How can I disable "inline="true"" ? In the mysql table created under the README from intralanman I see  field "type".  I can put "action" or "anti-action" only. How can I control "inline" parameter? Many thanks to anyone who can help. Date: Wed, 17 Aug 2011 09:12:44 +0000
From: chrisbware at interfree.it
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Lua not playing wav files

 
try:
 
 
 
session:streamFile(message)
 
instead of
 
session:execute("playback", message)
 
 
 
-----Messaggio originale-----
Da: Yuriy Nasida <nasida at live.ru>
Inviato il: 16 Ago 2011 - 10:46
A: <freeswitch-users at lists.freeswitch.org>


Hi Freeswitch-users,
 
My simple lua script:
 
freeswitch.consoleLog("err","start hello.lua\n")
session:answer();
message = "ivr/ivr-enter_destination_telephone_number.wav"
session:execute("playback", message)
session:hangup();
 
Script looks fine I think, but FS doesn't play audio. If I use corresponding XML dialplan all work fine.
 
 
logs when I use lua:
 
2011-08-16 13:18:00.336003 [DEBUG] switch_core_state_machine.c:371 (sofia/external/79213777785 at 65.98.107.130:5080) State EXECUTE going to sleep
2011-08-16 13:18:00.336003 [DEBUG] switch_core_state_machine.c:364 (sofia/external/79213777785 at 65.98.107.130:5080) State ROUTING
2011-08-16 13:18:00.336003 [DEBUG] mod_sofia.c:147 sofia/external/79213777785 at 65.98.107.130:5080 SOFIA ROUTING
2011-08-16 13:18:00.336003 [DEBUG] switch_core_state_machine.c:77 sofia/external/79213777785 at 65.98.107.130:5080 Standard ROUTING
2011-08-16 13:18:00.336003 [INFO] mod_dialplan_xml.c:331 Processing unknown <79213777785>->inbound_type_uri in context public
2011-08-16 13:18:00.345157 [ERR] switch_cpp.cpp:1197 start hello.lua
2011-08-16 13:18:00.345157 [DEBUG] sofia_glue.c:4650 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000]
2011-08-16 13:18:00.345157 [DEBUG] sofia_glue.c:4650 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
2011-08-16 13:18:00.345157 [DEBUG] sofia_glue.c:2773 Set Codec sofia/external/79213777785 at 65.98.107.130:5080 PCMU/8000 20 ms 160 samples 64000 bits
2011-08-16 13:18:00.346166 [DEBUG] sofia_glue.c:4764 Set 2833 dtmf send/recv payload to 101
2011-08-16 13:18:00.346166 [DEBUG] sofia_glue.c:3014 AUDIO RTP [sofia/external/79213777785 at 65.98.107.130:5080] 65.98.107.130 port 26266 -> 212.232.72.134 port 49276 codec: 0 ms: 20
2011-08-16 13:18:00.346166 [DEBUG] switch_rtp.c:1623 Starting timer [soft] 160 bytes per 20ms
2011-08-16 13:18:00.347205 [DEBUG] sofia_glue.c:3276 Set 2833 dtmf send payload to 101
2011-08-16 13:18:00.347205 [DEBUG] sofia_glue.c:3281 Set 2833 dtmf receive payload to 101
2011-08-16 13:18:00.347205 [DEBUG] mod_sofia.c:681 Local SDP sofia/external/79213777785 at 65.98.107.130:5080:
v=0
o=FreeSWITCH 1313488814 1313488815 IN IP4 65.98.107.130
s=FreeSWITCH
c=IN IP4 65.98.107.130
t=0 0
m=audio 26266 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
 
2011-08-16 13:18:00.348214 [DEBUG] sofia.c:4761 Channel sofia/external/79213777785 at 65.98.107.130:5080 entering state [completed][200]
2011-08-16 13:18:00.348214 [DEBUG] switch_core_session.c:1939 Application playback Requires media! pre_answering channel sofia/external/79213777785 at 65.98.107.130:5080
2011-08-16 13:18:00.348214 [DEBUG] switch_cpp.cpp:618 CoreSession::hangup
2011-08-16 13:18:00.348214 [DEBUG] switch_cpp.cpp:988 sofia/external/79213777785 at 65.98.107.130:5080 destroy/unlink session from object
2011-08-16 13:18:00.348214 [DEBUG] switch_core_state_machine.c:364 (sofia/external/79213777785 at 65.98.107.130:5080) State ROUTING going to sleep"
 
 
 
========================================
 
logs when I use XML dialplan:
 
2011-08-16 13:19:49.609765 [DEBUG] switch_core_state_machine.c:371 (sofia/external/79213777785 at 65.98.107.130:5080) State EXECUTE going to sleep
2011-08-16 13:19:49.609765 [DEBUG] switch_core_state_machine.c:364 (sofia/external/79213777785 at 65.98.107.130:5080) State ROUTING
2011-08-16 13:19:49.609765 [DEBUG] mod_sofia.c:147 sofia/external/79213777785 at 65.98.107.130:5080 SOFIA ROUTING
2011-08-16 13:19:49.609765 [DEBUG] switch_core_state_machine.c:77 sofia/external/79213777785 at 65.98.107.130:5080 Standard ROUTING
2011-08-16 13:19:49.609765 [INFO] mod_dialplan_xml.c:331 Processing unknown <79213777785>->inbound_type_uri in context public
2011-08-16 13:19:49.615894 [DEBUG] switch_core_state_machine.c:364 (sofia/external/79213777785 at 65.98.107.130:5080) State ROUTING going to sleep
2011-08-16 13:19:49.615894 [DEBUG] switch_core_state_machine.c:371 (sofia/external/79213777785 at 65.98.107.130:5080) State EXECUTE
2011-08-16 13:19:49.615894 [DEBUG] mod_sofia.c:240 sofia/external/79213777785 at 65.98.107.130:5080 SOFIA EXECUTE
2011-08-16 13:19:49.615894 [DEBUG] switch_core_state_machine.c:157 sofia/external/79213777785 at 65.98.107.130:5080 Standard EXECUTE
2011-08-16 13:19:49.615894 [DEBUG] sofia_glue.c:4650 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000]
2011-08-16 13:19:49.615894 [DEBUG] sofia_glue.c:4650 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
2011-08-16 13:19:49.616902 [DEBUG] sofia_glue.c:2773 Set Codec sofia/external/79213777785 at 65.98.107.130:5080 PCMU/8000 20 ms 160 samples 64000 bits
2011-08-16 13:19:49.616902 [DEBUG] sofia_glue.c:4764 Set 2833 dtmf send/recv payload to 101
2011-08-16 13:19:49.616902 [DEBUG] sofia_glue.c:3014 AUDIO RTP [sofia/external/79213777785 at 65.98.107.130:5080] 65.98.107.130 port 23946 -> 212.232.72.134 port 49278 codec: 0 ms: 20
2011-08-16 13:19:49.616902 [DEBUG] switch_rtp.c:1623 Starting timer [soft] 160 bytes per 20ms
2011-08-16 13:19:49.617998 [DEBUG] sofia_glue.c:3276 Set 2833 dtmf send payload to 101
2011-08-16 13:19:49.617998 [DEBUG] sofia_glue.c:3281 Set 2833 dtmf receive payload to 101
2011-08-16 13:19:49.617998 [DEBUG] mod_sofia.c:681 Local SDP sofia/external/79213777785 at 65.98.107.130:5080:
v=0
o=FreeSWITCH 1313491243 1313491244 IN IP4 65.98.107.130
s=FreeSWITCH
c=IN IP4 65.98.107.130
t=0 0
m=audio 23946 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
 
2011-08-16 13:19:49.619008 [DEBUG] sofia.c:4761 Channel sofia/external/79213777785 at 65.98.107.130:5080 entering state [completed][200]
2011-08-16 13:19:49.631344 [DEBUG] switch_ivr_play_say.c:1278 Codec Activated L16 at 8000hz 1 channels 20ms
2011-08-16 13:19:49.740067 [DEBUG] sofia.c:4761 Channel sofia/external/79213777785 at 65.98.107.130:5080 entering state [ready][200]
2011-08-16 13:19:52.879234 [DEBUG] switch_ivr_play_say.c:1648 done playing file
2011-08-16 13:19:52.880333 [NOTICE] switch_core_state_machine.c:189 sofia/external/79213777785 at 65.98.107.130:5080 has executed the last dialplan instruction, hanging up.
 
 
I have compared logs and saw that case without lua have some strings unlike case with lua.
 
"2011-08-16 13:19:49.615894 [DEBUG] switch_core_state_machine.c:364 (sofia/external/79213777785 at 65.98.107.130:5080) State ROUTING going to sleep
2011-08-16 13:19:49.615894 [DEBUG] switch_core_state_machine.c:371 (sofia/external/79213777785 at 65.98.107.130:5080) State EXECUTE
2011-08-16 13:19:49.615894 [DEBUG] mod_sofia.c:240 sofia/external/79213777785 at 65.98.107.130:5080 SOFIA EXECUTE
2011-08-16 13:19:49.615894 [DEBUG] switch_core_state_machine.c:157 sofia/external/79213777785 at 65.98.107.130:5080 Standard EXECUTE"
 
 
Many thanks to anyone who can help. 
 
-------------------------------------------------------------------------------
Valore legale alle tue mail
InterfreePEC - la tua Posta Elettronica Certificata
http://pec.interfree.it
-------------------------------------------------------------------------------

FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org!DSPAM:4e4b918432761631268876! 
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org 		 	   		  
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/44bb092b/attachment-0001.html 

From mi.ke at null.net  Wed Aug 17 18:51:31 2011
From: mi.ke at null.net (Mi Ke)
Date: Wed, 17 Aug 2011 14:51:31 +0000
Subject: [Freeswitch-users] getting disconnect cause for a leg after
 bridge in Lua
Message-ID: <20110817145132.12150@gmx.com>

Thanks for replying, Michael.

 Meanwhile if session creation fails, such session can also be considered as disconnected and gone, however its disconnect cause can be read from session:hangupCause(). To my opinion, it would be very useful if disconnect cause could be always readable in Lua after session completion since such mechanism already exists for failed sessions.

 Thanks / Mike



----- Original Message -----
From: Michael Collins
Sent: 08/16/11 03:03 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] getting disconnect cause for a leg after bridge in Lua

 No, you can't do this because the session you are checking is "gone" as soon as the call leg is disconnected. You are better off using a hangup hook or an event socket application if you need to get that value in realtime. Dialplan scripts are good for connecting endpoints and doing simple logic but they are absolutely not what you want for doing any kind of billing or reporting.

 -MC

 On Sat, Aug 13, 2011 at 2:46 AM, Mi Ke < mi.ke at null.net > wrote:
Hi All,

 Is there any way to get a real disconnection cause for leg B in the following script ?


 if (session_a:ready() and session_b:ready()) then

 freeswitch.bridge(session_a,session_b)

 -- session_b gets disconnect here ...

 local session_b_hangup_cause = session_b:hangupCause()



 session_b_hangup_cause is always "SUCCESS" after debridging while log and CDR shows correct value - can get it to my script ?

 Thanks / Mike

 _______________________________________________
 Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com  877-7-4ACLUE

 FreeSWITCH-users mailing list
 FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/dc605ff5/attachment.html 

From bryan at bryanlemon.com  Tue Aug 16 21:07:47 2011
From: bryan at bryanlemon.com (Bryan Lemon)
Date: Tue, 16 Aug 2011 13:07:47 -0400
Subject: [Freeswitch-users] Question about ext-rtp-ip and ext-sip-ip
In-Reply-To: <65727391-DF08-4074-BB7F-BDB766DF7942@freeswitch.org>
References: <CAEjbKNQ3tgWrVrr5uiTVKzv53avDLp3cRoavLuZsOiJfUaW_fg@mail.gmail.com>
	<65727391-DF08-4074-BB7F-BDB766DF7942@freeswitch.org>
Message-ID: <CAEjbKNSGi2prQqoeLSGbKjtYbqOsD7-hggPeEUcVEWPgtCp2UQ@mail.gmail.com>

<profile name="external">
  <gateways>
    <X-PRE-PROCESS cmd="include" data="external/*.xml"/>
  </gateways>

  <aliases>
    <!--


    <alias name="outbound"/>


    <alias name="nat"/>


    -->
  </aliases>

  <domains>
    <domain name="all" alias="false" parse="true"/>
  </domains>

  <settings>
    <param name="debug" value="0"/>
    <param name="sip-trace" value="no"/>
    <param name="rfc2833-pt" value="101"/>
    <param name="sip-port" value="$${external_sip_port}"/>
    <param name="dialplan" value="XML"/>
    <param name="context" value="public"/>
    <param name="dtmf-duration" value="2000"/>
    <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
    <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
    <param name="hold-music" value="$${hold_music}"/>
    <param name="rtp-timer-name" value="soft"/>
    <param name="local-network-acl" value="localnet.auto"/>
    <param name="manage-presence" value="false"/>
    <!--<param name="aggressive-nat-detection" value="true"/>-->
    <param name="inbound-codec-negotiation" value="generous"/>
    <param name="nonce-ttl" value="60"/>
    <param name="auth-calls" value="false"/>
    <param name="rtp-ip" value="$${local_ip_v4}"/>
    <param name="sip-ip" value="$${local_ip_v4}"/>
    <param name="ext-rtp-ip" value="auto-nat"/>
    <param name="ext-sip-ip" value="auto-nat"/>
    <param name="rtp-timeout-sec" value="300"/>
    <param name="rtp-hold-timeout-sec" value="1800"/>
    <param name="tls" value="$${external_ssl_enable}"/>
    <param name="tls-bind-params" value="transport=tls"/>
    <param name="tls-sip-port" value="$${external_tls_port}"/>
    <param name="tls-cert-dir" value="$${external_ssl_dir}"/>
    <param name="tls-version" value="$${sip_tls_version}"/>

  </settings>
</profile>


Thank you,
Bryan Lemon
(302) 648-2747



On Tue, Aug 16, 2011 at 13:03, Brian West <brian at freeswitch.org> wrote:

> Bryan,
>        Can you provide the sofia profile xml?
>
> /b
>
> On Aug 16, 2011, at 8:46 AM, Bryan Lemon wrote:
>
> >> From what I am seeing, freeswitch is not honoring the ext-*-ip variables
> in
> > the invite messages. Using the following command entered on
> > fs_cli: originate
> >
> {origination_caller_id_name='Something.com',origination_caller_id_number=5555551212,userid=7,rowid=ROWID,phonenumber=5555551212,initial=2,prompt=0,thankyou=0,whattosay='',ignore_early_media=true}sofia/gateway/didforsale/15555551212
> > &javascript(somejavascript.js), the invite message is below. Shouldn't
> the
> > instances of 10.0.10.144 be replaced with the ext-*-ip of 204.111.*.*?
> This
> > is causing the rtp packets to be sent to the incorrect location, and
> > resulting in 1-way audio.
> >
> >
> > send 1089 bytes to udp/[209.216.*.*]:5060 at 05:56:00.276988:
> >
> ------------------------------------------------------------------------
> >   INVITE sip:13044150838<
> https://www.google.com/voice/m/caller?number=+13044150838>@209.216.*.*
> > SIP/2.0
> >   Via: SIP/2.0/UDP 10.0.10.144:5080;rport;branch=z9hG4bKH0e2DU1Bc2KgD
> >   Max-Forwards: 69
> >   From: "SomeName" <sip:bryan_bryanlemon_com at 209.216.
> > *.*;transport=udp>;tag=HpU27XSQHmX1g
> >   To: <sip:15555551212 at 209.216.*.*>
> >   Call-ID: 425293d9-426f-122f-8fb5-f04da2846e9a
> >   CSeq: 16401016 INVITE
> >   Contact: <sip:gw+didforsale at 204.111.
> *.*:5080;transport=udp;gw=didforsale>
> >   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-decfdbb 2011-08-11
> 14-15-26
> > -0500
> >   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> > REGISTER, REFER, NOTIFY
> >   Supported: timer, precondition, path, replaces
> >   Allow-Events: talk, hold, refer
> >   Content-Type: application/sdp
> >   Content-Disposition: session
> >   Content-Length: 203
> >   X-FS-Support: update_display
> >   Remote-Party-ID: "SomeName" <sip:15555551212 at 10.0.10.144
> >> ;party=calling;screen=yes;privacy=off
> >
> >   v=0
> >   o=FreeSWITCH 1313441448 1313441449 IN IP4 10.0.10.144
> >   s=FreeSWITCH
> >   c=IN IP4 10.0.10.144
> >   t=0 0
> >   m=audio 32712 RTP/AVP 8 0 3 101 13
> >   a=rtpmap:101 telephone-event/8000
> >   a=fmtp:101 0-16
> >   a=ptime:20
> >
> >
> >
> >
> > sofia status profile internal
> >
> =================================================================================================
> > Name              internal
> > Domain Name       N/A
> > Auto-NAT          true
> > DBName            sofia_reg_internal
> > Pres Hosts        10.0.10.144,10.0.10.144
> > Dialplan          XML
> > Context           public
> > Challenge Realm   auto_from
> > RTP-IP            10.0.10.144
> > Ext-RTP-IP        204.111.*.*
> > SIP-IP            10.0.10.144
> > Ext-SIP-IP        204.111.*.*
> > URL               sip:mod_sofia at 10.0.10.144:5060
> > BIND-URL          sip:mod_sofia at 10.0.10.144:5060
> >
> >
> > freeswitch at internal> sofia status profile external
> >
> =================================================================================================
> > Name              external
> > Domain Name       N/A
> > Auto-NAT          true
> > DBName            sofia_reg_external
> > Pres Hosts
> > Dialplan          XML
> > Context           public
> > Challenge Realm   auto_to
> > RTP-IP            10.0.10.144
> > Ext-RTP-IP        204.111.*.*
> > SIP-IP            10.0.10.144
> > Ext-SIP-IP        204.111.*.*
> > URL               sip:mod_sofia at 10.0.10.144:5080
> > BIND-URL          sip:mod_sofia at 10.0.10.144:5080
> >
> >
> > Thank you,
> > Bryan Lemon
> > (302) 648-2747 <
> https://www.google.com/voice/m/caller?number=+13026482747>
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110816/097289c2/attachment-0001.html 

From SureshM at telesoftlabs.com  Wed Aug 17 09:06:12 2011
From: SureshM at telesoftlabs.com (Suresh M)
Date: Wed, 17 Aug 2011 10:36:12 +0530
Subject: [Freeswitch-users] mod_event_socket Outbound connection in Java.
In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD021FD8A873@NY1-EXMB-01.ip-soft.net>
Message-ID: <81BCC027A3DF104B89991D9E3621F9B42AF347@tslsrv.TSL.local>

Thank you Hector. My C# app is using ManagedESL dll which I compiled
using VC++. I thought I will have a Java version using esl.jar
(freeswitch core) in similar way so that my code migration would be
easier. Anyway, I think better switch to freeswitch-contrib library as
you recommended.
 


________________________________

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Hector Geraldino
Sent: Tuesday, August 16, 2011 11:32 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_event_socket Outbound connection in
Java.



Hi Suresh,

 

I haven't used the C# libraries before, but I can highly recommend you
to use the Java ESL client library  (org.freeswitch.esl.client) that is
listed on the wiki.

 

http://wiki.freeswitch.org/wiki/Java_ESL

 

I'm using it in one of my current developments and it works pretty good.
As it uses the JBoss netty for connection management, you don't have to
worry about connection handling, missing events or anything like that.
It's well designed and the performance is acceptable.

 

Give it a shot and let us know if you have any questions.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Suresh M
Sent: Tuesday, August 16, 2011 4:07 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] mod_event_socket Outbound connection in
Java.

 

Hi,

 

I am trying mod_event_socket in FreeSwitch [outbound ] using Java and I
am stuck:

 

I tried the following C# sample successfully.

 

ESLconnection eslConnection = new
ESLconnection(sckClient.Handle.ToInt32());

 

I need to do the same in Java. But what should I pass to new
ESLconnection constructor in Java in place of socket handle?!  I know
basically this is a Java question for which I tried a lot to find answer
but in vain.  Hope somebody out there would already have come across
this and got a solution.  Any help or clue or alternative method to
achieve the same is greatly appreciated.

 

Thanks in advance.

 

Suresh

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/ecb61d18/attachment-0001.html 

From enp at itx.ru  Wed Aug 17 10:31:36 2011
From: enp at itx.ru (Eugene Prokopiev)
Date: Wed, 17 Aug 2011 10:31:36 +0400
Subject: [Freeswitch-users] SIP/2.0 482 Request merged with
	Huawei-MC820/1.0.0
Message-ID: <CANM4RuhOZEDM5H954oto-UYR3PMF=82LmJ+ZhBm1oiDEjJ0P5Q@mail.gmail.com>

Hi,

I've got SIP/2.0 482 Request merged on INVITE from successfully
registered Huawei-MC820/1.0.0 UA. Conversation looks like:

freeswitch at internal> recv 834 bytes from udp/[10.0.0.102]:5060 at
10:28:55.614664:
   ------------------------------------------------------------------------
   INVITE sip:2101020 at 10.0.0.1 SIP/2.0
   CSeq: 1 INVITE
   Call-ID: 75b760e11a9f9c7b4be7251c4f381d55 at huawei.com
   Contact: <sip:102 at 10.0.0.102>
   Content-Length: 453
   From: <sip:102 at 10.0.0.1>;tag=1728f3d7
   To: <sip:2101020 at 10.0.0.1>
   Via: SIP/2.0/UDP 10.0.0.102;branch=z9hG4bK11fef50f3
   User-Agent: Huawei-MC820/1.0.0
   Supported: 100rel
   Content-Type: application/sdp
   Max-Forwards: 70

   v=0
   o=Huawei 1313566095 1313566095 IN IP4 10.0.0.102
   s=Sip Call
   c=IN IP4 10.0.0.102
   t=0 0
   m=audio 10050 RTP/AVP 8 0 15 18 101
   a=rtpmap:8 PCMA/8000
   a=rtpmap:0 PCMU/8000
   a=rtpmap:15 G728/8000
   a=rtpmap:18 G729/8000
   a=rtpmap:101 telephone-event/8000
   a=sendrecv
   m=video 10052 RTP/AVP 117 34
   b=AS:192
   a=rtpmap:117 H264/90000
   a=rtpmap:34 H263/90000
   a=fmtp:117 profile-level-id=42080D; max-br=194
   a=fmtp:34 CIF=1 QCIF=1 MaxBR=1940
   a=sendrecv
   ------------------------------------------------------------------------
send 298 bytes to udp/[10.0.0.102]:5060 at 10:28:55.615140:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.0.0.102;branch=z9hG4bK11fef50f3
   From: <sip:102 at 10.0.0.1>;tag=1728f3d7
   To: <sip:2101020 at 10.0.0.1>
   Call-ID: 75b760e11a9f9c7b4be7251c4f381d55 at huawei.com
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-hacked-20110816T130734Z
   Content-Length: 0

   ------------------------------------------------------------------------
send 653 bytes to udp/[10.0.0.102]:5060 at 10:28:55.615973:
   ------------------------------------------------------------------------
   SIP/2.0 407 Proxy Authentication Required
   Via: SIP/2.0/UDP 10.0.0.102;branch=z9hG4bK11fef50f3
   From: <sip:102 at 10.0.0.1>;tag=1728f3d7
   To: <sip:2101020 at 10.0.0.1>;tag=U9D95t1myaeNH
   Call-ID: 75b760e11a9f9c7b4be7251c4f381d55 at huawei.com
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-hacked-20110816T130734Z
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, hold, refer
   Proxy-Authenticate: Digest realm="10.0.0.1",
nonce="b5d47364-c8bb-11e0-9704-7d44b5f0bd2d", algorithm=MD5,
qop="auth"
   Content-Length: 0

   ------------------------------------------------------------------------
recv 260 bytes from udp/[10.0.0.102]:5060 at 10:28:55.687752:
   ------------------------------------------------------------------------
   ACK sip:2101020 at 10.0.0.1 SIP/2.0
   CSeq: 1 ACK
   Call-ID: 75b760e11a9f9c7b4be7251c4f381d55 at huawei.com
   Content-Length: 0
   From: <sip:102 at 10.0.0.1>;tag=1728f3d7
   To: <sip:2101020 at 10.0.0.1>;tag=U9D95t1myaeNH
   Via: SIP/2.0/UDP 10.0.0.102;branch=z9hG4bK11fef50f3

   ------------------------------------------------------------------------
recv 1072 bytes from udp/[10.0.0.102]:5060 at 10:28:55.706532:
   ------------------------------------------------------------------------
   INVITE sip:2101020 at 10.0.0.1 SIP/2.0
   CSeq: 1 INVITE
   Call-ID: 75b760e11a9f9c7b4be7251c4f381d55 at huawei.com
   Contact: <sip:102 at 10.0.0.102>
   Content-Length: 453
   From: <sip:102 at 10.0.0.1>;tag=1728f3d7
   To: <sip:2101020 at 10.0.0.1>
   Via: SIP/2.0/UDP 10.0.0.102;branch=z9hG4bK11a279953
   User-Agent: Huawei-MC820/1.0.0
   Supported: 100rel
   Proxy-Authorization: Digest
username="102",realm="10.0.0.1",nonce="b5d47364-c8bb-11e0-9704-7d44b5f0bd2d",uri="sip:2101020 at 10.0.0.1",response="199c0f493e04ab89c6b8ab2dcc1a4ae4",algorithm=MD5,cnonce="6c64988847d3f764",qop=auth,nc=00000001
   Content-Type: application/sdp
   Max-Forwards: 70

   v=0
   o=Huawei 1313566095 1313566096 IN IP4 10.0.0.102
   s=Sip Call
   c=IN IP4 10.0.0.102
   t=0 0
   m=audio 10050 RTP/AVP 8 0 15 18 101
   a=rtpmap:8 PCMA/8000
   a=rtpmap:0 PCMU/8000
   a=rtpmap:15 G728/8000
   a=rtpmap:18 G729/8000
   a=rtpmap:101 telephone-event/8000
   a=sendrecv
   m=video 10052 RTP/AVP 117 34
   b=AS:192
   a=rtpmap:117 H264/90000
   a=rtpmap:34 H263/90000
   a=fmtp:117 profile-level-id=42080D; max-br=194
   a=fmtp:34 CIF=1 QCIF=1 MaxBR=1940
   a=sendrecv
   ------------------------------------------------------------------------
send 257 bytes to udp/[10.0.0.102]:5060 at 10:28:55.706753:
   ------------------------------------------------------------------------
   SIP/2.0 482 Request merged
   Via: SIP/2.0/UDP 10.0.0.102;branch=z9hG4bK11a279953
   From: <sip:102 at 10.0.0.1>;tag=1728f3d7
   To: <sip:2101020 at 10.0.0.1>;tag=U9D95t1myaeNH
   Call-ID: 75b760e11a9f9c7b4be7251c4f381d55 at huawei.com
   CSeq: 1 INVITE
   Content-Length: 0

What is wrong?

--
Thanks,
Eugene Prokopiev


From basit.engg at gmail.com  Wed Aug 17 11:29:57 2011
From: basit.engg at gmail.com (Abdul Basit)
Date: Wed, 17 Aug 2011 12:29:57 +0500
Subject: [Freeswitch-users] Help with dual IP gateways
In-Reply-To: <A6DCFC10-7528-4364-8D4C-9DC4ACD7AD92@mgtech.com>
References: <005501cc5b67$be0ace00$3a206a00$@yahoo.com>
	<A6DCFC10-7528-4364-8D4C-9DC4ACD7AD92@mgtech.com>
Message-ID: <CAGQG5MYoJf2_bO5tCLKrrq4tTSdd2prbWz7EXY4eGkB19cTmHg@mail.gmail.com>

indeed.

On Wed, Aug 17, 2011 at 7:01 AM, Mario G <mario_fs at mgtech.com> wrote:

> I worked a year on this and was planning to put it on the wiki. Does this
> help: I have 2 DSLs, 1 static, 1 dynamic, FS is on a Mac mini with only 1
> nic. The router does dual wan/load balancing. And by golly.... FS can auto
> switch between DSL lines when one has a problem. Wondering if this info is
> worth my time putting on the wiki, I did the OS X page and rewrote other
> stuff and it took weeks. But if there is interest I will do it.
>
> On Aug 15, 2011, at 9:23 AM, Lars Zeb wrote:
>
> > Currently my LAN is connected to the internet via DSL. The FreeSWITCH box
> is
> > on this subnet. To save money, I am moving the data portion of my LAN to
> a
> > new ISP and I want to segregate the VOIP to another ISP. I am tired of
> > having a bad VOIP connection during lengthy downloads.
> >
> > My VOIP and FreeSWITCH skills are minimal. I have used FreeSWITCH for
> over a
> > year in a home/business environment. The only reason it is working is
> with
> > the help of this list.
> >
> > My knowledge of IP is similar. I do not know how to setup a LAN with two
> > gateways with all nodes seeing one another. I do want to be able to call
> out
> > via FreeSWITCH from a softphone on the data portion of the new LAN.
> >
> > A friend suggested I need a dual ported WAN firewall/router with load
> > balancing to enable all the nodes to be on the same subnet. Can anyone
> help
> > me with suggestions? Is there a consultant I can hire to help with this?
> >
> > Thanks, Lars
> >
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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,

Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/97889765/attachment-0001.html 

From tomasz at hyziak.pl  Wed Aug 17 12:40:36 2011
From: tomasz at hyziak.pl (Tomasz Hyziak)
Date: Wed, 17 Aug 2011 10:40:36 +0200
Subject: [Freeswitch-users] Problem with FIFO music and chime
Message-ID: <CAFv7ub1-7K0v5Fe7kLeSv_1NF1GW3m9zv5+bPJSBm6Q0G4Mm0Q@mail.gmail.com>

Hi

I created simple IVR. When caller presses 2, he is redirected to FIFO.
But i've got a problem with fifo_music and fifo_chime_* variables...

When I set only fifo_music - music plays.
When I set fifo_music and fifo_chime_freq (set to 10) and
fifo_chime_list - music play ONLY. There are no chime every 10
seconds.
When I set only fifo_chime_* variables (without fifo_music) - chime
plays every 10 seconds...

I've got no idea why it happend - it was working about 2 weeks ago...

I use FreeSwitch from git (downloaded @ 4 July).

Dialplan:

  <extension name="in_dialog_1">
    <condition field="destination_number" expression="^(hidden)$">
      <action application="set" data="RECORD_STEREO=true"/>
      <action application="record_session"
data="/srv/nagrania/aktualne/IN_${strftime(%Y%m%d-%H%M%S)}_${destination_number}_${caller_id_number}.wav"/>

      <action application="set" data="media_bug_answer_req=true"/>
      <action application="set" data="domain_name=$${domain}"/>
      <action application="export" data="absolute_codec_string=PCMU,PCMA"/>
      <action application="export" data="codec_string=PCMU,PCMA"/>
      <action application="set" data="jitterbuffer_msec=180"/>
      <action application="set" data="hangup_after_bridge=true"/>

      <action application="answer"/>
      <action application="sleep" data="2000"/>

      <action application="set"
data="fifo_chime_list=/srv/nagrania/ivr/prosze_czekac.wav"/>
      <action application="set" data="fifo_chime_freq=10"/>
      <action application="set"
data="fifo_music=/usr/local/freeswitch/sounds/music/8000/ponce-preludio-in-e-major.wav"/>

      <action application="ivr" data="ivr"/>

      <action application="fifo" data="fifo_ogolne in"/>

      <action application="hangup"/>
    </condition>
  </extension>


IVR:

<include>
  <menu name="ivr"
      greet-long="/srv/nagrania/ivr/powitanie_pelne.wav"
      greet-short="/srv/nagrania/ivr/powitanie_skrocone.wav"
      invalid-sound="/srv/nagrania/ivr/zla_opcja.wav"
      exit-sound="/srv/nagrania/ivr/laczenie_z_operatorem.wav"
      confirm-macro="" confirm-key="" tts-engine="flite"
tts-voice="rms" confirm-attempts="2"
      timeout="5000" inter-digit-timeout="2000" max-failures="2"
max-timeouts="2" digit-len="1">

    <entry action="menu-exec-app" digits="1" param="bridge user/1100"/>
    <entry action="menu-exec-app" digits="2" param="fifo fifo_ogolne in"/>
    <entry action="menu-top" digits="9"/>
  </menu>

</include>


FIFO:

<configuration name="fifo.conf" description="FIFO Configuration">
  <settings>
    <param name="delete-all-outbound-member-on-startup" value="false"/>
  </settings>
  <fifos>
    <fifo name="fifo_ogolne" importance="0">
      <member timeout="10" simo="1"
lag="2">{fifo_member_wait=nowait}user/1110</member>
      <member timeout="10" simo="1"
lag="2">{fifo_member_wait=nowait}user/1111</member>
      <member timeout="10" simo="1"
lag="2">{fifo_member_wait=nowait}user/1112</member>
      ...other users...
    </fifo>
  </fifos>
</configuration>


--
Greetings - Tomasz Hyziak


From prashant.lamba at gmail.com  Wed Aug 17 13:34:52 2011
From: prashant.lamba at gmail.com (Prashant Lamba)
Date: Wed, 17 Aug 2011 15:04:52 +0530
Subject: [Freeswitch-users] [Freeswitch-dev] Meetup Paris?
In-Reply-To: <CALXCt0pzErc5beBzNXJ_niDVm+4X5W1CTkV8Ts28vK=hoDD5ww@mail.gmail.com>
References: <E58D1B62-B7E9-4D15-A0AF-DC993AA70786@avgs.ca>
	<CAPHrvXHx-3u0hvQ2WzRZ7M1K5OTO6J-Bur=hC1z5wxXN43p2rA@mail.gmail.com>
	<CALXCt0pzErc5beBzNXJ_niDVm+4X5W1CTkV8Ts28vK=hoDD5ww@mail.gmail.com>
Message-ID: <CAPHrvXG=QQWeu1YHbP-t6X8fpY9c7MCn-bajZou1MOnxkwx_uA@mail.gmail.com>

On Tue, Aug 16, 2011 at 3:23 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

> I have fond memories of a conference I gave couple of years ago in
> Hyderabad and the nice reception has had.
>
> Certainly there are lot of people and companies that can be interested
> in FreeSWITCH, and Cluecon India.
>
> Most people is still on Asterisk, but I believe they'll be very
> interested into knowing first hand what are the advantages of using
> FS, maybe in powerful combination with OpenSips.
>
> Let's have the ball rolling, try to gauge what interest you can
> gather, and which companies can act as sponsors (money matters ;) ).
>
> -giovanni
>


Giovanni, I am all for it especially since there is very little awareness of
FreeSWITCH in India compared to Asterisk. Anyone in?

Prashant, Phonologies (India)
prashant.lamba at gmail.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/fc834b71/attachment-0001.html 

From bryan at bryanlemon.com  Wed Aug 17 19:10:17 2011
From: bryan at bryanlemon.com (Bryan Lemon)
Date: Wed, 17 Aug 2011 11:10:17 -0400
Subject: [Freeswitch-users] Question about ext-rtp-ip and ext-sip-ip
In-Reply-To: <CAEjbKNSGi2prQqoeLSGbKjtYbqOsD7-hggPeEUcVEWPgtCp2UQ@mail.gmail.com>
References: <CAEjbKNQ3tgWrVrr5uiTVKzv53avDLp3cRoavLuZsOiJfUaW_fg@mail.gmail.com>
	<65727391-DF08-4074-BB7F-BDB766DF7942@freeswitch.org>
	<CAEjbKNSGi2prQqoeLSGbKjtYbqOsD7-hggPeEUcVEWPgtCp2UQ@mail.gmail.com>
Message-ID: <CAEjbKNTt8Fd=Q6TnGz407j5N4zfe4X8FSLpf2ttmpUydT9YJpA@mail.gmail.com>

<profile name="external">
  <gateways>
    <X-PRE-PROCESS cmd="include" data="external/*.xml"/>
  </gateways>

  <aliases>
    <!--


    <alias name="outbound"/>


    <alias name="nat"/>


    -->
  </aliases>

  <domains>
    <domain name="all" alias="false" parse="true"/>
  </domains>

  <settings>
    <param name="debug" value="0"/>
    <param name="sip-trace" value="no"/>
    <param name="rfc2833-pt" value="101"/>
    <param name="sip-port" value="$${external_sip_port}"/>
    <param name="dialplan" value="XML"/>
    <param name="context" value="public"/>
    <param name="dtmf-duration" value="2000"/>
    <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
    <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
    <param name="hold-music" value="$${hold_music}"/>
    <param name="rtp-timer-name" value="soft"/>
    <param name="local-network-acl" value="localnet.auto"/>
    <param name="manage-presence" value="false"/>
    <!--<param name="aggressive-nat-detection" value="true"/>-->
    <param name="inbound-codec-negotiation" value="generous"/>
    <param name="nonce-ttl" value="60"/>
    <param name="auth-calls" value="false"/>
    <param name="rtp-ip" value="$${local_ip_v4}"/>
    <param name="sip-ip" value="$${local_ip_v4}"/>
    <param name="ext-rtp-ip" value="auto-nat"/>
    <param name="ext-sip-ip" value="auto-nat"/>
    <param name="rtp-timeout-sec" value="300"/>
    <param name="rtp-hold-timeout-sec" value="1800"/>
    <param name="tls" value="$${external_ssl_enable}"/>
    <param name="tls-bind-params" value="transport=tls"/>
    <param name="tls-sip-port" value="$${external_tls_port}"/>
    <param name="tls-cert-dir" value="$${external_ssl_dir}"/>
    <param name="tls-version" value="$${sip_tls_version}"/>

  </settings>
</profile>


Thank you,
Bryan Lemon
(302) 648-2747



On Tue, Aug 16, 2011 at 13:03, Brian West <brian at freeswitch.org> wrote:

> Bryan,
>        Can you provide the sofia profile xml?
>
> /b
>
> On Aug 16, 2011, at 8:46 AM, Bryan Lemon wrote:
>
> >> From what I am seeing, freeswitch is not honoring the ext-*-ip variables
> in
> > the invite messages. Using the following command entered on
> > fs_cli: originate
> >
> {origination_caller_id_name='Something.com',origination_caller_id_number=5555551212,userid=7,rowid=ROWID,phonenumber=5555551212,initial=2,prompt=0,thankyou=0,whattosay='',ignore_early_media=true}sofia/gateway/didforsale/15555551212
> > &javascript(somejavascript.js), the invite message is below. Shouldn't
> the
> > instances of 10.0.10.144 be replaced with the ext-*-ip of 204.111.*.*?
> This
> > is causing the rtp packets to be sent to the incorrect location, and
> > resulting in 1-way audio.
> >
> >
> > send 1089 bytes to udp/[209.216.*.*]:5060 at 05:56:00.276988:
> >
> ------------------------------------------------------------------------
> >   INVITE sip:13044150838<
> https://www.google.com/voice/m/caller?number=+13044150838>@209.216.*.*
> > SIP/2.0
> >   Via: SIP/2.0/UDP 10.0.10.144:5080;rport;branch=z9hG4bKH0e2DU1Bc2KgD
> >   Max-Forwards: 69
> >   From: "SomeName" <sip:bryan_bryanlemon_com at 209.216.
> > *.*;transport=udp>;tag=HpU27XSQHmX1g
> >   To: <sip:15555551212 at 209.216.*.*>
> >   Call-ID: 425293d9-426f-122f-8fb5-f04da2846e9a
> >   CSeq: 16401016 INVITE
> >   Contact: <sip:gw+didforsale at 204.111.
> *.*:5080;transport=udp;gw=didforsale>
> >   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-decfdbb 2011-08-11
> 14-15-26
> > -0500
> >   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> > REGISTER, REFER, NOTIFY
> >   Supported: timer, precondition, path, replaces
> >   Allow-Events: talk, hold, refer
> >   Content-Type: application/sdp
> >   Content-Disposition: session
> >   Content-Length: 203
> >   X-FS-Support: update_display
> >   Remote-Party-ID: "SomeName" <sip:15555551212 at 10.0.10.144
> >> ;party=calling;screen=yes;privacy=off
> >
> >   v=0
> >   o=FreeSWITCH 1313441448 1313441449 IN IP4 10.0.10.144
> >   s=FreeSWITCH
> >   c=IN IP4 10.0.10.144
> >   t=0 0
> >   m=audio 32712 RTP/AVP 8 0 3 101 13
> >   a=rtpmap:101 telephone-event/8000
> >   a=fmtp:101 0-16
> >   a=ptime:20
> >
> >
> >
> >
> > sofia status profile internal
> >
> =================================================================================================
> > Name              internal
> > Domain Name       N/A
> > Auto-NAT          true
> > DBName            sofia_reg_internal
> > Pres Hosts        10.0.10.144,10.0.10.144
> > Dialplan          XML
> > Context           public
> > Challenge Realm   auto_from
> > RTP-IP            10.0.10.144
> > Ext-RTP-IP        204.111.*.*
> > SIP-IP            10.0.10.144
> > Ext-SIP-IP        204.111.*.*
> > URL               sip:mod_sofia at 10.0.10.144:5060
> > BIND-URL          sip:mod_sofia at 10.0.10.144:5060
> >
> >
> > freeswitch at internal> sofia status profile external
> >
> =================================================================================================
> > Name              external
> > Domain Name       N/A
> > Auto-NAT          true
> > DBName            sofia_reg_external
> > Pres Hosts
> > Dialplan          XML
> > Context           public
> > Challenge Realm   auto_to
> > RTP-IP            10.0.10.144
> > Ext-RTP-IP        204.111.*.*
> > SIP-IP            10.0.10.144
> > Ext-SIP-IP        204.111.*.*
> > URL               sip:mod_sofia at 10.0.10.144:5080
> > BIND-URL          sip:mod_sofia at 10.0.10.144:5080
> >
> >
> > Thank you,
> > Bryan Lemon
> > (302) 648-2747 <
> https://www.google.com/voice/m/caller?number=+13026482747>
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/cb4e6883/attachment-0001.html 

From adrottenberg at gmail.com  Wed Aug 17 19:55:07 2011
From: adrottenberg at gmail.com (Duvid Rottenberg)
Date: Wed, 17 Aug 2011 11:55:07 -0400
Subject: [Freeswitch-users] Paging with Polycom Phones
Message-ID: <CAC4JBEAF4OMFeD9KV5LSAVZ7DxTG5ceK0Vj3cDEam6kU8hTD3Q@mail.gmail.com>

Has anyone successfully implemented paging (auto-answer) with a polycom
phone?
I am using the Conferencing and Intercom sample which sets the
sip_auto_answer variable to true, however on my polycom phone the result is
that the phone rings once and hangs up right away (the phone is sending a
BYE message). I tried adding an Alert-Info header, however it seems that the
Polycom format (Alert-Info: Ring Answer) isn't compliant with the RFC and I
couldn't get freeswitch to send the header in this format.

Has anyone else been able to either get polycom phones to work with
sip_auto_answer or to get freeswitch to send an Alert-Info header in the
polycom format?

Thanks,
Duvid Rottenberg
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/600667e2/attachment.html 

From yungwei at resolvity.com  Wed Aug 17 19:55:36 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Wed, 17 Aug 2011 11:55:36 -0400
Subject: [Freeswitch-users] voicemail doesn't work
In-Reply-To: <CAHKqMb6=znNGaG0pDOj_AbaZM-P1=JOJ2hCH3PDZ0Pqwm5CUEg@mail.gmail.com>
References: <33095823FD21DF429B481B5163264B7950FF12FD15@VMBX102.ihostexchange.net>
	<CAHKqMb6=znNGaG0pDOj_AbaZM-P1=JOJ2hCH3PDZ0Pqwm5CUEg@mail.gmail.com>
Message-ID: <33095823FD21DF429B481B5163264B7950FF130083@VMBX102.ihostexchange.net>

If I change my dialplan to the following, voicemail will work properly. What am I missing here? Thanks.
   <extension name="main_number_2" continue="true">
     <condition field="destination_number" expression="^1112223333$">
<!--
       <action application="set" bluebox="settingEndBridge" data="hangup_after_bridge=true"/>
       <action application="set" bluebox="settingFail" data="continue_on_fail=true"/>
       <action application="set" bluebox="autoPlay" data="vm_auto_play=false"/>
       <action application="set" bluebox="settingTimeout" data="call_timeout=30"/>
       <action application="set" bluebox="settingRing" data="ringback=${us-ring}"/>
       <action application="set" bluebox="settingRingXfer" data="transfer_ringback=${us-ring}"/>
       <action application="export" bluebox="sipCalleeIdName" data="sip_callee_id_name=m1"/>
       <action application="export" bluebox="sipCalleeIdNumber" data="sip_callee_id_number=1112223333"/>
-->
       <action application="answer"/>
       <action application="voicemail" data="default voicemail_2 91000"/>
       <action application="hangup"/>
     </condition>
   </extension>

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nandy Dagondon
Sent: Monday, August 15, 2011 6:52 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] voicemail is not saved

check the directory/file permissions
-nandy
On Tue, Aug 16, 2011 at 3:33 AM, Yungwei Chen wrote:
Hi,

I left several voicemails (Each is longer than 3 sec) to a user account, but none is available when I check the mailbox.
Relevant settings are listed below. What am I missing here? Thanks.

In conf/autoload_configs/modules.conf.xml, mod_voicemail is already loaded.
freeswitch at internal> load mod_voicemail
+OK Reloading XML
-ERR [Module already loaded]
freeswitch at internal> 2011-08-15 14:32:10.666978 [WARNING] switch_loadable_module.c:998 Module mod_voicemail Already Loaded!

Here's the content of conf/autoload_configs/voicemail.conf.xml:
<configuration name="voicemail.conf" description="Voicemail">
 <settings>
 </settings>
 <profiles>
   <profile name="default">
     <param name="file-extension" value="wav"/>
     <param name="terminator-key" value="#"/>
     <param name="max-login-attempts" value="3"/>
     <param name="digit-timeout" value="10000"/>
     <param name="min-record-len" value="3"/>
     <param name="max-record-len" value="300"/>
     <param name="max-retries" value="3"/>
     <param name="tone-spec" value="%(1000, 0, 640)"/>
     <param name="callback-dialplan" value="XML"/>
     <param name="callback-context" value="default"/>
     <param name="play-new-messages-key" value="1"/>
     <param name="play-saved-messages-key" value="2"/>
     <param name="login-keys" value="0"/>
     <param name="main-menu-key" value="0"/>
     <param name="config-menu-key" value="5"/>
     <param name="record-greeting-key" value="1"/>
     <param name="choose-greeting-key" value="2"/>
     <param name="change-pass-key" value="6"/>
     <param name="record-name-key" value="3"/>
     <param name="record-file-key" value="3"/>
     <param name="listen-file-key" value="1"/>
     <param name="save-file-key" value="2"/>
     <param name="delete-file-key" value="7"/>
     <param name="undelete-file-key" value="8"/>
     <param name="email-key" value="4"/>
     <param name="pause-key" value="0"/>
     <param name="restart-key" value="1"/>
     <param name="ff-key" value="6"/>
     <param name="rew-key" value="4"/>
     <param name="skip-greet-key" value="#"/>
     <param name="previous-message-key" value="1"/>
     <param name="next-message-key" value="3"/>
     <param name="skip-info-key" value="*"/>
     <param name="repeat-message-key" value="0"/>
     <param name="record-silence-threshold" value="200"/>
     <param name="record-silence-hits" value="2"/>
     <param name="web-template-file" value="web-vm.tpl"/>
     <!-- if you need to change the sample rate of the recorded files e.g. gmail voicemail player -->
     <!--<param name="record-sample-rate" value="11025"/>-->
     <!-- the next two both must be set for this to be enabled
          the extension is in the format of <dest> [<dialplan>] [<context>]
      -->
     <param name="operator-extension" value="operator XML default"/>
     <param name="operator-key" value="9"/>
     <param name="vmain-extension" value="vmain XML default"/>
     <param name="vmain-key" value="*"/>
     <!-- playback created files as soon as they were recorded by default -->
     <!--<param name="auto-playback-recordings" value="true"/>-->
     <email>
       <param name="template-file" value="voicemail.tpl"/>
       <param name="notify-template-file" value="notify-voicemail.tpl"/>
       <!-- this is the format voicemail_time will have -->
       <param name="date-fmt" value="%A, %B %d %Y, %I %M %p"/>
       <param name="email-from" value="${voicemail_account}@${voicemail_domain}"/>
     </email>
     <!--<param name="storage-dir" value="/tmp"/>-->
     <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
     <!--<param name="record-comment" value="Your Comment"/>-->
     <!--<param name="record-title" value="Your Title"/>-->
     <!--<param name="record-copyright" value="Your Copyright"/>-->
   </profile>
 </profiles>
</configuration>

In conf/directory/default.xml, user 91000 is defined in domain voicemail_2.
 <domain name="voicemail_2">
   <groups>
     <group name="default">
       <users>
         <user bluebox_id="2" id="91000">
           <params>
             <param name="sip-forbid-register" value="true"/>
             <param name="vm-password" value="xxxx"/>
             <param name="vm-message-ext" value="wav"/>
             <param name="vm-mailto" value="xxx at xxx.com<mailto:xxx at xxx.com>"/>
             <param name="email-addr" value="xxx at xxx.com<mailto:xxx at xxx.com>"/>
             <param name="vm-delete-file" value="false"/>
             <param name="vm-attach-file" value="true"/>
             <param name="vm-email-all-messages" value="true"/>
           </params>
           <variables>
             <variable name="timezone" value="America/Chicago"/>
           </variables>
         </user>
       </users>
     </group>
   </groups>
 </domain>

In my dialplan, calls to 1112223333 will be sent to user 91000's voicemail box if they are not answered.
   <extension name="main_number_2" continue="true">
     <condition field="destination_number" expression="^1112223333$">
       <action application="set" bluebox="settingEndBridge" data="hangup_after_bridge=true"/>
       <action application="set" bluebox="settingFail" data="continue_on_fail=true"/>
       <action application="set" bluebox="autoPlay" data="vm_auto_play=false"/>
       <action application="set" bluebox="settingTimeout" data="call_timeout=30"/>
       <action application="set" bluebox="settingRing" data="ringback=${us-ring}"/>
       <action application="set" bluebox="settingRingXfer" data="transfer_ringback=${us-ring}"/>
       <action application="export" bluebox="sipCalleeIdName" data="sip_callee_id_name=m1"/>
       <action application="export" bluebox="sipCalleeIdNumber" data="sip_callee_id_number=1112223333"/>
       <action application="bridge" data="user/91000@$${location_3}"/>
       <action application="answer"/>
       <action application="voicemail" data="default voicemail_2 91000"/>
       <action application="hangup"/>
     </condition>
   </extension>


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

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/27c3153d/attachment-0001.html 

From msc at freeswitch.org  Wed Aug 17 20:14:51 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 17 Aug 2011 09:14:51 -0700
Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
In-Reply-To: <1313543993.57876.YahooMailNeo@web161017.mail.bf1.yahoo.com>
References: <4E4164C0.8030507@tiendalinux.com>
	<1312937649.7702.YahooMailNeo@web161011.mail.bf1.yahoo.com>
	<CAKbxfG_802p+G1a9HtLEh_i-K1C06fpZgyR4_NP=jZJR68YK8g@mail.gmail.com>
	<1313446639.81086.YahooMailNeo@web161008.mail.bf1.yahoo.com>
	<CAKbxfG8jVqBMWODa72nS692buKjeX_LrAT05zz1HeQqg5MM3Yg@mail.gmail.com>
	<1313543993.57876.YahooMailNeo@web161017.mail.bf1.yahoo.com>
Message-ID: <CAKzWOxVMZgQr8JyaHMh=KN93gSxUi=GmTr1jHavYDDB=XZdC9w@mail.gmail.com>

On Tue, Aug 16, 2011 at 6:19 PM, Sam <lakersman2006 at yahoo.com> wrote:

> How come in some of the examples I see it calling answer()?
>
> http://wiki.freeswitch.org/wiki/Perl_Console_IVR_Example
>
>
The above example is a DISA-like function. The *only* way it would work is
for FreeSWITCH to answer the call. It's an IVR, therefore there is no b-leg.

-MC


> ------------------------------
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Tuesday, August 16, 2011 5:29 PM
>
> *Subject:* Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
> You should never answer a call before bridging it anyway, it breaks all of
> the accounting.
> It would make sense to find out why the provider is doing that and get it
> fixed.
>
>
> On Mon, Aug 15, 2011 at 5:17 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> Anthony,
>
> My gripe was not about simply having a DIALSTATUS variable in Freeswitch
> which copies what is from "originate_disposition" what I wanted is to be
> able to get the status of the B-Leg because right now when early media is
> played (which i wanted)  "originate_disposition" shows "ANSWER" which I
> think is caused by me explitly called the "answer" app in my dialplan before
> the bridge app, this is because my DID provider requires an answer/sip 200
> or else it will keep re-sending the sip invite, therefore causing freeswitch
> to keep creating new channels. All I want is to be able to get the proper
> sip/hangup/dial statuses of the B-leg.
>
> ------------------------------
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Wednesday, August 10, 2011 8:52 AM
> *Subject:* Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
> =D
>
> ok, sure.  If that's your only complaint.... see
> commit 9d98d49f0556fb79656c8403f285ae0a615439d3
>
>
>
> Some caveats
>
> 1) There is actually less specific, more generalized data in this
> DIALSTATUS variable than what we already report, when you're ready to move
> on see the originate_disposition variable:  It's kind of like going from
> reporting the precise geo-location of a cafe in Paris to generalizing it to
> "EUROPE"
>
> We follow the Q.850 standard for call cause codes and follow the SIP RFC to
> map sip response codes to/from the Q.850 equivalent.  Also each module has
> its own version "sip_hangup_disposition" for sip so you have both the real
> sip response code AND the official Q.850 equiv variables set on each call.
>
>
> 2) We don't have a torture feature so we never return that code.
>
>
> 3) Since our originate can return before a call is answered I added "EARLY"
> which means the originate succeeded but its still not answered.
>
> 4) For any others that do not map directly to FreeSWITCH, I just installed
> a copy of originate_disposition for good measure.
>
> P.S
>
> This email took longer to compose than the patch did while sitting in the
> middle of a crowded room so you probably could have simply parsed the
> originate originate_disposition yourself but if it helps people get over
> being stuck in a paradigm, it's worth it for me to write........
>
>
> On Tue, Aug 9, 2011 at 7:54 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> I find that Asterisk's AGI is much easier to use, they allow you to
> retrieve the dial status much easier than freeswitch's api's. Come on
> freeswitch, if you want to be better than asterisk, you should make it
> easier to get the dialstatus, etc. At this point asterisk is still defacto.
>
> ------------------------------
> *From:* Nestor A Diaz <nestor at tiendalinux.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Tuesday, August 9, 2011 9:48 AM
> *Subject:* [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
> Hi Guys.
>
> I am starting to use FreeSWITCH, i am an asterisk user since the 1.0.7
> release appears as a package on the debian distribution, at the beginning i
> was amazed by the fact i can build a PBX for my own business and i did,
> later i began to install this system for my customers and sooner i meet the
> problems, however being the software open source i always find a way to fix
> things using patchs from others, sometimes i felt how my life was at risk
> when the system stops working and that usually happens when i have to use
> queues and dealing with digium hardware.
>
> Fixing those problems either by applying patches or by changing the
> hardware where the digium cards were supposed to be installed helps me, but
> that was to much stress for me and seeking for a balance that will let me
> invest more time on services, configuration and hoping to have better
> hardware options brings me to freeswitch.
>
> I agree with freeswitch philosophy that instead of having thousands of
> modules that don't work fine i prefer just a few that works the way it
> should be, a rock solid system for a corporate pbx and a call center is what
> i want.
>
> So here i am trying to begin the conversion, and i hope the information we
> can transcript in this list will help others that want to try another
> alternative to asterisk.
>
> First of all i think the saner for a migration is to have the two systems
> running either on the same machine or different and use the stable features
> of each one.
>
> So could you please freeswitch users help me with this rosetta stone
> migration guide in order to post it to voip-info.org or freeswitch wiki (i
> list only the ones i currently use ):
>
>
>   *Technology* *Asterisk* *Freeswitch*  PSTN Connectivity (Digium /
> Sangoma) dahdi freetdm  IAX2 mod_iax ?? none stable yet.
> Use Asterisk to forward traffic via SIP.
> Enable Hardware HPET for IAX2 trunk if card not available for Asterisk  Bluetooth
> Channel chan_mobile ??
> Use asterisk via SIP
>   Skype Skypeforasterisk (no longer for sale) mod_skypeopen  CDR
> Stadistics Arternic cdr-stats ??  Queue Statistics  Asteriskguru
> queue-stats ??  Web Management Freepbx ??  IVR AGI / AMI Event Socket  Codec
> G.729 Transcodind Cards
> G.729 licenses
> Free G.729 (Intel IPP) Transcodind Cards
> G.729 licenses
> fsg729 Intel IPP(any experience with it ? )  Fax Handling Iaxmodem with
> Hylafax ??
> Iaxmodem via asterisk to FS via SIP ?
>   SIP chan_sip sofia  ACD app_queue mod_callcenter
>
> Thank you all
>
>
> --
> Nestor A. Diaz
> Ingeniero de Sistemas
> Tel. +57 1-485-3020 x 211
> Cel. +57 316-227-3593
> Tel. SIP: sip:211 at tiendalinux.com
> Email/MSN: nestor at tiendalinux.com
> http://www.tiendalinux.com/
> Bogota, Colombia
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/7b93fee9/attachment-0001.html 

From msc at freeswitch.org  Wed Aug 17 20:16:32 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 17 Aug 2011 09:16:32 -0700
Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
In-Reply-To: <1313558080.89178.YahooMailNeo@web161010.mail.bf1.yahoo.com>
References: <4E4164C0.8030507@tiendalinux.com>
	<1312937649.7702.YahooMailNeo@web161011.mail.bf1.yahoo.com>
	<CAKbxfG_802p+G1a9HtLEh_i-K1C06fpZgyR4_NP=jZJR68YK8g@mail.gmail.com>
	<1313446639.81086.YahooMailNeo@web161008.mail.bf1.yahoo.com>
	<CAKbxfG8jVqBMWODa72nS692buKjeX_LrAT05zz1HeQqg5MM3Yg@mail.gmail.com>
	<1313558080.89178.YahooMailNeo@web161010.mail.bf1.yahoo.com>
Message-ID: <CAKzWOxX5rb9BYUAtaW4FjPyq=M1Z9G9QkkdVg+WJS9VMxvDHoQ@mail.gmail.com>

On Tue, Aug 16, 2011 at 10:14 PM, Sam <lakersman2006 at yahoo.com> wrote:

> The DID provider I am using is from iCall, and I was searching through
> their website and noticed that they mentioned a quote with your name on it
> http://carriers.icall.com/open-source/
> so it appears you have had experience with them.
>
> We have a lot of experience with iCall. I'm not familiar with any hard
requirement to "answer" the inbound leg prior to bridging an outbound leg.
What happens in your dialplan if you don't explicitly answer the inbound leg
prior to calling the bridge app?
-MC


> ------------------------------
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Tuesday, August 16, 2011 5:29 PM
> *Subject:* Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
> You should never answer a call before bridging it anyway, it breaks all of
> the accounting.
> It would make sense to find out why the provider is doing that and get it
> fixed.
>
>
> On Mon, Aug 15, 2011 at 5:17 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> Anthony,
>
> My gripe was not about simply having a DIALSTATUS variable in Freeswitch
> which copies what is from "originate_disposition" what I wanted is to be
> able to get the status of the B-Leg because right now when early media is
> played (which i wanted)  "originate_disposition" shows "ANSWER" which I
> think is caused by me explitly called the "answer" app in my dialplan before
> the bridge app, this is because my DID provider requires an answer/sip 200
> or else it will keep re-sending the sip invite, therefore causing freeswitch
> to keep creating new channels. All I want is to be able to get the proper
> sip/hangup/dial statuses of the B-leg.
>
> ------------------------------
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Wednesday, August 10, 2011 8:52 AM
> *Subject:* Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
> =D
>
> ok, sure.  If that's your only complaint.... see
> commit 9d98d49f0556fb79656c8403f285ae0a615439d3
>
>
>
> Some caveats
>
> 1) There is actually less specific, more generalized data in this
> DIALSTATUS variable than what we already report, when you're ready to move
> on see the originate_disposition variable:  It's kind of like going from
> reporting the precise geo-location of a cafe in Paris to generalizing it to
> "EUROPE"
>
> We follow the Q.850 standard for call cause codes and follow the SIP RFC to
> map sip response codes to/from the Q.850 equivalent.  Also each module has
> its own version "sip_hangup_disposition" for sip so you have both the real
> sip response code AND the official Q.850 equiv variables set on each call.
>
>
> 2) We don't have a torture feature so we never return that code.
>
>
> 3) Since our originate can return before a call is answered I added "EARLY"
> which means the originate succeeded but its still not answered.
>
> 4) For any others that do not map directly to FreeSWITCH, I just installed
> a copy of originate_disposition for good measure.
>
> P.S
>
> This email took longer to compose than the patch did while sitting in the
> middle of a crowded room so you probably could have simply parsed the
> originate originate_disposition yourself but if it helps people get over
> being stuck in a paradigm, it's worth it for me to write........
>
>
> On Tue, Aug 9, 2011 at 7:54 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> I find that Asterisk's AGI is much easier to use, they allow you to
> retrieve the dial status much easier than freeswitch's api's. Come on
> freeswitch, if you want to be better than asterisk, you should make it
> easier to get the dialstatus, etc. At this point asterisk is still defacto.
>
> ------------------------------
> *From:* Nestor A Diaz <nestor at tiendalinux.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Tuesday, August 9, 2011 9:48 AM
> *Subject:* [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
> Hi Guys.
>
> I am starting to use FreeSWITCH, i am an asterisk user since the 1.0.7
> release appears as a package on the debian distribution, at the beginning i
> was amazed by the fact i can build a PBX for my own business and i did,
> later i began to install this system for my customers and sooner i meet the
> problems, however being the software open source i always find a way to fix
> things using patchs from others, sometimes i felt how my life was at risk
> when the system stops working and that usually happens when i have to use
> queues and dealing with digium hardware.
>
> Fixing those problems either by applying patches or by changing the
> hardware where the digium cards were supposed to be installed helps me, but
> that was to much stress for me and seeking for a balance that will let me
> invest more time on services, configuration and hoping to have better
> hardware options brings me to freeswitch.
>
> I agree with freeswitch philosophy that instead of having thousands of
> modules that don't work fine i prefer just a few that works the way it
> should be, a rock solid system for a corporate pbx and a call center is what
> i want.
>
> So here i am trying to begin the conversion, and i hope the information we
> can transcript in this list will help others that want to try another
> alternative to asterisk.
>
> First of all i think the saner for a migration is to have the two systems
> running either on the same machine or different and use the stable features
> of each one.
>
> So could you please freeswitch users help me with this rosetta stone
> migration guide in order to post it to voip-info.org or freeswitch wiki (i
> list only the ones i currently use ):
>
>
>   *Technology* *Asterisk* *Freeswitch*  PSTN Connectivity (Digium /
> Sangoma) dahdi freetdm  IAX2 mod_iax ?? none stable yet.
> Use Asterisk to forward traffic via SIP.
> Enable Hardware HPET for IAX2 trunk if card not available for Asterisk  Bluetooth
> Channel chan_mobile ??
> Use asterisk via SIP
>   Skype Skypeforasterisk (no longer for sale) mod_skypeopen  CDR
> Stadistics Arternic cdr-stats ??  Queue Statistics  Asteriskguru
> queue-stats ??  Web Management Freepbx ??  IVR AGI / AMI Event Socket  Codec
> G.729 Transcodind Cards
> G.729 licenses
> Free G.729 (Intel IPP) Transcodind Cards
> G.729 licenses
> fsg729 Intel IPP(any experience with it ? )  Fax Handling Iaxmodem with
> Hylafax ??
> Iaxmodem via asterisk to FS via SIP ?
>   SIP chan_sip sofia  ACD app_queue mod_callcenter
>
> Thank you all
>
>
> --
> Nestor A. Diaz
> Ingeniero de Sistemas
> Tel. +57 1-485-3020 x 211
> Cel. +57 316-227-3593
> Tel. SIP: sip:211 at tiendalinux.com
> Email/MSN: nestor at tiendalinux.com
> http://www.tiendalinux.com/
> Bogota, Colombia
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/a30204b3/attachment-0001.html 

From grsingh750 at gmail.com  Wed Aug 17 20:19:43 2011
From: grsingh750 at gmail.com (guru singh)
Date: Wed, 17 Aug 2011 21:49:43 +0530
Subject: [Freeswitch-users] [Freeswitch-dev] Meetup Paris?
In-Reply-To: <CAPHrvXG=QQWeu1YHbP-t6X8fpY9c7MCn-bajZou1MOnxkwx_uA@mail.gmail.com>
References: <E58D1B62-B7E9-4D15-A0AF-DC993AA70786@avgs.ca>
	<CAPHrvXHx-3u0hvQ2WzRZ7M1K5OTO6J-Bur=hC1z5wxXN43p2rA@mail.gmail.com>
	<CALXCt0pzErc5beBzNXJ_niDVm+4X5W1CTkV8Ts28vK=hoDD5ww@mail.gmail.com>
	<CAPHrvXG=QQWeu1YHbP-t6X8fpY9c7MCn-bajZou1MOnxkwx_uA@mail.gmail.com>
Message-ID: <CAFSQhGWmgZH4QGET7uBujsZvrRsVeubBd+b-W4HL26K99bbKaw@mail.gmail.com>

Hi Prashant,

I'm a FS user based in New Delhi. All the people doing voice related
stuff I've met or exchanged emails with are Asterisk users, some had
not even heard of FS. I'd be interested in trying to get some
initiative/meetup going.

Regards
guru

On Wed, Aug 17, 2011 at 3:04 PM, Prashant Lamba
<prashant.lamba at gmail.com> wrote:
> On Tue, Aug 16, 2011 at 3:23 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>>
>> I have fond memories of a conference I gave couple of years ago in
>> Hyderabad and the nice reception has had.
>>
>> Certainly there are lot of people and companies that can be interested
>> in FreeSWITCH, and Cluecon India.
>>
>> Most people is still on Asterisk, but I believe they'll be very
>> interested into knowing first hand what are the advantages of using
>> FS, maybe in powerful combination with OpenSips.
>>
>> Let's have the ball rolling, try to gauge what interest you can
>> gather, and which companies can act as sponsors (money matters ;) ).
>>
>> -giovanni
>
>
> Giovanni, I am all for it especially since there is very little awareness of
> FreeSWITCH in India compared to Asterisk. Anyone in?
>
> Prashant, Phonologies (India)
> prashant.lamba at gmail.com
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From anthony.minessale at gmail.com  Wed Aug 17 20:34:20 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Aug 2011 11:34:20 -0500
Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
In-Reply-To: <1313558080.89178.YahooMailNeo@web161010.mail.bf1.yahoo.com>
References: <4E4164C0.8030507@tiendalinux.com>
	<1312937649.7702.YahooMailNeo@web161011.mail.bf1.yahoo.com>
	<CAKbxfG_802p+G1a9HtLEh_i-K1C06fpZgyR4_NP=jZJR68YK8g@mail.gmail.com>
	<1313446639.81086.YahooMailNeo@web161008.mail.bf1.yahoo.com>
	<CAKbxfG8jVqBMWODa72nS692buKjeX_LrAT05zz1HeQqg5MM3Yg@mail.gmail.com>
	<1313558080.89178.YahooMailNeo@web161010.mail.bf1.yahoo.com>
Message-ID: <CAKbxfG9OxnYb78L6bv9x+cF240o3=j_7BNNfBzrxct5xAgXJwA@mail.gmail.com>

I asked iCall and they have acknowledged your issue and there is someone
looking into it.


On Wed, Aug 17, 2011 at 12:14 AM, Sam <lakersman2006 at yahoo.com> wrote:

> The DID provider I am using is from iCall, and I was searching through
> their website and noticed that they mentioned a quote with your name on it
> http://carriers.icall.com/open-source/
> so it appears you have had experience with them.
>
> ------------------------------
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Tuesday, August 16, 2011 5:29 PM
> *Subject:* Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
> You should never answer a call before bridging it anyway, it breaks all of
> the accounting.
> It would make sense to find out why the provider is doing that and get it
> fixed.
>
>
> On Mon, Aug 15, 2011 at 5:17 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> Anthony,
>
> My gripe was not about simply having a DIALSTATUS variable in Freeswitch
> which copies what is from "originate_disposition" what I wanted is to be
> able to get the status of the B-Leg because right now when early media is
> played (which i wanted)  "originate_disposition" shows "ANSWER" which I
> think is caused by me explitly called the "answer" app in my dialplan before
> the bridge app, this is because my DID provider requires an answer/sip 200
> or else it will keep re-sending the sip invite, therefore causing freeswitch
> to keep creating new channels. All I want is to be able to get the proper
> sip/hangup/dial statuses of the B-leg.
>
> ------------------------------
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Wednesday, August 10, 2011 8:52 AM
> *Subject:* Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
> =D
>
> ok, sure.  If that's your only complaint.... see
> commit 9d98d49f0556fb79656c8403f285ae0a615439d3
>
>
>
> Some caveats
>
> 1) There is actually less specific, more generalized data in this
> DIALSTATUS variable than what we already report, when you're ready to move
> on see the originate_disposition variable:  It's kind of like going from
> reporting the precise geo-location of a cafe in Paris to generalizing it to
> "EUROPE"
>
> We follow the Q.850 standard for call cause codes and follow the SIP RFC to
> map sip response codes to/from the Q.850 equivalent.  Also each module has
> its own version "sip_hangup_disposition" for sip so you have both the real
> sip response code AND the official Q.850 equiv variables set on each call.
>
>
> 2) We don't have a torture feature so we never return that code.
>
>
> 3) Since our originate can return before a call is answered I added "EARLY"
> which means the originate succeeded but its still not answered.
>
> 4) For any others that do not map directly to FreeSWITCH, I just installed
> a copy of originate_disposition for good measure.
>
> P.S
>
> This email took longer to compose than the patch did while sitting in the
> middle of a crowded room so you probably could have simply parsed the
> originate originate_disposition yourself but if it helps people get over
> being stuck in a paradigm, it's worth it for me to write........
>
>
> On Tue, Aug 9, 2011 at 7:54 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> I find that Asterisk's AGI is much easier to use, they allow you to
> retrieve the dial status much easier than freeswitch's api's. Come on
> freeswitch, if you want to be better than asterisk, you should make it
> easier to get the dialstatus, etc. At this point asterisk is still defacto.
>
> ------------------------------
> *From:* Nestor A Diaz <nestor at tiendalinux.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Tuesday, August 9, 2011 9:48 AM
> *Subject:* [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
> Hi Guys.
>
> I am starting to use FreeSWITCH, i am an asterisk user since the 1.0.7
> release appears as a package on the debian distribution, at the beginning i
> was amazed by the fact i can build a PBX for my own business and i did,
> later i began to install this system for my customers and sooner i meet the
> problems, however being the software open source i always find a way to fix
> things using patchs from others, sometimes i felt how my life was at risk
> when the system stops working and that usually happens when i have to use
> queues and dealing with digium hardware.
>
> Fixing those problems either by applying patches or by changing the
> hardware where the digium cards were supposed to be installed helps me, but
> that was to much stress for me and seeking for a balance that will let me
> invest more time on services, configuration and hoping to have better
> hardware options brings me to freeswitch.
>
> I agree with freeswitch philosophy that instead of having thousands of
> modules that don't work fine i prefer just a few that works the way it
> should be, a rock solid system for a corporate pbx and a call center is what
> i want.
>
> So here i am trying to begin the conversion, and i hope the information we
> can transcript in this list will help others that want to try another
> alternative to asterisk.
>
> First of all i think the saner for a migration is to have the two systems
> running either on the same machine or different and use the stable features
> of each one.
>
> So could you please freeswitch users help me with this rosetta stone
> migration guide in order to post it to voip-info.org or freeswitch wiki (i
> list only the ones i currently use ):
>
>
>   *Technology* *Asterisk* *Freeswitch*  PSTN Connectivity (Digium /
> Sangoma) dahdi freetdm  IAX2 mod_iax ?? none stable yet.
> Use Asterisk to forward traffic via SIP.
> Enable Hardware HPET for IAX2 trunk if card not available for Asterisk  Bluetooth
> Channel chan_mobile ??
> Use asterisk via SIP
>   Skype Skypeforasterisk (no longer for sale) mod_skypeopen  CDR
> Stadistics Arternic cdr-stats ??  Queue Statistics  Asteriskguru
> queue-stats ??  Web Management Freepbx ??  IVR AGI / AMI Event Socket  Codec
> G.729 Transcodind Cards
> G.729 licenses
> Free G.729 (Intel IPP) Transcodind Cards
> G.729 licenses
> fsg729 Intel IPP(any experience with it ? )  Fax Handling Iaxmodem with
> Hylafax ??
> Iaxmodem via asterisk to FS via SIP ?
>   SIP chan_sip sofia  ACD app_queue mod_callcenter
>
> Thank you all
>
>
> --
> Nestor A. Diaz
> Ingeniero de Sistemas
> Tel. +57 1-485-3020 x 211
> Cel. +57 316-227-3593
> Tel. SIP: sip:211 at tiendalinux.com
> Email/MSN: nestor at tiendalinux.com
> http://www.tiendalinux.com/
> Bogota, Colombia
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/91b146f2/attachment-0001.html 

From msc at freeswitch.org  Wed Aug 17 20:48:36 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 17 Aug 2011 09:48:36 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today
Message-ID: <CAKzWOxU47nSthpqKkYPqPmR4X9Dgm5YwKsF0T=UNAEhorbY6jQ@mail.gmail.com>

Hello all!

We are going to have a conf call today and talk about a few things, like how
awesome ClueCon was this year!

http://wiki.freeswitch.org/wiki/FS_weekly_2011_08_17

Come join us and be part of the discussion. I think we might have a few
first-timers involved also.

Thanks,
MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/dae3eafc/attachment.html 

From bnaylor at sirran.com  Wed Aug 17 20:32:12 2011
From: bnaylor at sirran.com (Ben Naylor)
Date: Wed, 17 Aug 2011 17:32:12 +0100
Subject: [Freeswitch-users] Sangoma Media Gateway
Message-ID: <017601cc5cfb$38b5b180$aa211480$@sirran.com>

Hello

 

Apologies if this is in the wrong place, I am a complete beginner at
Freeswitch!

 

Has anyone had much joy with using the above to route calls from ISDN to a
SIP provider?  I have setup the SMG which is a stripped down version of
Freeswitch, but am struggling to work out what to configure to get this
working.

So far I have tried to set up an external gateway to my provider, but this
hasn't appeared in the SMG Web-gui as a SIP profile.  My provider doesn't
required auth by the way, they have just given me an IP to connect to.

 

Any help is greatly appreciated

 

Kind regards

 

Ben

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/2a08980b/attachment.html 

From msc at freeswitch.org  Wed Aug 17 20:51:17 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 17 Aug 2011 09:51:17 -0700
Subject: [Freeswitch-users] Paging with Polycom Phones
In-Reply-To: <CAC4JBEAF4OMFeD9KV5LSAVZ7DxTG5ceK0Vj3cDEam6kU8hTD3Q@mail.gmail.com>
References: <CAC4JBEAF4OMFeD9KV5LSAVZ7DxTG5ceK0Vj3cDEam6kU8hTD3Q@mail.gmail.com>
Message-ID: <CAKzWOxUMbUYM_+eiQG98OpAYTHdur1GVADHzY1OmfV-E+NBHDw@mail.gmail.com>

Try using the "mad boss" example found in the default dialplan. I've tested
that with Polycom phones. It's a nice workaround for all these different
phone vendors who do things so differently.

-MC

On Wed, Aug 17, 2011 at 8:55 AM, Duvid Rottenberg <adrottenberg at gmail.com>wrote:

> Has anyone successfully implemented paging (auto-answer) with a polycom
> phone?
> I am using the Conferencing and Intercom sample which sets the
> sip_auto_answer variable to true, however on my polycom phone the result is
> that the phone rings once and hangs up right away (the phone is sending a
> BYE message). I tried adding an Alert-Info header, however it seems that the
> Polycom format (Alert-Info: Ring Answer) isn't compliant with the RFC and I
> couldn't get freeswitch to send the header in this format.
>
> Has anyone else been able to either get polycom phones to work with
> sip_auto_answer or to get freeswitch to send an Alert-Info header in the
> polycom format?
>
> Thanks,
> Duvid Rottenberg
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/c3d59d7a/attachment.html 

From msc at freeswitch.org  Wed Aug 17 20:53:53 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 17 Aug 2011 09:53:53 -0700
Subject: [Freeswitch-users] Inband DTMF when rfc2833 negotiated?
In-Reply-To: <CALNZuwkD8q5L7ErTw7rLTTRD=ZB4QNmNF_nbyLuhL5cAwPdFmw@mail.gmail.com>
References: <CALNZuwkD8q5L7ErTw7rLTTRD=ZB4QNmNF_nbyLuhL5cAwPdFmw@mail.gmail.com>
Message-ID: <CAKzWOxVOVRvg3_Tu0e1_dh-8aE3p9pp1QcfUxOxve7OECRmSyw@mail.gmail.com>

Which end is sending inband? Do you have a pcap of this call?
-MC

On Tue, Aug 16, 2011 at 5:44 AM, Avi Marcus <avi at avimarcus.net> wrote:

> Why is this call using inband when the SDP says rfc2833? I got complaints
> that the DTMF wasn't working, despite the dtmf numbers in the log being
> correct.
>
> Trace on leg B: http://pastebin.freeswitch.org/17053
>
>
> Thanks,
> -Avi
>
> p.s. I think I've seen this with toll free gateway, too.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/4df78012/attachment.html 

From brian at freeswitch.org  Wed Aug 17 20:55:54 2011
From: brian at freeswitch.org (Brian West)
Date: Wed, 17 Aug 2011 11:55:54 -0500
Subject: [Freeswitch-users] Question about ext-rtp-ip and ext-sip-ip
In-Reply-To: <CAEjbKNTt8Fd=Q6TnGz407j5N4zfe4X8FSLpf2ttmpUydT9YJpA@mail.gmail.com>
References: <CAEjbKNQ3tgWrVrr5uiTVKzv53avDLp3cRoavLuZsOiJfUaW_fg@mail.gmail.com>
	<65727391-DF08-4074-BB7F-BDB766DF7942@freeswitch.org>
	<CAEjbKNSGi2prQqoeLSGbKjtYbqOsD7-hggPeEUcVEWPgtCp2UQ@mail.gmail.com>
	<CAEjbKNTt8Fd=Q6TnGz407j5N4zfe4X8FSLpf2ttmpUydT9YJpA@mail.gmail.com>
Message-ID: <7C7183C2-3601-47D4-B8DE-D9E292B592D3@freeswitch.org>

You need to change these to be "autonat:x.x.x.x"  and specify the IP to use.

/b

On Aug 17, 2011, at 10:10 AM, Bryan Lemon wrote:

>    <param name="ext-rtp-ip" value="auto-nat"/>
>    <param name="ext-sip-ip" value="auto-nat"/>



From anthony.minessale at gmail.com  Wed Aug 17 20:56:50 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Aug 2011 11:56:50 -0500
Subject: [Freeswitch-users] Question about ext-rtp-ip and ext-sip-ip
In-Reply-To: <CAEjbKNSGi2prQqoeLSGbKjtYbqOsD7-hggPeEUcVEWPgtCp2UQ@mail.gmail.com>
References: <CAEjbKNQ3tgWrVrr5uiTVKzv53avDLp3cRoavLuZsOiJfUaW_fg@mail.gmail.com>
	<65727391-DF08-4074-BB7F-BDB766DF7942@freeswitch.org>
	<CAEjbKNSGi2prQqoeLSGbKjtYbqOsD7-hggPeEUcVEWPgtCp2UQ@mail.gmail.com>
Message-ID: <CAKbxfG8H=7EuUHLrpN=5WjyFuECMojTSbTssvjXwJFNBwN1_qg@mail.gmail.com>

if you know the exact value of the ip use that in place of the word
"auto-nat" which is specifically designed for nat-pnp enabled routers.


On Tue, Aug 16, 2011 at 12:07 PM, Bryan Lemon <bryan at bryanlemon.com> wrote:
> <profile name="external">
> ? <gateways>
> ? ? <X-PRE-PROCESS cmd="include" data="external/*.xml"/>
> ? </gateways>
> ? <aliases>
> ? ? <!--
>
>
> ? ? <alias name="outbound"/>
>
>
> ? ? <alias name="nat"/>
>
>
> ? ? -->
> ? </aliases>
> ? <domains>
> ? ? <domain name="all" alias="false" parse="true"/>
> ? </domains>
> ? <settings>
> ? ? <param name="debug" value="0"/>
> ? ? <param name="sip-trace" value="no"/>
> ? ? <param name="rfc2833-pt" value="101"/>
> ? ? <param name="sip-port" value="$${external_sip_port}"/>
> ? ? <param name="dialplan" value="XML"/>
> ? ? <param name="context" value="public"/>
> ? ? <param name="dtmf-duration" value="2000"/>
> ? ? <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
> ? ? <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
> ? ? <param name="hold-music" value="$${hold_music}"/>
> ? ? <param name="rtp-timer-name" value="soft"/>
> ? ? <param name="local-network-acl" value="localnet.auto"/>
> ? ? <param name="manage-presence" value="false"/>
> ? ? <!--<param name="aggressive-nat-detection" value="true"/>-->
> ? ? <param name="inbound-codec-negotiation" value="generous"/>
> ? ? <param name="nonce-ttl" value="60"/>
> ? ? <param name="auth-calls" value="false"/>
> ? ? <param name="rtp-ip" value="$${local_ip_v4}"/>
> ? ? <param name="sip-ip" value="$${local_ip_v4}"/>
> ? ? <param name="ext-rtp-ip" value="auto-nat"/>
> ? ? <param name="ext-sip-ip" value="auto-nat"/>
> ? ? <param name="rtp-timeout-sec" value="300"/>
> ? ? <param name="rtp-hold-timeout-sec" value="1800"/>
> ? ? <param name="tls" value="$${external_ssl_enable}"/>
> ? ? <param name="tls-bind-params" value="transport=tls"/>
> ? ? <param name="tls-sip-port" value="$${external_tls_port}"/>
> ? ? <param name="tls-cert-dir" value="$${external_ssl_dir}"/>
> ? ? <param name="tls-version" value="$${sip_tls_version}"/>
> ? </settings>
> </profile>
>
> Thank you,
> Bryan Lemon
> (302) 648-2747
>
>
>
> On Tue, Aug 16, 2011 at 13:03, Brian West <brian at freeswitch.org> wrote:
>>
>> Bryan,
>> ? ? ? ?Can you provide the sofia profile xml?
>>
>> /b
>>
>> On Aug 16, 2011, at 8:46 AM, Bryan Lemon wrote:
>>
>> >> From what I am seeing, freeswitch is not honoring the ext-*-ip
>> >> variables in
>> > the invite messages. Using the following command entered on
>> > fs_cli: originate
>> >
>> > {origination_caller_id_name='Something.com',origination_caller_id_number=5555551212,userid=7,rowid=ROWID,phonenumber=5555551212,initial=2,prompt=0,thankyou=0,whattosay='',ignore_early_media=true}sofia/gateway/didforsale/15555551212
>> > &javascript(somejavascript.js), the invite message is below. Shouldn't
>> > the
>> > instances of 10.0.10.144 be replaced with the ext-*-ip of 204.111.*.*?
>> > This
>> > is causing the rtp packets to be sent to the incorrect location, and
>> > resulting in 1-way audio.
>> >
>> >
>> > send 1089 bytes to udp/[209.216.*.*]:5060 at 05:56:00.276988:
>> >
>> > ------------------------------------------------------------------------
>> > ? INVITE
>> > sip:13044150838<https://www.google.com/voice/m/caller?number=+13044150838>@209.216.*.*
>> > SIP/2.0
>> > ? Via: SIP/2.0/UDP 10.0.10.144:5080;rport;branch=z9hG4bKH0e2DU1Bc2KgD
>> > ? Max-Forwards: 69
>> > ? From: "SomeName" <sip:bryan_bryanlemon_com at 209.216.
>> > *.*;transport=udp>;tag=HpU27XSQHmX1g
>> > ? To: <sip:15555551212 at 209.216.*.*>
>> > ? Call-ID: 425293d9-426f-122f-8fb5-f04da2846e9a
>> > ? CSeq: 16401016 INVITE
>> > ? Contact:
>> > <sip:gw+didforsale at 204.111.*.*:5080;transport=udp;gw=didforsale>
>> > ? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-decfdbb 2011-08-11
>> > 14-15-26
>> > -0500
>> > ? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> > REGISTER, REFER, NOTIFY
>> > ? Supported: timer, precondition, path, replaces
>> > ? Allow-Events: talk, hold, refer
>> > ? Content-Type: application/sdp
>> > ? Content-Disposition: session
>> > ? Content-Length: 203
>> > ? X-FS-Support: update_display
>> > ? Remote-Party-ID: "SomeName" <sip:15555551212 at 10.0.10.144
>> >> ;party=calling;screen=yes;privacy=off
>> >
>> > ? v=0
>> > ? o=FreeSWITCH 1313441448 1313441449 IN IP4 10.0.10.144
>> > ? s=FreeSWITCH
>> > ? c=IN IP4 10.0.10.144
>> > ? t=0 0
>> > ? m=audio 32712 RTP/AVP 8 0 3 101 13
>> > ? a=rtpmap:101 telephone-event/8000
>> > ? a=fmtp:101 0-16
>> > ? a=ptime:20
>> >
>> >
>> >
>> >
>> > sofia status profile internal
>> >
>> > =================================================================================================
>> > Name ? ? ? ? ? ? ?internal
>> > Domain Name ? ? ? N/A
>> > Auto-NAT ? ? ? ? ?true
>> > DBName ? ? ? ? ? ?sofia_reg_internal
>> > Pres Hosts ? ? ? ?10.0.10.144,10.0.10.144
>> > Dialplan ? ? ? ? ?XML
>> > Context ? ? ? ? ? public
>> > Challenge Realm ? auto_from
>> > RTP-IP ? ? ? ? ? ?10.0.10.144
>> > Ext-RTP-IP ? ? ? ?204.111.*.*
>> > SIP-IP ? ? ? ? ? ?10.0.10.144
>> > Ext-SIP-IP ? ? ? ?204.111.*.*
>> > URL ? ? ? ? ? ? ? sip:mod_sofia at 10.0.10.144:5060
>> > BIND-URL ? ? ? ? ?sip:mod_sofia at 10.0.10.144:5060
>> >
>> >
>> > freeswitch at internal> sofia status profile external
>> >
>> > =================================================================================================
>> > Name ? ? ? ? ? ? ?external
>> > Domain Name ? ? ? N/A
>> > Auto-NAT ? ? ? ? ?true
>> > DBName ? ? ? ? ? ?sofia_reg_external
>> > Pres Hosts
>> > Dialplan ? ? ? ? ?XML
>> > Context ? ? ? ? ? public
>> > Challenge Realm ? auto_to
>> > RTP-IP ? ? ? ? ? ?10.0.10.144
>> > Ext-RTP-IP ? ? ? ?204.111.*.*
>> > SIP-IP ? ? ? ? ? ?10.0.10.144
>> > Ext-SIP-IP ? ? ? ?204.111.*.*
>> > URL ? ? ? ? ? ? ? sip:mod_sofia at 10.0.10.144:5080
>> > BIND-URL ? ? ? ? ?sip:mod_sofia at 10.0.10.144:5080
>> >
>> >
>> > Thank you,
>> > Bryan Lemon
>> > (302) 648-2747
>> > <https://www.google.com/voice/m/caller?number=+13026482747>
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From msc at freeswitch.org  Wed Aug 17 20:59:24 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 17 Aug 2011 09:59:24 -0700
Subject: [Freeswitch-users] Problem with receiving a NOTIFY after
 sending a SUBSCRIBE request
In-Reply-To: <CANSA4prMK6zbUX4xBQdBKkFS1qxuEg0cRacSpsJY=Jn51B_e6w@mail.gmail.com>
References: <CANSA4prMK6zbUX4xBQdBKkFS1qxuEg0cRacSpsJY=Jn51B_e6w@mail.gmail.com>
Message-ID: <CAKzWOxXvx4HSVnMUCvd49sspkgv1NfCtHjXqs74ge2o7Cg0f_w@mail.gmail.com>

This is interesting. I'm looking at the SIP book by Alan Johnston (one of
the guys who authored the SIP spec) and I can't see any indication that the
To: field can contain an invalid URI like that.

Question: why, exactly, does the client feel that it needs to violate such a
basic principle?

-MC

On Tue, Aug 16, 2011 at 7:58 AM, Lorem Ipsum <lrmipsum0 at gmail.com> wrote:

> Hello,
>
> I'm testing a SIP stack for an embedded device. The device, among other
> things, is capable of informing a user about pending messages on the
> voicemail. It does that by subscribing to the message-summary. Below some
> wireshark traces (172.16.30.68 is my device, 172.16.31.10 is FreeSWITCH):
>
> REGISTER sip:172.16.31.10 SIP/2.0
> Via: SIP/2.0/UDP 172.16.30.68:5080
> ;rport;branch=z9hG4bKPj051b000000035ea40edf
> Route: <sip:172.16.31.10:5060;lr>
> Max-Forwards: 70
> From: <sip:399510002 at 172.16.31.10>;tag=051b000000025ea40edf
> To: <sip:399510002 at 172.16.31.10>
> Call-ID: 051b000000015ea40edf
> CSeq: 1 REGISTER
> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY
> User-Agent: My_Sip_Device
> Contact: <sip:399510002 at 172.16.30.68:5080>;transport=udp
> Content-Length:  0
>
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP 172.16.30.68:5080
> ;rport=5080;branch=z9hG4bKPj051b000000035ea40edf
> From: <sip:399510002 at 172.16.31.10>;tag=051b000000025ea40edf
> To: <sip:399510002 at 172.16.31.10>;tag=mQrcXUcvrtUmS
> Call-ID: 051b000000015ea40edf
> CSeq: 1 REGISTER
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-38e3f5f 2011-08-09 03-09-19
> -0400
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> WWW-Authenticate: Digest realm="172.16.31.10",
> nonce="3139f996-c814-11e0-93d2-05aa0ee343d6", algorithm=MD5, qop="auth"
> Content-Length: 0
>
> REGISTER sip:172.16.31.10 SIP/2.0
> Via: SIP/2.0/UDP 172.16.30.68:5080
> ;rport;branch=z9hG4bKPj051b000000065ea40edf
> Route: <sip:172.16.31.10:5060;lr>
> Max-Forwards: 70
> From: <sip:399510002 at 172.16.31.10>;tag=051b000000045ea40edf
> To: <sip:399510002 at 172.16.31.10>
> Call-ID: 051b000000015ea40edf
> CSeq: 2 REGISTER
> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY
> User-Agent: My_Sip_Device
> Contact: <sip:399510002 at 172.16.30.68:5080>;transport=udp
> Authorization: Digest username="399510002", realm="172.16.31.10",
> nonce="3139f996-c814-11e0-93d2-05aa0ee343d6", uri="sip:172.16.31.10",
> response="1e95409a562c074cbe6df148a85107ef", algorithm=MD5,
> cnonce="051b000000055ea40edf", qop=auth, nc=00000001
> Content-Length:  0
>
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 172.16.30.68:5080
> ;rport=5080;branch=z9hG4bKPj051b000000065ea40edf
> From: <sip:399510002 at 172.16.31.10>;tag=051b000000045ea40edf
> To: <sip:399510002 at 172.16.31.10>;tag=N0H5ypXZN3H7m
> Call-ID: 051b000000015ea40edf
> CSeq: 2 REGISTER
> Contact: <sip:399510002 at 172.16.30.68:5080>;transport=udp;expires=180
> Date: Tue, 16 Aug 2011 14:29:47 GMT
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-38e3f5f 2011-08-09 03-09-19
> -0400
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Content-Length: 0
>
>
> SUBSCRIBE sip:172.16.31.10 SIP/2.0
> Via: SIP/2.0/UDP 172.16.30.68:5080
> ;rport;branch=z9hG4bKPj051b0000000a5ea40edf
> Max-Forwards: 69
> From: "399510002" <sip:399510002 at 172.16.31.10>;tag=051b000000085ea40edf
> To: <sip:172.16.31.10>
> Contact: <sip:399510002 at 172.16.30.68:5080>
> Call-ID: 051b000000095ea40edf
> CSeq: 1 SUBSCRIBE
> Event: message-summary
> Accept: application/simple-message-summary
> Allow-Events: message-summary
> User-Agent: My_Sip_Device
> X-Serialnumber: LMZ091218000026
> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY
> Route: <sip:172.16.31.10:5060;lr>
> Content-Length:  0
>
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 172.16.30.68:5080
> ;rport=5080;branch=z9hG4bKPj051b0000000a5ea40edf
> From: "399510002" <sip:399510002 at 172.16.31.10>;tag=051b000000085ea40edf
> To: <sip:172.16.31.10>;tag=p9ay0He3jc8Sg
> Call-ID: 051b000000095ea40edf
> CSeq: 1 SUBSCRIBE
> Contact: <sip:@172.16.31.10:5060>
> Expires: 60
> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-38e3f5f 2011-08-09 03-09-19
> -0400
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
> REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
> Subscription-State: active;expires=60
> Content-Length: 0
>
>
> After sending the 200 OK, FreeSWITCH does not send the NOTIFY.
> If you look at the Contact header of the answer to the SUBSCRIBE you will
> notice that the part before the "@" is missing. I guess this is because
> SUBSCRIBE request does not contain the whole URI, just the host part. That
> is because our customer wants it done this way; the request line should look
> like this:
> SUBSCRIBE sip:voicemail_server SIP/2.0
>
> and the To: header should look like this:
> To: <sip:voicemail_server>
>
> My question is: how can I make FreeSWITCH (the NOTIFY part anyway, other
> things are working OK) work with such a device? Thanks.
>
> Regards,
> Tom
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/7bd19bf8/attachment.html 

From anthony.minessale at gmail.com  Wed Aug 17 21:02:22 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Aug 2011 12:02:22 -0500
Subject: [Freeswitch-users] Problem with FIFO music and chime
In-Reply-To: <CAFv7ub1-7K0v5Fe7kLeSv_1NF1GW3m9zv5+bPJSBm6Q0G4Mm0Q@mail.gmail.com>
References: <CAFv7ub1-7K0v5Fe7kLeSv_1NF1GW3m9zv5+bPJSBm6Q0G4Mm0Q@mail.gmail.com>
Message-ID: <CAKbxfG-GaiSSEzqAZoXADQzR10Ui_jqVw=1LwCmqkUgMjCHEyg@mail.gmail.com>

Can you update again and confirm that is still the case.  We tested it
and it seems to work.


On Wed, Aug 17, 2011 at 3:40 AM, Tomasz Hyziak <tomasz at hyziak.pl> wrote:
> Hi
>
> I created simple IVR. When caller presses 2, he is redirected to FIFO.
> But i've got a problem with fifo_music and fifo_chime_* variables...
>
> When I set only fifo_music - music plays.
> When I set fifo_music and fifo_chime_freq (set to 10) and
> fifo_chime_list - music play ONLY. There are no chime every 10
> seconds.
> When I set only fifo_chime_* variables (without fifo_music) - chime
> plays every 10 seconds...
>
> I've got no idea why it happend - it was working about 2 weeks ago...
>
> I use FreeSwitch from git (downloaded @ 4 July).
>
> Dialplan:
>
> ?<extension name="in_dialog_1">
> ? ?<condition field="destination_number" expression="^(hidden)$">
> ? ? ?<action application="set" data="RECORD_STEREO=true"/>
> ? ? ?<action application="record_session"
> data="/srv/nagrania/aktualne/IN_${strftime(%Y%m%d-%H%M%S)}_${destination_number}_${caller_id_number}.wav"/>
>
> ? ? ?<action application="set" data="media_bug_answer_req=true"/>
> ? ? ?<action application="set" data="domain_name=$${domain}"/>
> ? ? ?<action application="export" data="absolute_codec_string=PCMU,PCMA"/>
> ? ? ?<action application="export" data="codec_string=PCMU,PCMA"/>
> ? ? ?<action application="set" data="jitterbuffer_msec=180"/>
> ? ? ?<action application="set" data="hangup_after_bridge=true"/>
>
> ? ? ?<action application="answer"/>
> ? ? ?<action application="sleep" data="2000"/>
>
> ? ? ?<action application="set"
> data="fifo_chime_list=/srv/nagrania/ivr/prosze_czekac.wav"/>
> ? ? ?<action application="set" data="fifo_chime_freq=10"/>
> ? ? ?<action application="set"
> data="fifo_music=/usr/local/freeswitch/sounds/music/8000/ponce-preludio-in-e-major.wav"/>
>
> ? ? ?<action application="ivr" data="ivr"/>
>
> ? ? ?<action application="fifo" data="fifo_ogolne in"/>
>
> ? ? ?<action application="hangup"/>
> ? ?</condition>
> ?</extension>
>
>
> IVR:
>
> <include>
> ?<menu name="ivr"
> ? ? ?greet-long="/srv/nagrania/ivr/powitanie_pelne.wav"
> ? ? ?greet-short="/srv/nagrania/ivr/powitanie_skrocone.wav"
> ? ? ?invalid-sound="/srv/nagrania/ivr/zla_opcja.wav"
> ? ? ?exit-sound="/srv/nagrania/ivr/laczenie_z_operatorem.wav"
> ? ? ?confirm-macro="" confirm-key="" tts-engine="flite"
> tts-voice="rms" confirm-attempts="2"
> ? ? ?timeout="5000" inter-digit-timeout="2000" max-failures="2"
> max-timeouts="2" digit-len="1">
>
> ? ?<entry action="menu-exec-app" digits="1" param="bridge user/1100"/>
> ? ?<entry action="menu-exec-app" digits="2" param="fifo fifo_ogolne in"/>
> ? ?<entry action="menu-top" digits="9"/>
> ?</menu>
>
> </include>
>
>
> FIFO:
>
> <configuration name="fifo.conf" description="FIFO Configuration">
> ?<settings>
> ? ?<param name="delete-all-outbound-member-on-startup" value="false"/>
> ?</settings>
> ?<fifos>
> ? ?<fifo name="fifo_ogolne" importance="0">
> ? ? ?<member timeout="10" simo="1"
> lag="2">{fifo_member_wait=nowait}user/1110</member>
> ? ? ?<member timeout="10" simo="1"
> lag="2">{fifo_member_wait=nowait}user/1111</member>
> ? ? ?<member timeout="10" simo="1"
> lag="2">{fifo_member_wait=nowait}user/1112</member>
> ? ? ?...other users...
> ? ?</fifo>
> ?</fifos>
> </configuration>
>
>
> --
> Greetings - Tomasz Hyziak
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From msc at freeswitch.org  Wed Aug 17 21:02:41 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 17 Aug 2011 10:02:41 -0700
Subject: [Freeswitch-users] g729 licenses usage
In-Reply-To: <CAKmcL2nwh-Jp_1wAf4+MtUxe0c_PWZbwy1cJzg7pPOb1mT5PAw@mail.gmail.com>
References: <CAKmcL2nwh-Jp_1wAf4+MtUxe0c_PWZbwy1cJzg7pPOb1mT5PAw@mail.gmail.com>
Message-ID: <CAKzWOxWLeLWNA=f87GFbD0_P9BCpGPk+tWueWoZGHVSAs_MgYA@mail.gmail.com>

Yes, you need the extra license since one license includes a single pair of
encoder and decoder.

g.729 is everywhere, however there is hope. The OPUS codec guys are really
pushing that. Stay tuned for an update. We had Jean-Marc Valin from Mozilla
do a presentation about OPUS at ClueCon last week. I will send out notices
when the recording in available.

-MC

On Tue, Aug 16, 2011 at 7:13 AM, Julien Chavanton <jchavanton at gmail.com>wrote:

> Hi,
>
> Before we purchase more g729 licenses we would like to confirm, how many
> licenses are used when you have two call leg (g.729)  bridged with session
> recording ?
>
> I would expect 2 licenses, is this correct ?
>
> ------------------------------------------------------
>
> I am sure I am not alone to hate this kind of licensing limitation, after
> all theses years do we still need g.729, if it did not exist it would be
> replaced by something just as good or better.
>
> I know this as nothing to do with FS, since the market abuse or takeover of
> g.729 is enforced everywhere in the network switches.
>
>
> <freeswitch-users at lists.freeswitch.org>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/87181450/attachment.html 

From lakersman2006 at yahoo.com  Wed Aug 17 21:03:25 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Wed, 17 Aug 2011 10:03:25 -0700 (PDT)
Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
In-Reply-To: <CAKzWOxVMZgQr8JyaHMh=KN93gSxUi=GmTr1jHavYDDB=XZdC9w@mail.gmail.com>
References: <4E4164C0.8030507@tiendalinux.com>
	<1312937649.7702.YahooMailNeo@web161011.mail.bf1.yahoo.com>
	<CAKbxfG_802p+G1a9HtLEh_i-K1C06fpZgyR4_NP=jZJR68YK8g@mail.gmail.com>
	<1313446639.81086.YahooMailNeo@web161008.mail.bf1.yahoo.com>
	<CAKbxfG8jVqBMWODa72nS692buKjeX_LrAT05zz1HeQqg5MM3Yg@mail.gmail.com>
	<1313543993.57876.YahooMailNeo@web161017.mail.bf1.yahoo.com>
	<CAKzWOxVMZgQr8JyaHMh=KN93gSxUi=GmTr1jHavYDDB=XZdC9w@mail.gmail.com>
Message-ID: <1313600605.9925.YahooMailNeo@web161005.mail.bf1.yahoo.com>

My app is basically a calling card app which is kind of like that DISA example, correct?



________________________________
From: Michael Collins <msc at freeswitch.org>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, August 17, 2011 9:14 AM
Subject: Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide





On Tue, Aug 16, 2011 at 6:19 PM, Sam <lakersman2006 at yahoo.com> wrote:

How come in some of the examples I see it calling answer()?
>
>
>http://wiki.freeswitch.org/wiki/Perl_Console_IVR_Example
>
>
>

The above example is a DISA-like function. The *only* way it would work is for FreeSWITCH to answer the call. It's an IVR, therefore there is no b-leg.

-MC
?

>________________________________
>From: Anthony Minessale <anthony.minessale at gmail.com>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>Sent: Tuesday, August 16, 2011 5:29 PM
>
>Subject: Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
>
>
>You should never answer a call before bridging it anyway, it breaks all of the accounting.
>It would make sense to find out why the provider is doing that and get it fixed.
>
>
>
>On Mon, Aug 15, 2011 at 5:17 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
>Anthony,
>>
>>My gripe was not about simply having a DIALSTATUS variable in Freeswitch which copies what is from "originate_disposition" what I wanted is to be able to get the status of the B-Leg because right now when early media is played (which i wanted)? "originate_disposition" shows "ANSWER" which I think is caused by me explitly called the "answer" app in my dialplan before the bridge app, this is because my DID provider requires an answer/sip 200 or else it will keep re-sending the sip invite, therefore causing freeswitch to keep creating new channels. All I want is to be able to get the proper sip/hangup/dial statuses of the B-leg.
>>
>>
>>
>>
>>________________________________
>> From: Anthony Minessale <anthony.minessale at gmail.com>
>>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>Sent: Wednesday, August 10, 2011 8:52 AM
>>Subject: Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>>
>>
>>
>>=D?
>>
>>
>>ok, sure. ?If that's your only complaint.... see commit?9d98d49f0556fb79656c8403f285ae0a615439d3
>>
>>
>>
>>Some caveats
>>
>>
>>1) There is actually less?specific, more generalized data in this DIALSTATUS variable than what we already report, when you're ready to move on see the originate_disposition variable: ?It's kind of like going from reporting the precise geo-location of a cafe in Paris to generalizing it to "EUROPE"?
>>
>>
>>We follow the Q.850 standard for call cause codes and follow the SIP RFC to map sip response codes to/from the Q.850?equivalent. ?Also each module has its own version "sip_hangup_disposition" for sip so you have both the real sip response code AND the official Q.850 equiv variables set on each call.
>>
>>
>>
>>
>>2) We don't have a torture feature so we never return that code.
>>
>>
>>
>>
>>3) Since our originate can return before a call is answered I added "EARLY" which means the originate succeeded but its still not answered.
>>
>>
>>4) For any others that do not map directly to FreeSWITCH, I just installed a copy of originate_disposition for good measure.
>>
>>P.S?
>>
>>
>>This email took longer to compose than the patch did while sitting in the middle of a crowded room so you probably could have simply parsed the originate originate_disposition yourself but if it helps people get over being stuck in a?paradigm, it's worth it for me to write........
>>?
>>
>>
>>On Tue, Aug 9, 2011 at 7:54 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>
>>I find that Asterisk's AGI is much easier to use, they allow you to retrieve the dial status much easier than freeswitch's api's. Come on freeswitch, if you want to be better than asterisk, you should make it easier to get the dialstatus, etc. At this point asterisk is still defacto.
>>>
>>>
>>>
>>>
>>>________________________________
>>> From: Nestor A Diaz <nestor at tiendalinux.com>
>>>To: freeswitch-users at lists.freeswitch.org
>>>Sent: Tuesday, August 9, 2011 9:48 AM
>>>Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>>>
>>>
>>>
>>>Hi Guys.
>>>
>>>I am starting to use FreeSWITCH, i am an asterisk user since the
1.0.7
release appears as a package on the debian distribution, at the
beginning i was amazed by the fact i can build a PBX for my own
business
and i did, later i began to install this system for my customers and
sooner i meet the problems, however being the software open source i
always find a way to fix things using patchs from others, sometimes i
felt how my life was at risk when the system stops working and that
usually happens when i have to use queues and dealing with digium
hardware.
>>>
>>>Fixing those problems either by applying patches or by changing
the
hardware where the digium cards were supposed to be installed helps me,
but that was to much stress for me and seeking for a balance that will
let me invest more time on services, configuration and hoping to have
better hardware
options brings me to freeswitch.
>>>
>>>I agree with freeswitch philosophy that instead of having
thousands of
modules
that don't work fine i prefer just a few that works the way it should
be, a rock solid system for a corporate pbx and a call center is what i
want.
>>>
>>>So here i am trying to begin the conversion, and i hope the
information
we can transcript in this list will help others that want to try
another
alternative to asterisk.
>>>
>>>First of all i think the saner for a migration is to have the two
systems
running either on the same machine or different and use the stable
features of each one.
>>>
>>>So could you please freeswitch users help me with this rosetta
stone migration guide in order to post it to voip-info.org or
freeswitch wiki (i list only the ones i currently use ):
>>>
>>>
>>>
>>>Technology Asterisk Freeswitch 
>>>PSTN Connectivity (Digium /
Sangoma) dahdi freetdm 
>>>IAX2 mod_iax ?? none stable yet.
>>>Use Asterisk to forward traffic via SIP.
>>>Enable Hardware HPET for IAX2 trunk if card not available for Asterisk  
>>>Bluetooth Channel chan_mobile ??
>>>Use asterisk via SIP
>>> 
>>>Skype Skypeforasterisk (no longer for sale) mod_skypeopen 
>>>CDR Stadistics Arternic cdr-stats  ?? 
>>>Queue Statistics  Asteriskguru queue-stats ?? 
>>>Web Management Freepbx ?? 
>>>IVR AGI / AMI Event Socket 
>>>Codec G.729 Transcodind Cards
>>>G.729 licenses
>>>Free G.729 (Intel IPP) Transcodind Cards
>>>G.729 licenses
>>>fsg729 Intel IPP(any experience with it ? ) 
>>>Fax Handling Iaxmodem with Hylafax ??
>>>Iaxmodem via asterisk to FS via SIP ?
>>> 
>>>SIP chan_sip sofia 
>>>ACD app_queue mod_callcenter 
>>>
>>>Thank you all
>>>
>>>
>>>-- 
>>>Nestor A. Diaz
>>>Ingeniero de Sistemas
>>>Tel. +57 1-485-3020 x 211
>>>Cel. +57 316-227-3593
>>>Tel. SIP: sip:211 at tiendalinux.com
>>>Email/MSN: nestor at tiendalinux.com
>>>http://www.tiendalinux.com/
>>>Bogota, Colombia 
>>> 
>>>
>>>
>>>_______________________________________________
>>>Join us at ClueCon 2011, Aug 9-11, Chicago
>>>http://www.cluecon.com 877-7-4ACLUE
>>>
>>>FreeSWITCH-users mailing list
>>>FreeSWITCH-users at lists.freeswitch.org
>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>http://www.freeswitch.org
>>>
>>>
>>>
>>>_______________________________________________
>>>Join us at ClueCon 2011, Aug 9-11, Chicago
>>>http://www.cluecon.com 877-7-4ACLUE
>>>
>>>FreeSWITCH-users mailing list
>>>FreeSWITCH-users at lists.freeswitch.org
>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>http://www.freeswitch.org
>>>
>>>
>>
>>
>>-- 
>>Anthony Minessale II
>>
>>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
>>googletalk:conf+888 at conference.freeswitch.org
>>pstn:+19193869900
>>
>>_______________________________________________
>>Join us at ClueCon 2011, Aug 9-11, Chicago
>>http://www.cluecon.com 877-7-4ACLUE
>>
>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>>
>>
>>
>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>>
>
>
>
>-- 
>Anthony Minessale II
>
>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
>googletalk:conf+888 at conference.freeswitch.org
>pstn:+19193869900
>
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>
>
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>


FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/1716d246/attachment-0001.html 

From lakersman2006 at yahoo.com  Wed Aug 17 21:05:04 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Wed, 17 Aug 2011 10:05:04 -0700 (PDT)
Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
In-Reply-To: <CAKzWOxX5rb9BYUAtaW4FjPyq=M1Z9G9QkkdVg+WJS9VMxvDHoQ@mail.gmail.com>
References: <4E4164C0.8030507@tiendalinux.com>
	<1312937649.7702.YahooMailNeo@web161011.mail.bf1.yahoo.com>
	<CAKbxfG_802p+G1a9HtLEh_i-K1C06fpZgyR4_NP=jZJR68YK8g@mail.gmail.com>
	<1313446639.81086.YahooMailNeo@web161008.mail.bf1.yahoo.com>
	<CAKbxfG8jVqBMWODa72nS692buKjeX_LrAT05zz1HeQqg5MM3Yg@mail.gmail.com>
	<1313558080.89178.YahooMailNeo@web161010.mail.bf1.yahoo.com>
	<CAKzWOxX5rb9BYUAtaW4FjPyq=M1Z9G9QkkdVg+WJS9VMxvDHoQ@mail.gmail.com>
Message-ID: <1313600704.12764.YahooMailNeo@web161010.mail.bf1.yahoo.com>

What happens is iCall will continuously try to resend invites until it timeouts.



________________________________
From: Michael Collins <msc at freeswitch.org>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, August 17, 2011 9:16 AM
Subject: Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide





On Tue, Aug 16, 2011 at 10:14 PM, Sam <lakersman2006 at yahoo.com> wrote:

The DID provider I am using is from iCall, and I was searching through their website and noticed that they mentioned a quote with your name on it http://carriers.icall.com/open-source/
>so it appears you have had experience with them.
>
>
>
We have a lot of experience with iCall. I'm not familiar with any hard requirement to "answer" the inbound leg prior to bridging an outbound leg. What happens in your dialplan if you don't explicitly answer the inbound leg prior to calling the bridge app?
-MC
?

>________________________________
> From: Anthony Minessale <anthony.minessale at gmail.com>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>
>Sent: Tuesday, August 16, 2011 5:29 PM
>
>Subject: Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
>
>
>You should never answer a call before bridging it anyway, it breaks all of the accounting.
>It would make sense to find out why the provider is doing that and get it fixed.
>
>
>
>On Mon, Aug 15, 2011 at 5:17 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
>Anthony,
>>
>>My gripe was not about simply having a DIALSTATUS variable in Freeswitch which copies what is from "originate_disposition" what I wanted is to be able to get the status of the B-Leg because right now when early media is played (which i wanted)? "originate_disposition" shows "ANSWER" which I think is caused by me explitly called the "answer" app in my dialplan before the bridge app, this is because my DID provider requires an answer/sip 200 or else it will keep re-sending the sip invite, therefore causing freeswitch to keep creating new channels. All I want is to be able to get the proper sip/hangup/dial statuses of the B-leg.
>>
>>
>>
>>
>>________________________________
>> From: Anthony Minessale <anthony.minessale at gmail.com>
>>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>Sent: Wednesday, August 10, 2011 8:52 AM
>>Subject: Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>>
>>
>>
>>=D?
>>
>>
>>ok, sure. ?If that's your only complaint.... see commit?9d98d49f0556fb79656c8403f285ae0a615439d3
>>
>>
>>
>>Some caveats
>>
>>
>>1) There is actually less?specific, more generalized data in this DIALSTATUS variable than what we already report, when you're ready to move on see the originate_disposition variable: ?It's kind of like going from reporting the precise geo-location of a cafe in Paris to generalizing it to "EUROPE"?
>>
>>
>>We follow the Q.850 standard for call cause codes and follow the SIP RFC to map sip response codes to/from the Q.850?equivalent. ?Also each module has its own version "sip_hangup_disposition" for sip so you have both the real sip response code AND the official Q.850 equiv variables set on each call.
>>
>>
>>
>>
>>2) We don't have a torture feature so we never return that code.
>>
>>
>>
>>
>>3) Since our originate can return before a call is answered I added "EARLY" which means the originate succeeded but its still not answered.
>>
>>
>>4) For any others that do not map directly to FreeSWITCH, I just installed a copy of originate_disposition for good measure.
>>
>>P.S?
>>
>>
>>This email took longer to compose than the patch did while sitting in the middle of a crowded room so you probably could have simply parsed the originate originate_disposition yourself but if it helps people get over being stuck in a?paradigm, it's worth it for me to write........
>>?
>>
>>
>>On Tue, Aug 9, 2011 at 7:54 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>
>>I find that Asterisk's AGI is much easier to use, they allow you to retrieve the dial status much easier than freeswitch's api's. Come on freeswitch, if you want to be better than asterisk, you should make it easier to get the dialstatus, etc. At this point asterisk is still defacto.
>>>
>>>
>>>
>>>
>>>________________________________
>>> From: Nestor A Diaz <nestor at tiendalinux.com>
>>>To: freeswitch-users at lists.freeswitch.org
>>>Sent: Tuesday, August 9, 2011 9:48 AM
>>>Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>>>
>>>
>>>
>>>Hi Guys.
>>>
>>>I am starting to use FreeSWITCH, i am an asterisk user since the
1.0.7
release appears as a package on the debian distribution, at the
beginning i was amazed by the fact i can build a PBX for my own
business
and i did, later i began to install this system for my customers and
sooner i meet the problems, however being the software open source i
always find a way to fix things using patchs from others, sometimes i
felt how my life was at risk when the system stops working and that
usually happens when i have to use queues and dealing with digium
hardware.
>>>
>>>Fixing those problems either by applying patches or by changing
the
hardware where the digium cards were supposed to be installed helps me,
but that was to much stress for me and seeking for a balance that will
let me invest more time on services, configuration and hoping to have
better hardware
options brings me to freeswitch.
>>>
>>>I agree with freeswitch philosophy that instead of having
thousands of
modules
that don't work fine i prefer just a few that works the way it should
be, a rock solid system for a corporate pbx and a call center is what i
want.
>>>
>>>So here i am trying to begin the conversion, and i hope the
information
we can transcript in this list will help others that want to try
another
alternative to asterisk.
>>>
>>>First of all i think the saner for a migration is to have the two
systems
running either on the same machine or different and use the stable
features of each one.
>>>
>>>So could you please freeswitch users help me with this rosetta
stone migration guide in order to post it to voip-info.org or
freeswitch wiki (i list only the ones i currently use ):
>>>
>>>
>>>
>>>Technology Asterisk Freeswitch 
>>>PSTN Connectivity (Digium /
Sangoma) dahdi freetdm 
>>>IAX2 mod_iax ?? none stable yet.
>>>Use Asterisk to forward traffic via SIP.
>>>Enable Hardware HPET for IAX2 trunk if card not available for Asterisk  
>>>Bluetooth Channel chan_mobile ??
>>>Use asterisk via SIP
>>> 
>>>Skype Skypeforasterisk (no longer for sale) mod_skypeopen 
>>>CDR Stadistics Arternic cdr-stats  ?? 
>>>Queue Statistics  Asteriskguru queue-stats ?? 
>>>Web Management Freepbx ?? 
>>>IVR AGI / AMI Event Socket 
>>>Codec G.729 Transcodind Cards
>>>G.729 licenses
>>>Free G.729 (Intel IPP) Transcodind Cards
>>>G.729 licenses
>>>fsg729 Intel IPP(any experience with it ? ) 
>>>Fax Handling Iaxmodem with Hylafax ??
>>>Iaxmodem via asterisk to FS via SIP ?
>>> 
>>>SIP chan_sip sofia 
>>>ACD app_queue mod_callcenter 
>>>
>>>Thank you all
>>>
>>>
>>>-- 
>>>Nestor A. Diaz
>>>Ingeniero de Sistemas
>>>Tel. +57 1-485-3020 x 211
>>>Cel. +57 316-227-3593
>>>Tel. SIP: sip:211 at tiendalinux.com
>>>Email/MSN: nestor at tiendalinux.com
>>>http://www.tiendalinux.com/
>>>Bogota, Colombia 
>>> 
>>>
>>>
>>>_______________________________________________
>>>Join us at ClueCon 2011, Aug 9-11, Chicago
>>>http://www.cluecon.com 877-7-4ACLUE
>>>
>>>FreeSWITCH-users mailing list
>>>FreeSWITCH-users at lists.freeswitch.org
>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>http://www.freeswitch.org
>>>
>>>
>>>
>>>_______________________________________________
>>>Join us at ClueCon 2011, Aug 9-11, Chicago
>>>http://www.cluecon.com 877-7-4ACLUE
>>>
>>>FreeSWITCH-users mailing list
>>>FreeSWITCH-users at lists.freeswitch.org
>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>http://www.freeswitch.org
>>>
>>>
>>
>>
>>-- 
>>Anthony Minessale II
>>
>>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
>>googletalk:conf+888 at conference.freeswitch.org
>>pstn:+19193869900
>>
>>_______________________________________________
>>Join us at ClueCon 2011, Aug 9-11, Chicago
>>http://www.cluecon.com 877-7-4ACLUE
>>
>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>>
>>
>>
>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>>
>
>
>
>-- 
>Anthony Minessale II
>
>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
>googletalk:conf+888 at conference.freeswitch.org
>pstn:+19193869900
>
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>
>
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>


FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/065e9602/attachment-0001.html 

From lakersman2006 at yahoo.com  Wed Aug 17 21:05:56 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Wed, 17 Aug 2011 10:05:56 -0700 (PDT)
Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
In-Reply-To: <CAKbxfG9OxnYb78L6bv9x+cF240o3=j_7BNNfBzrxct5xAgXJwA@mail.gmail.com>
References: <4E4164C0.8030507@tiendalinux.com>
	<1312937649.7702.YahooMailNeo@web161011.mail.bf1.yahoo.com>
	<CAKbxfG_802p+G1a9HtLEh_i-K1C06fpZgyR4_NP=jZJR68YK8g@mail.gmail.com>
	<1313446639.81086.YahooMailNeo@web161008.mail.bf1.yahoo.com>
	<CAKbxfG8jVqBMWODa72nS692buKjeX_LrAT05zz1HeQqg5MM3Yg@mail.gmail.com>
	<1313558080.89178.YahooMailNeo@web161010.mail.bf1.yahoo.com>
	<CAKbxfG9OxnYb78L6bv9x+cF240o3=j_7BNNfBzrxct5xAgXJwA@mail.gmail.com>
Message-ID: <1313600756.10193.YahooMailNeo@web161011.mail.bf1.yahoo.com>

Okay, wonderful, thanks for the update.



________________________________
From: Anthony Minessale <anthony.minessale at gmail.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, August 17, 2011 9:34 AM
Subject: Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide


I asked iCall and they have acknowledged your issue and there is someone looking into it.



On Wed, Aug 17, 2011 at 12:14 AM, Sam <lakersman2006 at yahoo.com> wrote:

The DID provider I am using is from iCall, and I was searching through their website and noticed that they mentioned a quote with your name on it http://carriers.icall.com/open-source/
>so it appears you have had experience with them.
>
>
>
>
>________________________________
>From: Anthony Minessale <anthony.minessale at gmail.com>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>
>Sent: Tuesday, August 16, 2011 5:29 PM
>
>Subject: Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
>
>
>You should never answer a call before bridging it anyway, it breaks all of the accounting.
>It would make sense to find out why the provider is doing that and get it fixed.
>
>
>
>On Mon, Aug 15, 2011 at 5:17 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
>Anthony,
>>
>>My gripe was not about simply having a DIALSTATUS variable in Freeswitch which copies what is from "originate_disposition" what I wanted is to be able to get the status of the B-Leg because right now when early media is played (which i wanted)? "originate_disposition" shows "ANSWER" which I think is caused by me explitly called the "answer" app in my dialplan before the bridge app, this is because my DID provider requires an answer/sip 200 or else it will keep re-sending the sip invite, therefore causing freeswitch to keep creating new channels. All I want is to be able to get the proper sip/hangup/dial statuses of the B-leg.
>>
>>
>>
>>
>>________________________________
>> From: Anthony Minessale <anthony.minessale at gmail.com>
>>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>Sent: Wednesday, August 10, 2011 8:52 AM
>>Subject: Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>>
>>
>>
>>=D?
>>
>>
>>ok, sure. ?If that's your only complaint.... see commit?9d98d49f0556fb79656c8403f285ae0a615439d3
>>
>>
>>
>>Some caveats
>>
>>
>>1) There is actually less?specific, more generalized data in this DIALSTATUS variable than what we already report, when you're ready to move on see the originate_disposition variable: ?It's kind of like going from reporting the precise geo-location of a cafe in Paris to generalizing it to "EUROPE"?
>>
>>
>>We follow the Q.850 standard for call cause codes and follow the SIP RFC to map sip response codes to/from the Q.850?equivalent. ?Also each module has its own version "sip_hangup_disposition" for sip so you have both the real sip response code AND the official Q.850 equiv variables set on each call.
>>
>>
>>
>>
>>2) We don't have a torture feature so we never return that code.
>>
>>
>>
>>
>>3) Since our originate can return before a call is answered I added "EARLY" which means the originate succeeded but its still not answered.
>>
>>
>>4) For any others that do not map directly to FreeSWITCH, I just installed a copy of originate_disposition for good measure.
>>
>>P.S?
>>
>>
>>This email took longer to compose than the patch did while sitting in the middle of a crowded room so you probably could have simply parsed the originate originate_disposition yourself but if it helps people get over being stuck in a?paradigm, it's worth it for me to write........
>>?
>>
>>
>>On Tue, Aug 9, 2011 at 7:54 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>
>>I find that Asterisk's AGI is much easier to use, they allow you to retrieve the dial status much easier than freeswitch's api's. Come on freeswitch, if you want to be better than asterisk, you should make it easier to get the dialstatus, etc. At this point asterisk is still defacto.
>>>
>>>
>>>
>>>
>>>________________________________
>>> From: Nestor A Diaz <nestor at tiendalinux.com>
>>>To: freeswitch-users at lists.freeswitch.org
>>>Sent: Tuesday, August 9, 2011 9:48 AM
>>>Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>>>
>>>
>>>
>>>Hi Guys.
>>>
>>>I am starting to use FreeSWITCH, i am an asterisk user since the
1.0.7
release appears as a package on the debian distribution, at the
beginning i was amazed by the fact i can build a PBX for my own
business
and i did, later i began to install this system for my customers and
sooner i meet the problems, however being the software open source i
always find a way to fix things using patchs from others, sometimes i
felt how my life was at risk when the system stops working and that
usually happens when i have to use queues and dealing with digium
hardware.
>>>
>>>Fixing those problems either by applying patches or by changing
the
hardware where the digium cards were supposed to be installed helps me,
but that was to much stress for me and seeking for a balance that will
let me invest more time on services, configuration and hoping to have
better hardware
options brings me to freeswitch.
>>>
>>>I agree with freeswitch philosophy that instead of having
thousands of
modules
that don't work fine i prefer just a few that works the way it should
be, a rock solid system for a corporate pbx and a call center is what i
want.
>>>
>>>So here i am trying to begin the conversion, and i hope the
information
we can transcript in this list will help others that want to try
another
alternative to asterisk.
>>>
>>>First of all i think the saner for a migration is to have the two
systems
running either on the same machine or different and use the stable
features of each one.
>>>
>>>So could you please freeswitch users help me with this rosetta
stone migration guide in order to post it to voip-info.org or
freeswitch wiki (i list only the ones i currently use ):
>>>
>>>
>>>
>>>Technology Asterisk Freeswitch 
>>>PSTN Connectivity (Digium /
Sangoma) dahdi freetdm 
>>>IAX2 mod_iax ?? none stable yet.
>>>Use Asterisk to forward traffic via SIP.
>>>Enable Hardware HPET for IAX2 trunk if card not available for Asterisk  
>>>Bluetooth Channel chan_mobile ??
>>>Use asterisk via SIP
>>> 
>>>Skype Skypeforasterisk (no longer for sale) mod_skypeopen 
>>>CDR Stadistics Arternic cdr-stats  ?? 
>>>Queue Statistics  Asteriskguru queue-stats ?? 
>>>Web Management Freepbx ?? 
>>>IVR AGI / AMI Event Socket 
>>>Codec G.729 Transcodind Cards
>>>G.729 licenses
>>>Free G.729 (Intel IPP) Transcodind Cards
>>>G.729 licenses
>>>fsg729 Intel IPP(any experience with it ? ) 
>>>Fax Handling Iaxmodem with Hylafax ??
>>>Iaxmodem via asterisk to FS via SIP ?
>>> 
>>>SIP chan_sip sofia 
>>>ACD app_queue mod_callcenter 
>>>
>>>Thank you all
>>>
>>>
>>>-- 
>>>Nestor A. Diaz
>>>Ingeniero de Sistemas
>>>Tel. +57 1-485-3020 x 211
>>>Cel. +57 316-227-3593
>>>Tel. SIP: sip:211 at tiendalinux.com
>>>Email/MSN: nestor at tiendalinux.com
>>>http://www.tiendalinux.com/
>>>Bogota, Colombia 
>>> 
>>>
>>>
>>>_______________________________________________
>>>Join us at ClueCon 2011, Aug 9-11, Chicago
>>>http://www.cluecon.com 877-7-4ACLUE
>>>
>>>FreeSWITCH-users mailing list
>>>FreeSWITCH-users at lists.freeswitch.org
>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>http://www.freeswitch.org
>>>
>>>
>>>
>>>_______________________________________________
>>>Join us at ClueCon 2011, Aug 9-11, Chicago
>>>http://www.cluecon.com 877-7-4ACLUE
>>>
>>>FreeSWITCH-users mailing list
>>>FreeSWITCH-users at lists.freeswitch.org
>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>http://www.freeswitch.org
>>>
>>>
>>
>>
>>-- 
>>Anthony Minessale II
>>
>>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
>>googletalk:conf+888 at conference.freeswitch.org
>>pstn:+19193869900
>>
>>_______________________________________________
>>Join us at ClueCon 2011, Aug 9-11, Chicago
>>http://www.cluecon.com 877-7-4ACLUE
>>
>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>>
>>
>>
>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>>
>
>
>
>-- 
>Anthony Minessale II
>
>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
>googletalk:conf+888 at conference.freeswitch.org
>pstn:+19193869900
>
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>
>
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/338d79fd/attachment-0001.html 

From anthony.minessale at gmail.com  Wed Aug 17 21:12:15 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Aug 2011 12:12:15 -0500
Subject: [Freeswitch-users] g729 licenses usage
In-Reply-To: <CAKmcL2nwh-Jp_1wAf4+MtUxe0c_PWZbwy1cJzg7pPOb1mT5PAw@mail.gmail.com>
References: <CAKmcL2nwh-Jp_1wAf4+MtUxe0c_PWZbwy1cJzg7pPOb1mT5PAw@mail.gmail.com>
Message-ID: <CAKbxfG9KaTxZX1k6dGEUc=wsKBW0kAfSJRsLd2skR0kb0myDMA@mail.gmail.com>

if you record the ulaw leg you only need one pair.


On Tue, Aug 16, 2011 at 9:13 AM, Julien Chavanton <jchavanton at gmail.com> wrote:
> Hi,
>
> Before we purchase more g729 licenses we would like to confirm, how many
> licenses are used when you have two call leg (g.729)? bridged with session
> recording ?
>
> I would expect 2 licenses, is this correct ?
>
> ------------------------------------------------------
>
> I am sure I am not alone to hate this kind of licensing limitation, after
> all theses years do we still need g.729, if it did not exist it would be
> replaced by something just as good or better.
>
> I know this as nothing to do with FS, since the market abuse or takeover of
> g.729 is enforced everywhere in the network switches.
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From spencer at 5ninesolutions.com  Wed Aug 17 21:21:17 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Wed, 17 Aug 2011 12:21:17 -0500
Subject: [Freeswitch-users] Bad port when using rport
Message-ID: <CFCBEE64-C506-4A74-8203-7F135AEC248A@5ninesolutions.com>

Hello all,
Our FS server is on a public IP on a dedicated server at one location and all clients are behind nat or vpn from other locations.  We have a few "road warrior" clients when our employees travel.  I'm seeing the following from one of the client devices.  I'm not sure of any of the details of the router (or have anyway to control it) but for some reason the rport is sent with an invalid port number.  Notice the 8 at the end of the port number is duplicated.  I'm seeing the same behavior from both a Bria client and Zoiper.  Is there any way to work around this?


An excerpt from Wireshark:

    466 2011-08-17 11:36:04.432912 80.34.4.95            206.125.40.171        SIP      Request: REGISTER sip:x.x.x.x

Internet Protocol, Src: 80.34.4.95 (80.34.4.95), Dst: 206.125.40.171 (x.x.x.x)
User Datagram Protocol, Src Port: 56238 (56238), Dst Port: sip (5060)
Session Initiation Protocol
    Request-Line: REGISTER sip:x.x.x.x SIP/2.0
    Message Header
        Via: SIP/2.0/UDP 80.34.4.95:562388;rport;branch=z9hG4bKPjSAbzV32ywqJ8xWjWLx81Ml6oQcNlrM.s
        Max-Forwards: 70
        From: "XXXX" <sip:1001 at x.x.x.x>;tag=rPFQBJFIKAQpE88mHeiZ7ic59suQY70w
        To: "XXXX" <sip:1001 at x.x.x.x>
        Call-ID: awr-KAMoJpMYEt2mqkOkknkRrCXUj.Jv
        CSeq: 27137 REGISTER
        User-Agent: Bria iPhone 1.2.12
        Contact: "XXXX" <sip:1001 at 80.34.4.95:562388>
        Expires: 900
        Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
        Content-Length:  0

From lakersman2006 at yahoo.com  Wed Aug 17 21:32:02 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Wed, 17 Aug 2011 10:32:02 -0700 (PDT)
Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
In-Reply-To: <CAKzWOxX5rb9BYUAtaW4FjPyq=M1Z9G9QkkdVg+WJS9VMxvDHoQ@mail.gmail.com>
References: <4E4164C0.8030507@tiendalinux.com>
	<1312937649.7702.YahooMailNeo@web161011.mail.bf1.yahoo.com>
	<CAKbxfG_802p+G1a9HtLEh_i-K1C06fpZgyR4_NP=jZJR68YK8g@mail.gmail.com>
	<1313446639.81086.YahooMailNeo@web161008.mail.bf1.yahoo.com>
	<CAKbxfG8jVqBMWODa72nS692buKjeX_LrAT05zz1HeQqg5MM3Yg@mail.gmail.com>
	<1313558080.89178.YahooMailNeo@web161010.mail.bf1.yahoo.com>
	<CAKzWOxX5rb9BYUAtaW4FjPyq=M1Z9G9QkkdVg+WJS9VMxvDHoQ@mail.gmail.com>
Message-ID: <1313602322.16798.YahooMailNeo@web161015.mail.bf1.yahoo.com>

I have also found a side effect when I do not explicitly call answer on the inbound leg for b-leg calls that do not return "answer" when using another DID provider (VOIPInnovations). The side effect is that the a-leg can hear the telco network messages from the carrier like "I'm sorry the number you dialed is not a working number ..." or "The user is not accepting calls at the moment."


If I do explicitly call answer, then I cannot hear those telco messages, which would seem to be better fitting for my case.



________________________________
From: Michael Collins <msc at freeswitch.org>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, August 17, 2011 9:16 AM
Subject: Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide





On Tue, Aug 16, 2011 at 10:14 PM, Sam <lakersman2006 at yahoo.com> wrote:

The DID provider I am using is from iCall, and I was searching through their website and noticed that they mentioned a quote with your name on it http://carriers.icall.com/open-source/
>so it appears you have had experience with them.
>
>
>
We have a lot of experience with iCall. I'm not familiar with any hard requirement to "answer" the inbound leg prior to bridging an outbound leg. What happens in your dialplan if you don't explicitly answer the inbound leg prior to calling the bridge app?
-MC
?

>________________________________
> From: Anthony Minessale <anthony.minessale at gmail.com>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>
>Sent: Tuesday, August 16, 2011 5:29 PM
>
>Subject: Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
>
>
>You should never answer a call before bridging it anyway, it breaks all of the accounting.
>It would make sense to find out why the provider is doing that and get it fixed.
>
>
>
>On Mon, Aug 15, 2011 at 5:17 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
>Anthony,
>>
>>My gripe was not about simply having a DIALSTATUS variable in Freeswitch which copies what is from "originate_disposition" what I wanted is to be able to get the status of the B-Leg because right now when early media is played (which i wanted)? "originate_disposition" shows "ANSWER" which I think is caused by me explitly called the "answer" app in my dialplan before the bridge app, this is because my DID provider requires an answer/sip 200 or else it will keep re-sending the sip invite, therefore causing freeswitch to keep creating new channels. All I want is to be able to get the proper sip/hangup/dial statuses of the B-leg.
>>
>>
>>
>>
>>________________________________
>> From: Anthony Minessale <anthony.minessale at gmail.com>
>>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>Sent: Wednesday, August 10, 2011 8:52 AM
>>Subject: Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>>
>>
>>
>>=D?
>>
>>
>>ok, sure. ?If that's your only complaint.... see commit?9d98d49f0556fb79656c8403f285ae0a615439d3
>>
>>
>>
>>Some caveats
>>
>>
>>1) There is actually less?specific, more generalized data in this DIALSTATUS variable than what we already report, when you're ready to move on see the originate_disposition variable: ?It's kind of like going from reporting the precise geo-location of a cafe in Paris to generalizing it to "EUROPE"?
>>
>>
>>We follow the Q.850 standard for call cause codes and follow the SIP RFC to map sip response codes to/from the Q.850?equivalent. ?Also each module has its own version "sip_hangup_disposition" for sip so you have both the real sip response code AND the official Q.850 equiv variables set on each call.
>>
>>
>>
>>
>>2) We don't have a torture feature so we never return that code.
>>
>>
>>
>>
>>3) Since our originate can return before a call is answered I added "EARLY" which means the originate succeeded but its still not answered.
>>
>>
>>4) For any others that do not map directly to FreeSWITCH, I just installed a copy of originate_disposition for good measure.
>>
>>P.S?
>>
>>
>>This email took longer to compose than the patch did while sitting in the middle of a crowded room so you probably could have simply parsed the originate originate_disposition yourself but if it helps people get over being stuck in a?paradigm, it's worth it for me to write........
>>?
>>
>>
>>On Tue, Aug 9, 2011 at 7:54 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>
>>I find that Asterisk's AGI is much easier to use, they allow you to retrieve the dial status much easier than freeswitch's api's. Come on freeswitch, if you want to be better than asterisk, you should make it easier to get the dialstatus, etc. At this point asterisk is still defacto.
>>>
>>>
>>>
>>>
>>>________________________________
>>> From: Nestor A Diaz <nestor at tiendalinux.com>
>>>To: freeswitch-users at lists.freeswitch.org
>>>Sent: Tuesday, August 9, 2011 9:48 AM
>>>Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>>>
>>>
>>>
>>>Hi Guys.
>>>
>>>I am starting to use FreeSWITCH, i am an asterisk user since the
1.0.7
release appears as a package on the debian distribution, at the
beginning i was amazed by the fact i can build a PBX for my own
business
and i did, later i began to install this system for my customers and
sooner i meet the problems, however being the software open source i
always find a way to fix things using patchs from others, sometimes i
felt how my life was at risk when the system stops working and that
usually happens when i have to use queues and dealing with digium
hardware.
>>>
>>>Fixing those problems either by applying patches or by changing
the
hardware where the digium cards were supposed to be installed helps me,
but that was to much stress for me and seeking for a balance that will
let me invest more time on services, configuration and hoping to have
better hardware
options brings me to freeswitch.
>>>
>>>I agree with freeswitch philosophy that instead of having
thousands of
modules
that don't work fine i prefer just a few that works the way it should
be, a rock solid system for a corporate pbx and a call center is what i
want.
>>>
>>>So here i am trying to begin the conversion, and i hope the
information
we can transcript in this list will help others that want to try
another
alternative to asterisk.
>>>
>>>First of all i think the saner for a migration is to have the two
systems
running either on the same machine or different and use the stable
features of each one.
>>>
>>>So could you please freeswitch users help me with this rosetta
stone migration guide in order to post it to voip-info.org or
freeswitch wiki (i list only the ones i currently use ):
>>>
>>>
>>>
>>>Technology Asterisk Freeswitch 
>>>PSTN Connectivity (Digium /
Sangoma) dahdi freetdm 
>>>IAX2 mod_iax ?? none stable yet.
>>>Use Asterisk to forward traffic via SIP.
>>>Enable Hardware HPET for IAX2 trunk if card not available for Asterisk  
>>>Bluetooth Channel chan_mobile ??
>>>Use asterisk via SIP
>>> 
>>>Skype Skypeforasterisk (no longer for sale) mod_skypeopen 
>>>CDR Stadistics Arternic cdr-stats  ?? 
>>>Queue Statistics  Asteriskguru queue-stats ?? 
>>>Web Management Freepbx ?? 
>>>IVR AGI / AMI Event Socket 
>>>Codec G.729 Transcodind Cards
>>>G.729 licenses
>>>Free G.729 (Intel IPP) Transcodind Cards
>>>G.729 licenses
>>>fsg729 Intel IPP(any experience with it ? ) 
>>>Fax Handling Iaxmodem with Hylafax ??
>>>Iaxmodem via asterisk to FS via SIP ?
>>> 
>>>SIP chan_sip sofia 
>>>ACD app_queue mod_callcenter 
>>>
>>>Thank you all
>>>
>>>
>>>-- 
>>>Nestor A. Diaz
>>>Ingeniero de Sistemas
>>>Tel. +57 1-485-3020 x 211
>>>Cel. +57 316-227-3593
>>>Tel. SIP: sip:211 at tiendalinux.com
>>>Email/MSN: nestor at tiendalinux.com
>>>http://www.tiendalinux.com/
>>>Bogota, Colombia 
>>> 
>>>
>>>
>>>_______________________________________________
>>>Join us at ClueCon 2011, Aug 9-11, Chicago
>>>http://www.cluecon.com 877-7-4ACLUE
>>>
>>>FreeSWITCH-users mailing list
>>>FreeSWITCH-users at lists.freeswitch.org
>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>http://www.freeswitch.org
>>>
>>>
>>>
>>>_______________________________________________
>>>Join us at ClueCon 2011, Aug 9-11, Chicago
>>>http://www.cluecon.com 877-7-4ACLUE
>>>
>>>FreeSWITCH-users mailing list
>>>FreeSWITCH-users at lists.freeswitch.org
>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>http://www.freeswitch.org
>>>
>>>
>>
>>
>>-- 
>>Anthony Minessale II
>>
>>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
>>googletalk:conf+888 at conference.freeswitch.org
>>pstn:+19193869900
>>
>>_______________________________________________
>>Join us at ClueCon 2011, Aug 9-11, Chicago
>>http://www.cluecon.com 877-7-4ACLUE
>>
>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>>
>>
>>
>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>>
>
>
>
>-- 
>Anthony Minessale II
>
>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
>googletalk:conf+888 at conference.freeswitch.org
>pstn:+19193869900
>
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>
>
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>


FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/9256140b/attachment-0001.html 

From anthony.minessale at gmail.com  Wed Aug 17 22:00:34 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 17 Aug 2011 13:00:34 -0500
Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
In-Reply-To: <1313602322.16798.YahooMailNeo@web161015.mail.bf1.yahoo.com>
References: <4E4164C0.8030507@tiendalinux.com>
	<1312937649.7702.YahooMailNeo@web161011.mail.bf1.yahoo.com>
	<CAKbxfG_802p+G1a9HtLEh_i-K1C06fpZgyR4_NP=jZJR68YK8g@mail.gmail.com>
	<1313446639.81086.YahooMailNeo@web161008.mail.bf1.yahoo.com>
	<CAKbxfG8jVqBMWODa72nS692buKjeX_LrAT05zz1HeQqg5MM3Yg@mail.gmail.com>
	<1313558080.89178.YahooMailNeo@web161010.mail.bf1.yahoo.com>
	<CAKzWOxX5rb9BYUAtaW4FjPyq=M1Z9G9QkkdVg+WJS9VMxvDHoQ@mail.gmail.com>
	<1313602322.16798.YahooMailNeo@web161015.mail.bf1.yahoo.com>
Message-ID: <CAKbxfG-S_QPcOY4Jha9mRg85MW2OEh=U+gaAAFc4L9BQ-YCuaA@mail.gmail.com>

This is another problem related to the callflow of the provider that can be
fixed.

In an ideal world, using the defaults, when the early media comes up on the
b leg it will pass to the a leg which also will start sending early media
and it will happily pass through.

My hunch is they have calls to you set on some kine of LCR hunt that is
misconfigured and it's trying to get the answer to stop hunting which is not
right.


On Wed, Aug 17, 2011 at 12:32 PM, Sam <lakersman2006 at yahoo.com> wrote:

> I have also found a side effect when I do not explicitly call answer on the
> inbound leg for b-leg calls that do not return "answer" when using another
> DID provider (VOIPInnovations). The side effect is that the a-leg can hear
> the telco network messages from the carrier like "I'm sorry the number you
> dialed is not a working number ..." or "The user is not accepting calls at
> the moment."
>
> If I do explicitly call answer, then I cannot hear those telco messages,
> which would seem to be better fitting for my case.
>
> ------------------------------
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Wednesday, August 17, 2011 9:16 AM
>
> *Subject:* Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
>
>
> On Tue, Aug 16, 2011 at 10:14 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> The DID provider I am using is from iCall, and I was searching through
> their website and noticed that they mentioned a quote with your name on it
> http://carriers.icall.com/open-source/
> so it appears you have had experience with them.
>
> We have a lot of experience with iCall. I'm not familiar with any hard
> requirement to "answer" the inbound leg prior to bridging an outbound leg.
> What happens in your dialplan if you don't explicitly answer the inbound leg
> prior to calling the bridge app?
> -MC
>
>
> ------------------------------
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Tuesday, August 16, 2011 5:29 PM
> *Subject:* Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
> You should never answer a call before bridging it anyway, it breaks all of
> the accounting.
> It would make sense to find out why the provider is doing that and get it
> fixed.
>
>
> On Mon, Aug 15, 2011 at 5:17 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> Anthony,
>
> My gripe was not about simply having a DIALSTATUS variable in Freeswitch
> which copies what is from "originate_disposition" what I wanted is to be
> able to get the status of the B-Leg because right now when early media is
> played (which i wanted)  "originate_disposition" shows "ANSWER" which I
> think is caused by me explitly called the "answer" app in my dialplan before
> the bridge app, this is because my DID provider requires an answer/sip 200
> or else it will keep re-sending the sip invite, therefore causing freeswitch
> to keep creating new channels. All I want is to be able to get the proper
> sip/hangup/dial statuses of the B-leg.
>
> ------------------------------
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Wednesday, August 10, 2011 8:52 AM
> *Subject:* Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
> =D
>
> ok, sure.  If that's your only complaint.... see
> commit 9d98d49f0556fb79656c8403f285ae0a615439d3
>
>
>
> Some caveats
>
> 1) There is actually less specific, more generalized data in this
> DIALSTATUS variable than what we already report, when you're ready to move
> on see the originate_disposition variable:  It's kind of like going from
> reporting the precise geo-location of a cafe in Paris to generalizing it to
> "EUROPE"
>
> We follow the Q.850 standard for call cause codes and follow the SIP RFC to
> map sip response codes to/from the Q.850 equivalent.  Also each module has
> its own version "sip_hangup_disposition" for sip so you have both the real
> sip response code AND the official Q.850 equiv variables set on each call.
>
>
> 2) We don't have a torture feature so we never return that code.
>
>
> 3) Since our originate can return before a call is answered I added "EARLY"
> which means the originate succeeded but its still not answered.
>
> 4) For any others that do not map directly to FreeSWITCH, I just installed
> a copy of originate_disposition for good measure.
>
> P.S
>
> This email took longer to compose than the patch did while sitting in the
> middle of a crowded room so you probably could have simply parsed the
> originate originate_disposition yourself but if it helps people get over
> being stuck in a paradigm, it's worth it for me to write........
>
>
> On Tue, Aug 9, 2011 at 7:54 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> I find that Asterisk's AGI is much easier to use, they allow you to
> retrieve the dial status much easier than freeswitch's api's. Come on
> freeswitch, if you want to be better than asterisk, you should make it
> easier to get the dialstatus, etc. At this point asterisk is still defacto.
>
> ------------------------------
> *From:* Nestor A Diaz <nestor at tiendalinux.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Tuesday, August 9, 2011 9:48 AM
> *Subject:* [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
> Hi Guys.
>
> I am starting to use FreeSWITCH, i am an asterisk user since the 1.0.7
> release appears as a package on the debian distribution, at the beginning i
> was amazed by the fact i can build a PBX for my own business and i did,
> later i began to install this system for my customers and sooner i meet the
> problems, however being the software open source i always find a way to fix
> things using patchs from others, sometimes i felt how my life was at risk
> when the system stops working and that usually happens when i have to use
> queues and dealing with digium hardware.
>
> Fixing those problems either by applying patches or by changing the
> hardware where the digium cards were supposed to be installed helps me, but
> that was to much stress for me and seeking for a balance that will let me
> invest more time on services, configuration and hoping to have better
> hardware options brings me to freeswitch.
>
> I agree with freeswitch philosophy that instead of having thousands of
> modules that don't work fine i prefer just a few that works the way it
> should be, a rock solid system for a corporate pbx and a call center is what
> i want.
>
> So here i am trying to begin the conversion, and i hope the information we
> can transcript in this list will help others that want to try another
> alternative to asterisk.
>
> First of all i think the saner for a migration is to have the two systems
> running either on the same machine or different and use the stable features
> of each one.
>
> So could you please freeswitch users help me with this rosetta stone
> migration guide in order to post it to voip-info.org or freeswitch wiki (i
> list only the ones i currently use ):
>
>
>   *Technology* *Asterisk* *Freeswitch*  PSTN Connectivity (Digium /
> Sangoma) dahdi freetdm  IAX2 mod_iax ?? none stable yet.
> Use Asterisk to forward traffic via SIP.
> Enable Hardware HPET for IAX2 trunk if card not available for Asterisk  Bluetooth
> Channel chan_mobile ??
> Use asterisk via SIP
>   Skype Skypeforasterisk (no longer for sale) mod_skypeopen  CDR
> Stadistics Arternic cdr-stats ??  Queue Statistics  Asteriskguru
> queue-stats ??  Web Management Freepbx ??  IVR AGI / AMI Event Socket  Codec
> G.729 Transcodind Cards
> G.729 licenses
> Free G.729 (Intel IPP) Transcodind Cards
> G.729 licenses
> fsg729 Intel IPP(any experience with it ? )  Fax Handling Iaxmodem with
> Hylafax ??
> Iaxmodem via asterisk to FS via SIP ?
>   SIP chan_sip sofia  ACD app_queue mod_callcenter
>
> Thank you all
>
>
> --
> Nestor A. Diaz
> Ingeniero de Sistemas
> Tel. +57 1-485-3020 x 211
> Cel. +57 316-227-3593
> Tel. SIP: sip:211 at tiendalinux.com
> Email/MSN: nestor at tiendalinux.com
> http://www.tiendalinux.com/
> Bogota, Colombia
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/5eff5ac5/attachment-0001.html 

From msc at freeswitch.org  Wed Aug 17 22:33:09 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 17 Aug 2011 11:33:09 -0700
Subject: [Freeswitch-users] getting disconnect cause for a leg after
 bridge in Lua
In-Reply-To: <20110817145132.12150@gmx.com>
References: <20110817145132.12150@gmx.com>
Message-ID: <CAKzWOxXEzD9MV74-JeYrwVXp97t0-GMEYJMQUiddBde6SE3aTw@mail.gmail.com>

Well, it's possible if you're using an event socket program, however if you
are using a dialplan script you simply cannot bend the laws of physics to
magically know what's going on with the b leg when your program is "inside"
the a leg.

I recommend that you use a socket-based program if you want this level of
control. A socket-based program gives you the perspective that you need: you
can "oversee" multiple call legs. Just a thought...

-MC

On Wed, Aug 17, 2011 at 7:51 AM, Mi Ke <mi.ke at null.net> wrote:

> Thanks for replying, Michael.
>
> Meanwhile if session creation fails, such session can also be considered as
> disconnected and gone, however its disconnect cause can be read from
> session:hangupCause(). To my opinion, it would be very useful if disconnect
> cause could be always readable in Lua after session completion since such
> mechanism already exists for failed sessions.
>
> Thanks / Mike
>
>
>
>
>
>
>  ----- Original Message -----
>
> From: Michael Collins
>
> Sent: 08/16/11 03:03 AM
>
> To: FreeSWITCH Users Help
>
> Subject: Re: [Freeswitch-users] getting disconnect cause for a leg after
> bridge in Lua
>
>  No, you can't do this because the session you are checking is "gone" as
> soon as the call leg is disconnected. You are better off using a hangup hook
> or an event socket application if you need to get that value in realtime.
> Dialplan scripts are good for connecting endpoints and doing simple logic
> but they are absolutely not what you want for doing any kind of billing or
> reporting.
>
>  -MC
>
>  On Sat, Aug 13, 2011 at 2:46 AM, Mi Ke <mi.ke at null.net> wrote:
>
>> Hi All,
>>
>> Is there any way to get a real disconnection cause for leg B in the
>> following script ?
>>
>>
>> if (session_a:ready() and session_b:ready()) then
>>
>>    freeswitch.bridge(session_a,session_b)
>>
>>    -- session_b gets disconnect here ...
>>
>>     local session_b_hangup_cause = session_b:hangupCause()
>>
>>
>>
>> session_b_hangup_cause is always "SUCCESS" after debridging while log and
>> CDR shows correct value - can get it to my script ?
>>
>> Thanks / Mike
>>
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/a7afde20/attachment.html 

From adrottenberg at gmail.com  Wed Aug 17 22:33:25 2011
From: adrottenberg at gmail.com (Duvid Rottenberg)
Date: Wed, 17 Aug 2011 14:33:25 -0400
Subject: [Freeswitch-users] Paging with Polycom Phones
In-Reply-To: <CAKzWOxUMbUYM_+eiQG98OpAYTHdur1GVADHzY1OmfV-E+NBHDw@mail.gmail.com>
References: <CAC4JBEAF4OMFeD9KV5LSAVZ7DxTG5ceK0Vj3cDEam6kU8hTD3Q@mail.gmail.com>
	<CAKzWOxUMbUYM_+eiQG98OpAYTHdur1GVADHzY1OmfV-E+NBHDw@mail.gmail.com>
Message-ID: <CAC4JBECX77JNE5htRm2nD8UOtfhVLqK2tnp3N2tnN+ep6DOAqA@mail.gmail.com>

The mad boss example looks like it uses the same commands as the
conferencing and intercom sample. I copied the  few settings that appear to
be different, but I'm still having the issue that the polycom phone answers
and hangs up right away. Did you have to make any changes to your polycom
config file to get this to work?

Thanks,
Duvid

On Wed, Aug 17, 2011 at 12:51 PM, Michael Collins <msc at freeswitch.org>wrote:

> Try using the "mad boss" example found in the default dialplan. I've tested
> that with Polycom phones. It's a nice workaround for all these different
> phone vendors who do things so differently.
>
> -MC
>
>   On Wed, Aug 17, 2011 at 8:55 AM, Duvid Rottenberg <
> adrottenberg at gmail.com> wrote:
>
>>   Has anyone successfully implemented paging (auto-answer) with a polycom
>> phone?
>> I am using the Conferencing and Intercom sample which sets the
>> sip_auto_answer variable to true, however on my polycom phone the result is
>> that the phone rings once and hangs up right away (the phone is sending a
>> BYE message). I tried adding an Alert-Info header, however it seems that the
>> Polycom format (Alert-Info: Ring Answer) isn't compliant with the RFC and I
>> couldn't get freeswitch to send the header in this format.
>>
>> Has anyone else been able to either get polycom phones to work with
>> sip_auto_answer or to get freeswitch to send an Alert-Info header in the
>> polycom format?
>>
>> Thanks,
>> Duvid Rottenberg
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/6cbeffc8/attachment.html 

From lakersman2006 at yahoo.com  Wed Aug 17 22:53:04 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Wed, 17 Aug 2011 11:53:04 -0700 (PDT)
Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
In-Reply-To: <CAKbxfG-S_QPcOY4Jha9mRg85MW2OEh=U+gaAAFc4L9BQ-YCuaA@mail.gmail.com>
References: <4E4164C0.8030507@tiendalinux.com>
	<1312937649.7702.YahooMailNeo@web161011.mail.bf1.yahoo.com>
	<CAKbxfG_802p+G1a9HtLEh_i-K1C06fpZgyR4_NP=jZJR68YK8g@mail.gmail.com>
	<1313446639.81086.YahooMailNeo@web161008.mail.bf1.yahoo.com>
	<CAKbxfG8jVqBMWODa72nS692buKjeX_LrAT05zz1HeQqg5MM3Yg@mail.gmail.com>
	<1313558080.89178.YahooMailNeo@web161010.mail.bf1.yahoo.com>
	<CAKzWOxX5rb9BYUAtaW4FjPyq=M1Z9G9QkkdVg+WJS9VMxvDHoQ@mail.gmail.com>
	<1313602322.16798.YahooMailNeo@web161015.mail.bf1.yahoo.com>
	<CAKbxfG-S_QPcOY4Jha9mRg85MW2OEh=U+gaAAFc4L9BQ-YCuaA@mail.gmail.com>
Message-ID: <1313607184.29837.YahooMailNeo@web161007.mail.bf1.yahoo.com>

I don't see much difference in terms of "originate_disposition" when calling answer explicitly opposed to not calling it, so since it appears there is more issues not calling it I? guess for now I should just call it.



________________________________
From: Anthony Minessale <anthony.minessale at gmail.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, August 17, 2011 11:00 AM
Subject: Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide


This is another problem related to the callflow of the provider that can be fixed.

In an ideal world, using the defaults, when the early media comes up on the b leg it will pass to the a leg which also will start sending early media and it will happily pass through.

My hunch is they have calls to you set on some kine of LCR hunt that is misconfigured and it's trying to get the answer to stop hunting which is not right.



On Wed, Aug 17, 2011 at 12:32 PM, Sam <lakersman2006 at yahoo.com> wrote:

I have also found a side effect when I do not explicitly call answer on the inbound leg for b-leg calls that do not return "answer" when using another DID provider (VOIPInnovations). The side effect is that the a-leg can hear the telco network messages from the carrier like "I'm sorry the number you dialed is not a working number ..." or "The user is not accepting calls at the moment." 
>
>
>
>If I do explicitly call answer, then I cannot hear those telco messages, which would seem to be better fitting for my case.
>
>
>
>
>________________________________
>From: Michael Collins <msc at freeswitch.org>
>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>Sent: Wednesday, August 17, 2011 9:16 AM
>
>Subject: Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
>
>
>
>
>
>On Tue, Aug 16, 2011 at 10:14 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
>The DID provider I am using is from iCall, and I was searching through their website and noticed that they mentioned a quote with your name on it http://carriers.icall.com/open-source/
>>so it appears you have had experience with them.
>>
>>
>>
>We have a lot of experience with iCall. I'm not familiar with any hard requirement to "answer" the inbound leg prior to bridging an outbound leg. What happens in your dialplan if you don't explicitly answer the inbound leg prior to calling the bridge app?
>-MC
>?
>
>>________________________________
>> From: Anthony Minessale <anthony.minessale at gmail.com>
>>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>
>>Sent: Tuesday, August 16, 2011 5:29 PM
>>
>>Subject: Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>>
>>
>>
>>You should never answer a call before bridging it anyway, it breaks all of the accounting.
>>It would make sense to find out why the provider is doing that and get it fixed.
>>
>>
>>
>>On Mon, Aug 15, 2011 at 5:17 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>
>>Anthony,
>>>
>>>My gripe was not about simply having a DIALSTATUS variable in Freeswitch which copies what is from "originate_disposition" what I wanted is to be able to get the status of the B-Leg because right now when early media is played (which i wanted)? "originate_disposition" shows "ANSWER" which I think is caused by me explitly called the "answer" app in my dialplan before the bridge app, this is because my DID provider requires an answer/sip 200 or else it will keep re-sending the sip invite, therefore causing freeswitch to keep creating new channels. All I want is to be able to get the proper sip/hangup/dial statuses of the B-leg.
>>>
>>>
>>>
>>>
>>>________________________________
>>> From: Anthony Minessale <anthony.minessale at gmail.com>
>>>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>Sent: Wednesday, August 10, 2011 8:52 AM
>>>Subject: Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>>>
>>>
>>>
>>>=D?
>>>
>>>
>>>ok, sure. ?If that's your only complaint.... see commit?9d98d49f0556fb79656c8403f285ae0a615439d3
>>>
>>>
>>>
>>>Some caveats
>>>
>>>
>>>1) There is actually less?specific, more generalized data in this DIALSTATUS variable than what we already report, when you're ready to move on see the originate_disposition variable: ?It's kind of like going from reporting the precise geo-location of a cafe in Paris to generalizing it to "EUROPE"?
>>>
>>>
>>>We follow the Q.850 standard for call cause codes and follow the SIP RFC to map sip response codes to/from the Q.850?equivalent. ?Also each module has its own version "sip_hangup_disposition" for sip so you have both the real sip response code AND the official Q.850 equiv variables set on each call.
>>>
>>>
>>>
>>>
>>>2) We don't have a torture feature so we never return that code.
>>>
>>>
>>>
>>>
>>>3) Since our originate can return before a call is answered I added "EARLY" which means the originate succeeded but its still not answered.
>>>
>>>
>>>4) For any others that do not map directly to FreeSWITCH, I just installed a copy of originate_disposition for good measure.
>>>
>>>P.S?
>>>
>>>
>>>This email took longer to compose than the patch did while sitting in the middle of a crowded room so you probably could have simply parsed the originate originate_disposition yourself but if it helps people get over being stuck in a?paradigm, it's worth it for me to write........
>>>?
>>>
>>>
>>>On Tue, Aug 9, 2011 at 7:54 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>>
>>>I find that Asterisk's AGI is much easier to use, they allow you to retrieve the dial status much easier than freeswitch's api's. Come on freeswitch, if you want to be better than asterisk, you should make it easier to get the dialstatus, etc. At this point asterisk is still defacto.
>>>>
>>>>
>>>>
>>>>
>>>>________________________________
>>>> From: Nestor A Diaz <nestor at tiendalinux.com>
>>>>To: freeswitch-users at lists.freeswitch.org
>>>>Sent: Tuesday, August 9, 2011 9:48 AM
>>>>Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>>>>
>>>>
>>>>
>>>>Hi Guys.
>>>>
>>>>I am starting to use FreeSWITCH, i am an asterisk user since the
1.0.7
release appears as a package on the debian distribution, at the
beginning i was amazed by the fact i can build a PBX for my own
business
and i did, later i began to install this system for my customers and
sooner i meet the problems, however being the software open source i
always find a way to fix things using patchs from others, sometimes i
felt how my life was at risk when the system stops working and that
usually happens when i have to use queues and dealing with digium
hardware.
>>>>
>>>>Fixing those problems either by applying patches or by changing
the
hardware where the digium cards were supposed to be installed helps me,
but that was to much stress for me and seeking for a balance that will
let me invest more time on services, configuration and hoping to have
better hardware
options brings me to freeswitch.
>>>>
>>>>I agree with freeswitch philosophy that instead of having
thousands of
modules
that don't work fine i prefer just a few that works the way it should
be, a rock solid system for a corporate pbx and a call center is what i
want.
>>>>
>>>>So here i am trying to begin the conversion, and i hope the
information
we can transcript in this list will help others that want to try
another
alternative to asterisk.
>>>>
>>>>First of all i think the saner for a migration is to have the two
systems
running either on the same machine or different and use the stable
features of each one.
>>>>
>>>>So could you please freeswitch users help me with this rosetta
stone migration guide in order to post it to voip-info.org or
freeswitch wiki (i list only the ones i currently use ):
>>>>
>>>>
>>>>
>>>>Technology Asterisk Freeswitch 
>>>>PSTN Connectivity (Digium /
Sangoma) dahdi freetdm 
>>>>IAX2 mod_iax ?? none stable yet.
>>>>Use Asterisk to forward traffic via SIP.
>>>>Enable Hardware HPET for IAX2 trunk if card not available for Asterisk  
>>>>Bluetooth Channel chan_mobile ??
>>>>Use asterisk via SIP
>>>> 
>>>>Skype Skypeforasterisk (no longer for sale) mod_skypeopen 
>>>>CDR Stadistics Arternic cdr-stats  ?? 
>>>>Queue Statistics  Asteriskguru queue-stats ?? 
>>>>Web Management Freepbx ?? 
>>>>IVR AGI / AMI Event Socket 
>>>>Codec G.729 Transcodind Cards
>>>>G.729 licenses
>>>>Free G.729 (Intel IPP) Transcodind Cards
>>>>G.729 licenses
>>>>fsg729 Intel IPP(any experience with it ? ) 
>>>>Fax Handling Iaxmodem with Hylafax ??
>>>>Iaxmodem via asterisk to FS via SIP ?
>>>> 
>>>>SIP chan_sip sofia 
>>>>ACD app_queue mod_callcenter 
>>>>
>>>>Thank you all
>>>>
>>>>
>>>>-- 
>>>>Nestor A. Diaz
>>>>Ingeniero de Sistemas
>>>>Tel. +57 1-485-3020 x 211
>>>>Cel. +57 316-227-3593
>>>>Tel. SIP: sip:211 at tiendalinux.com
>>>>Email/MSN: nestor at tiendalinux.com
>>>>http://www.tiendalinux.com/
>>>>Bogota, Colombia 
>>>> 
>>>>
>>>>
>>>>_______________________________________________
>>>>Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>>FreeSWITCH-users mailing list
>>>>FreeSWITCH-users at lists.freeswitch.org
>>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>_______________________________________________
>>>>Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>>FreeSWITCH-users mailing list
>>>>FreeSWITCH-users at lists.freeswitch.org
>>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>>-- 
>>>Anthony Minessale II
>>>
>>>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
>>>googletalk:conf+888 at conference.freeswitch.org
>>>pstn:+19193869900
>>>
>>>_______________________________________________
>>>Join us at ClueCon 2011, Aug 9-11, Chicago
>>>http://www.cluecon.com 877-7-4ACLUE
>>>
>>>FreeSWITCH-users mailing list
>>>FreeSWITCH-users at lists.freeswitch.org
>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>>FreeSWITCH-users mailing list
>>>FreeSWITCH-users at lists.freeswitch.org
>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>http://www.freeswitch.org
>>>
>>>
>>
>>
>>
>>-- 
>>Anthony Minessale II
>>
>>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
>>googletalk:conf+888 at conference.freeswitch.org
>>pstn:+19193869900
>>
>>
>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>>
>>
>>
>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>>
>
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>
>
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/7e813b2a/attachment-0001.html 

From djbinter at gmail.com  Thu Aug 18 00:05:39 2011
From: djbinter at gmail.com (DJB International)
Date: Wed, 17 Aug 2011 13:05:39 -0700
Subject: [Freeswitch-users] Multiple Registration
Message-ID: <CAAB3jWuzV-m=N6cN+ftSme+Ls9vuteWYMjc8TgKPm17Z3Do-Sg@mail.gmail.com>

I have some question regarding the multiple registration.  When I registered
2 devices, even though both of them got registered and I can make calls from
both devices, but when I list, I can only see the last device that got
registered on the list from sofia status profile internal.

However, when I comment out the multiple-registrations value=true in sofia
profile, I can then see 2 devices listed.

Questions:

1)  Is that normal behavior that I need to make multiple-registrations
value=true before I can see 2 devices listed when I run sofia status profile
internal ?

2)  How come I can only see one device listed even though both devices got
registered in the first situation?

Thank you,
Dorn B.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/7a1da880/attachment.html 

From msc at freeswitch.org  Thu Aug 18 00:29:50 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 17 Aug 2011 13:29:50 -0700
Subject: [Freeswitch-users] Paging with Polycom Phones
In-Reply-To: <CAC4JBECX77JNE5htRm2nD8UOtfhVLqK2tnp3N2tnN+ep6DOAqA@mail.gmail.com>
References: <CAC4JBEAF4OMFeD9KV5LSAVZ7DxTG5ceK0Vj3cDEam6kU8hTD3Q@mail.gmail.com>
	<CAKzWOxUMbUYM_+eiQG98OpAYTHdur1GVADHzY1OmfV-E+NBHDw@mail.gmail.com>
	<CAC4JBECX77JNE5htRm2nD8UOtfhVLqK2tnp3N2tnN+ep6DOAqA@mail.gmail.com>
Message-ID: <CAKzWOxW27QZx=5MLKxtVZaHCr_y0zWpXUEx59_sfmDBu4C_BTA@mail.gmail.com>

I have plain jane out-of-box Polycoms and they all seem to work. I can't
tell you if there are any magic settings.

Does anyone else know if there are specific settings on the Polys that need
to be set in order for the mad-boss to work?

-MC

On Wed, Aug 17, 2011 at 11:33 AM, Duvid Rottenberg
<adrottenberg at gmail.com>wrote:

> The mad boss example looks like it uses the same commands as the
> conferencing and intercom sample. I copied the  few settings that appear to
> be different, but I'm still having the issue that the polycom phone answers
> and hangs up right away. Did you have to make any changes to your polycom
> config file to get this to work?
>
> Thanks,
> Duvid
>
> On Wed, Aug 17, 2011 at 12:51 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Try using the "mad boss" example found in the default dialplan. I've
>> tested that with Polycom phones. It's a nice workaround for all these
>> different phone vendors who do things so differently.
>>
>> -MC
>>
>>   On Wed, Aug 17, 2011 at 8:55 AM, Duvid Rottenberg <
>> adrottenberg at gmail.com> wrote:
>>
>>>   Has anyone successfully implemented paging (auto-answer) with a
>>> polycom phone?
>>> I am using the Conferencing and Intercom sample which sets the
>>> sip_auto_answer variable to true, however on my polycom phone the result is
>>> that the phone rings once and hangs up right away (the phone is sending a
>>> BYE message). I tried adding an Alert-Info header, however it seems that the
>>> Polycom format (Alert-Info: Ring Answer) isn't compliant with the RFC and I
>>> couldn't get freeswitch to send the header in this format.
>>>
>>> Has anyone else been able to either get polycom phones to work with
>>> sip_auto_answer or to get freeswitch to send an Alert-Info header in the
>>> polycom format?
>>>
>>> Thanks,
>>> Duvid Rottenberg
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/f9244ec9/attachment.html 

From msc at freeswitch.org  Thu Aug 18 00:33:23 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 17 Aug 2011 13:33:23 -0700
Subject: [Freeswitch-users] mod_spidermonkey loading error
In-Reply-To: <4E4B9352.4010006@hw.ac.uk>
References: <20110817091244.5582.qmail@community37.interfree.it>
	<4E4B9352.4010006@hw.ac.uk>
Message-ID: <CAKzWOxVyTnNMeHvxAWviHNM5MfPFW8XQoF_M0X=0ghzLv3Zhpw@mail.gmail.com>

This is an odd duck. Most likely you're on a 64 bit platform. There's an
issue with CentOS6 and evidently Fedora14. Try re-running configure like
this:

./configure --without-libcurl

See http://jira.freeswitch.org/browse/FS-3393 for a discussion of this bug.

-MC

On Wed, Aug 17, 2011 at 3:09 AM, xl127 <x.liu at hw.ac.uk> wrote:

> Hi,
>
> I can run the FreeSwitch on CentOS 5 without any problem.
> When I run it on Fedora 14 (tried latest git version and latest snapshot
> version), I got following error
>
> 011-08-16 18:19:52.928857 [CRIT] switch_loadable_module.c:929 Error
> Loading module /usr/local/freeswitch/mod/mod_spidermonkey.so
> **/usr/lib/libldap-2.4.so.2: undefined symbol: PR_GetDirectorySeparator**
>
> I'm stuck here for a while, googled around but didn't figure out a
> solution.
>
> Any suggestions?
>
> Thanks,
>
> Xing
>
>
>
> --
> Heriot-Watt University is a Scottish charity
> registered under charity number SC000278.
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/7b0b5555/attachment.html 

From msc at freeswitch.org  Thu Aug 18 00:37:55 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 17 Aug 2011 13:37:55 -0700
Subject: [Freeswitch-users] voicemail doesn't work
In-Reply-To: <33095823FD21DF429B481B5163264B7950FF130083@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B7950FF12FD15@VMBX102.ihostexchange.net>
	<CAHKqMb6=znNGaG0pDOj_AbaZM-P1=JOJ2hCH3PDZ0Pqwm5CUEg@mail.gmail.com>
	<33095823FD21DF429B481B5163264B7950FF130083@VMBX102.ihostexchange.net>
Message-ID: <CAKzWOxX9rrQ_PJBBhhT+vc4Es4yq-U=q1rB7Zm43kFQHxsOreA@mail.gmail.com>

Use pastebin.freeswitch.org and put the console debug output there. Capture
the traffic for both the working and non-working dialplans. Hopefully there
will be an error or warning that gives a clue as to what is happening. Hint:
use "FreeSWITCH Log" as the syntax highlighting and it will be much easier
to read.

-MC

On Wed, Aug 17, 2011 at 8:55 AM, Yungwei Chen <yungwei at resolvity.com> wrote:

> If I change my dialplan to the following, voicemail will work properly.
> What am I missing here? Thanks.****
>
>    <extension name="main_number_2" continue="true">
>      <condition field="destination_number" expression="^1112223333$">****
>
> <!--****
>
>        <action application="set" bluebox="settingEndBridge"
> data="hangup_after_bridge=true"/>
>        <action application="set" bluebox="settingFail"
> data="continue_on_fail=true"/>
>        <action application="set" bluebox="autoPlay"
> data="vm_auto_play=false"/>
>        <action application="set" bluebox="settingTimeout"
> data="call_timeout=30"/>
>        <action application="set" bluebox="settingRing"
> data="ringback=${us-ring}"/>
>        <action application="set" bluebox="settingRingXfer"
> data="transfer_ringback=${us-ring}"/>
>        <action application="export" bluebox="sipCalleeIdName"
> data="sip_callee_id_name=m1"/>
>        <action application="export" bluebox="sipCalleeIdNumber"
> data="sip_callee_id_number=1112223333"/>
> -->****
>
>        <action application="answer"/>
>        <action application="voicemail" data="default voicemail_2 91000"/>
>        <action application="hangup"/>
>      </condition>
>    </extension>****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Nandy
> Dagondon
> *Sent:* Monday, August 15, 2011 6:52 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] voicemail is not saved****
>
> ** **
>
> check the directory/file permissions
> -nandy****
>
> On Tue, Aug 16, 2011 at 3:33 AM, Yungwei Chen wrote:****
>
> Hi,
>
> I left several voicemails (Each is longer than 3 sec) to a user account,
> but none is available when I check the mailbox.
> Relevant settings are listed below. What am I missing here? Thanks.
>
> In conf/autoload_configs/modules.conf.xml, mod_voicemail is already loaded.
> freeswitch at internal> load mod_voicemail
> +OK Reloading XML
> -ERR [Module already loaded]
> freeswitch at internal> 2011-08-15 14:32:10.666978 [WARNING]
> switch_loadable_module.c:998 Module mod_voicemail Already Loaded!
>
> Here's the content of conf/autoload_configs/voicemail.conf.xml:
> <configuration name="voicemail.conf" description="Voicemail">
>  <settings>
>  </settings>
>  <profiles>
>    <profile name="default">
>      <param name="file-extension" value="wav"/>
>      <param name="terminator-key" value="#"/>
>      <param name="max-login-attempts" value="3"/>
>      <param name="digit-timeout" value="10000"/>
>      <param name="min-record-len" value="3"/>
>      <param name="max-record-len" value="300"/>
>      <param name="max-retries" value="3"/>
>      <param name="tone-spec" value="%(1000, 0, 640)"/>
>      <param name="callback-dialplan" value="XML"/>
>      <param name="callback-context" value="default"/>
>      <param name="play-new-messages-key" value="1"/>
>      <param name="play-saved-messages-key" value="2"/>
>      <param name="login-keys" value="0"/>
>      <param name="main-menu-key" value="0"/>
>      <param name="config-menu-key" value="5"/>
>      <param name="record-greeting-key" value="1"/>
>      <param name="choose-greeting-key" value="2"/>
>      <param name="change-pass-key" value="6"/>
>      <param name="record-name-key" value="3"/>
>      <param name="record-file-key" value="3"/>
>      <param name="listen-file-key" value="1"/>
>      <param name="save-file-key" value="2"/>
>      <param name="delete-file-key" value="7"/>
>      <param name="undelete-file-key" value="8"/>
>      <param name="email-key" value="4"/>
>      <param name="pause-key" value="0"/>
>      <param name="restart-key" value="1"/>
>      <param name="ff-key" value="6"/>
>      <param name="rew-key" value="4"/>
>      <param name="skip-greet-key" value="#"/>
>      <param name="previous-message-key" value="1"/>
>      <param name="next-message-key" value="3"/>
>      <param name="skip-info-key" value="*"/>
>      <param name="repeat-message-key" value="0"/>
>      <param name="record-silence-threshold" value="200"/>
>      <param name="record-silence-hits" value="2"/>
>      <param name="web-template-file" value="web-vm.tpl"/>
>      <!-- if you need to change the sample rate of the recorded files e.g.
> gmail voicemail player -->
>      <!--<param name="record-sample-rate" value="11025"/>-->
>      <!-- the next two both must be set for this to be enabled
>           the extension is in the format of <dest> [<dialplan>] [<context>]
>       -->
>      <param name="operator-extension" value="operator XML default"/>
>      <param name="operator-key" value="9"/>
>      <param name="vmain-extension" value="vmain XML default"/>
>      <param name="vmain-key" value="*"/>
>      <!-- playback created files as soon as they were recorded by default
> -->
>      <!--<param name="auto-playback-recordings" value="true"/>-->
>      <email>
>        <param name="template-file" value="voicemail.tpl"/>
>        <param name="notify-template-file" value="notify-voicemail.tpl"/>
>        <!-- this is the format voicemail_time will have -->
>        <param name="date-fmt" value="%A, %B %d %Y, %I %M %p"/>
>        <param name="email-from" value="${voicemail_account}@
> ${voicemail_domain}"/>
>      </email>
>      <!--<param name="storage-dir" value="/tmp"/>-->
>      <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
>      <!--<param name="record-comment" value="Your Comment"/>-->
>      <!--<param name="record-title" value="Your Title"/>-->
>      <!--<param name="record-copyright" value="Your Copyright"/>-->
>    </profile>
>  </profiles>
> </configuration>
>
> In conf/directory/default.xml, user 91000 is defined in domain voicemail_2.
>  <domain name="voicemail_2">
>    <groups>
>      <group name="default">
>        <users>
>          <user bluebox_id="2" id="91000">
>            <params>
>              <param name="sip-forbid-register" value="true"/>
>              <param name="vm-password" value="xxxx"/>
>              <param name="vm-message-ext" value="wav"/>
>              <param name="vm-mailto" value="xxx at xxx.com"/>
>              <param name="email-addr" value="xxx at xxx.com"/>
>              <param name="vm-delete-file" value="false"/>
>              <param name="vm-attach-file" value="true"/>
>              <param name="vm-email-all-messages" value="true"/>
>            </params>
>            <variables>
>              <variable name="timezone" value="America/Chicago"/>
>            </variables>
>          </user>
>        </users>
>      </group>
>    </groups>
>  </domain>
>
> In my dialplan, calls to 1112223333 will be sent to user 91000's voicemail
> box if they are not answered.
>    <extension name="main_number_2" continue="true">
>      <condition field="destination_number" expression="^1112223333$">
>        <action application="set" bluebox="settingEndBridge"
> data="hangup_after_bridge=true"/>
>        <action application="set" bluebox="settingFail"
> data="continue_on_fail=true"/>
>        <action application="set" bluebox="autoPlay"
> data="vm_auto_play=false"/>
>        <action application="set" bluebox="settingTimeout"
> data="call_timeout=30"/>
>        <action application="set" bluebox="settingRing"
> data="ringback=${us-ring}"/>
>        <action application="set" bluebox="settingRingXfer"
> data="transfer_ringback=${us-ring}"/>
>        <action application="export" bluebox="sipCalleeIdName"
> data="sip_callee_id_name=m1"/>
>        <action application="export" bluebox="sipCalleeIdNumber"
> data="sip_callee_id_number=1112223333"/>
>        <action application="bridge" data="user/91000@$${location_3}"/>
>        <action application="answer"/>
>        <action application="voicemail" data="default voicemail_2 91000"/>
>        <action application="hangup"/>
>      </condition>
>    </extension>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
> ** **
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/976fea75/attachment-0001.html 

From msc at freeswitch.org  Thu Aug 18 00:44:47 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 17 Aug 2011 13:44:47 -0700
Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
In-Reply-To: <1313607184.29837.YahooMailNeo@web161007.mail.bf1.yahoo.com>
References: <4E4164C0.8030507@tiendalinux.com>
	<1312937649.7702.YahooMailNeo@web161011.mail.bf1.yahoo.com>
	<CAKbxfG_802p+G1a9HtLEh_i-K1C06fpZgyR4_NP=jZJR68YK8g@mail.gmail.com>
	<1313446639.81086.YahooMailNeo@web161008.mail.bf1.yahoo.com>
	<CAKbxfG8jVqBMWODa72nS692buKjeX_LrAT05zz1HeQqg5MM3Yg@mail.gmail.com>
	<1313558080.89178.YahooMailNeo@web161010.mail.bf1.yahoo.com>
	<CAKzWOxX5rb9BYUAtaW4FjPyq=M1Z9G9QkkdVg+WJS9VMxvDHoQ@mail.gmail.com>
	<1313602322.16798.YahooMailNeo@web161015.mail.bf1.yahoo.com>
	<CAKbxfG-S_QPcOY4Jha9mRg85MW2OEh=U+gaAAFc4L9BQ-YCuaA@mail.gmail.com>
	<1313607184.29837.YahooMailNeo@web161007.mail.bf1.yahoo.com>
Message-ID: <CAKzWOxWhdBGAk13aD9KKKr=LvAXyq0vCo6PkgrLJqWizk7cD3Q@mail.gmail.com>

Sam,

Did you already pastebin a copy of your script and dialplan? I know we had
talked about it. In any case, I'm hoping to see what you're doing so that we
can offer you some alternative ideas.

-MC

On Wed, Aug 17, 2011 at 11:53 AM, Sam <lakersman2006 at yahoo.com> wrote:

> I don't see much difference in terms of "originate_disposition" when
> calling answer explicitly opposed to not calling it, so since it appears
> there is more issues not calling it I  guess for now I should just call it.
>
> ------------------------------
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Wednesday, August 17, 2011 11:00 AM
>
> *Subject:* Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
> This is another problem related to the callflow of the provider that can be
> fixed.
>
> In an ideal world, using the defaults, when the early media comes up on the
> b leg it will pass to the a leg which also will start sending early media
> and it will happily pass through.
>
> My hunch is they have calls to you set on some kine of LCR hunt that is
> misconfigured and it's trying to get the answer to stop hunting which is not
> right.
>
>
> On Wed, Aug 17, 2011 at 12:32 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> I have also found a side effect when I do not explicitly call answer on the
> inbound leg for b-leg calls that do not return "answer" when using another
> DID provider (VOIPInnovations). The side effect is that the a-leg can hear
> the telco network messages from the carrier like "I'm sorry the number you
> dialed is not a working number ..." or "The user is not accepting calls at
> the moment."
>
> If I do explicitly call answer, then I cannot hear those telco messages,
> which would seem to be better fitting for my case.
>
> ------------------------------
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Wednesday, August 17, 2011 9:16 AM
>
> *Subject:* Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
>
>
> On Tue, Aug 16, 2011 at 10:14 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> The DID provider I am using is from iCall, and I was searching through
> their website and noticed that they mentioned a quote with your name on it
> http://carriers.icall.com/open-source/
> so it appears you have had experience with them.
>
> We have a lot of experience with iCall. I'm not familiar with any hard
> requirement to "answer" the inbound leg prior to bridging an outbound leg.
> What happens in your dialplan if you don't explicitly answer the inbound leg
> prior to calling the bridge app?
> -MC
>
>
> ------------------------------
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Tuesday, August 16, 2011 5:29 PM
> *Subject:* Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
> You should never answer a call before bridging it anyway, it breaks all of
> the accounting.
> It would make sense to find out why the provider is doing that and get it
> fixed.
>
>
> On Mon, Aug 15, 2011 at 5:17 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> Anthony,
>
> My gripe was not about simply having a DIALSTATUS variable in Freeswitch
> which copies what is from "originate_disposition" what I wanted is to be
> able to get the status of the B-Leg because right now when early media is
> played (which i wanted)  "originate_disposition" shows "ANSWER" which I
> think is caused by me explitly called the "answer" app in my dialplan before
> the bridge app, this is because my DID provider requires an answer/sip 200
> or else it will keep re-sending the sip invite, therefore causing freeswitch
> to keep creating new channels. All I want is to be able to get the proper
> sip/hangup/dial statuses of the B-leg.
>
> ------------------------------
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Wednesday, August 10, 2011 8:52 AM
> *Subject:* Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
> =D
>
> ok, sure.  If that's your only complaint.... see
> commit 9d98d49f0556fb79656c8403f285ae0a615439d3
>
>
>
> Some caveats
>
> 1) There is actually less specific, more generalized data in this
> DIALSTATUS variable than what we already report, when you're ready to move
> on see the originate_disposition variable:  It's kind of like going from
> reporting the precise geo-location of a cafe in Paris to generalizing it to
> "EUROPE"
>
> We follow the Q.850 standard for call cause codes and follow the SIP RFC to
> map sip response codes to/from the Q.850 equivalent.  Also each module has
> its own version "sip_hangup_disposition" for sip so you have both the real
> sip response code AND the official Q.850 equiv variables set on each call.
>
>
> 2) We don't have a torture feature so we never return that code.
>
>
> 3) Since our originate can return before a call is answered I added "EARLY"
> which means the originate succeeded but its still not answered.
>
> 4) For any others that do not map directly to FreeSWITCH, I just installed
> a copy of originate_disposition for good measure.
>
> P.S
>
> This email took longer to compose than the patch did while sitting in the
> middle of a crowded room so you probably could have simply parsed the
> originate originate_disposition yourself but if it helps people get over
> being stuck in a paradigm, it's worth it for me to write........
>
>
> On Tue, Aug 9, 2011 at 7:54 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> I find that Asterisk's AGI is much easier to use, they allow you to
> retrieve the dial status much easier than freeswitch's api's. Come on
> freeswitch, if you want to be better than asterisk, you should make it
> easier to get the dialstatus, etc. At this point asterisk is still defacto.
>
> ------------------------------
> *From:* Nestor A Diaz <nestor at tiendalinux.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Tuesday, August 9, 2011 9:48 AM
> *Subject:* [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
> Hi Guys.
>
> I am starting to use FreeSWITCH, i am an asterisk user since the 1.0.7
> release appears as a package on the debian distribution, at the beginning i
> was amazed by the fact i can build a PBX for my own business and i did,
> later i began to install this system for my customers and sooner i meet the
> problems, however being the software open source i always find a way to fix
> things using patchs from others, sometimes i felt how my life was at risk
> when the system stops working and that usually happens when i have to use
> queues and dealing with digium hardware.
>
> Fixing those problems either by applying patches or by changing the
> hardware where the digium cards were supposed to be installed helps me, but
> that was to much stress for me and seeking for a balance that will let me
> invest more time on services, configuration and hoping to have better
> hardware options brings me to freeswitch.
>
> I agree with freeswitch philosophy that instead of having thousands of
> modules that don't work fine i prefer just a few that works the way it
> should be, a rock solid system for a corporate pbx and a call center is what
> i want.
>
> So here i am trying to begin the conversion, and i hope the information we
> can transcript in this list will help others that want to try another
> alternative to asterisk.
>
> First of all i think the saner for a migration is to have the two systems
> running either on the same machine or different and use the stable features
> of each one.
>
> So could you please freeswitch users help me with this rosetta stone
> migration guide in order to post it to voip-info.org or freeswitch wiki (i
> list only the ones i currently use ):
>
>
>   *Technology* *Asterisk* *Freeswitch*  PSTN Connectivity (Digium /
> Sangoma) dahdi freetdm  IAX2 mod_iax ?? none stable yet.
> Use Asterisk to forward traffic via SIP.
> Enable Hardware HPET for IAX2 trunk if card not available for Asterisk  Bluetooth
> Channel chan_mobile ??
> Use asterisk via SIP
>   Skype Skypeforasterisk (no longer for sale) mod_skypeopen  CDR
> Stadistics Arternic cdr-stats ??  Queue Statistics  Asteriskguru
> queue-stats ??  Web Management Freepbx ??  IVR AGI / AMI Event Socket  Codec
> G.729 Transcodind Cards
> G.729 licenses
> Free G.729 (Intel IPP) Transcodind Cards
> G.729 licenses
> fsg729 Intel IPP(any experience with it ? )  Fax Handling Iaxmodem with
> Hylafax ??
> Iaxmodem via asterisk to FS via SIP ?
>   SIP chan_sip sofia  ACD app_queue mod_callcenter
>
> Thank you all
>
>
> --
> Nestor A. Diaz
> Ingeniero de Sistemas
> Tel. +57 1-485-3020 x 211
> Cel. +57 316-227-3593
> Tel. SIP: sip:211 at tiendalinux.com
> Email/MSN: nestor at tiendalinux.com
> http://www.tiendalinux.com/
> Bogota, Colombia
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/24444183/attachment-0001.html 

From mi.ke at null.net  Thu Aug 18 00:58:12 2011
From: mi.ke at null.net (Mi Ke)
Date: Wed, 17 Aug 2011 20:58:12 +0000
Subject: [Freeswitch-users] getting disconnect cause for a leg after
 bridge in Lua
Message-ID: <20110817205812.12160@gmx.com>

I see ... Thanks, Michael



----- Original Message -----
From: Michael Collins
Sent: 08/17/11 09:33 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] getting disconnect cause for a leg after bridge in Lua

 Well, it's possible if you're using an event socket program, however if you are using a dialplan script you simply cannot bend the laws of physics to magically know what's going on with the b leg when your program is "inside" the a leg.

 I recommend that you use a socket-based program if you want this level of control. A socket-based program gives you the perspective that you need: you can "oversee" multiple call legs. Just a thought...

 -MC

 On Wed, Aug 17, 2011 at 7:51 AM, Mi Ke < mi.ke at null.net > wrote:
Thanks for replying, Michael.

 Meanwhile if session creation fails, such session can also be considered as disconnected and gone, however its disconnect cause can be read from session:hangupCause(). To my opinion, it would be very useful if disconnect cause could be always readable in Lua after session completion since such mechanism already exists for failed sessions.

 Thanks / Mike



----- Original Message -----
From: Michael Collins
Sent: 08/16/11 03:03 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] getting disconnect cause for a leg after bridge in Lua

No, you can't do this because the session you are checking is "gone" as soon as the call leg is disconnected. You are better off using a hangup hook or an event socket application if you need to get that value in realtime. Dialplan scripts are good for connecting endpoints and doing simple logic but they are absolutely not what you want for doing any kind of billing or reporting.

-MC

On Sat, Aug 13, 2011 at 2:46 AM, Mi Ke < mi.ke at null.net > wrote:
Hi All,

 Is there any way to get a real disconnection cause for leg B in the following script ?


 if (session_a:ready() and session_b:ready()) then

 freeswitch.bridge(session_a,session_b)

 -- session_b gets disconnect here ...

 local session_b_hangup_cause = session_b:hangupCause()



 session_b_hangup_cause is always "SUCCESS" after debridging while log and CDR shows correct value - can get it to my script ?

 Thanks / Mike

_______________________________________________
 Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com  877-7-4ACLUE

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


 FreeSWITCH-users mailing list
 FreeSWITCH-users at lists.freeswitch.org 
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/2a26eb55/attachment.html 

From avi at avimarcus.net  Thu Aug 18 01:15:02 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 18 Aug 2011 00:15:02 +0300
Subject: [Freeswitch-users] Multiple Registration
In-Reply-To: <CAAB3jWuzV-m=N6cN+ftSme+Ls9vuteWYMjc8TgKPm17Z3Do-Sg@mail.gmail.com>
References: <CAAB3jWuzV-m=N6cN+ftSme+Ls9vuteWYMjc8TgKPm17Z3Do-Sg@mail.gmail.com>
Message-ID: <CALNZuw=-DLxDMGV7iqxrOdRpZV_WVMQoPRPwXtb0EP152TMiGA@mail.gmail.com>

Both devices can make outgoing calls - they have the proper password.
However, that's kind of what multiple registrations=false means: when a user
registers, remove any older registrations from that user.
-Avi


On Wed, Aug 17, 2011 at 11:05 PM, DJB International <djbinter at gmail.com>wrote:

> I have some question regarding the multiple registration.  When I
> registered 2 devices, even though both of them got registered and I can make
> calls from both devices, but when I list, I can only see the last device
> that got registered on the list from sofia status profile internal.
>
> However, when I comment out the multiple-registrations value=true in sofia
> profile, I can then see 2 devices listed.
>
> Questions:
>
> 1)  Is that normal behavior that I need to make multiple-registrations
> value=true before I can see 2 devices listed when I run sofia status profile
> internal ?
>
> 2)  How come I can only see one device listed even though both devices got
> registered in the first situation?
>
> Thank you,
> Dorn B.
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110818/ade2bfee/attachment.html 

From lakersman2006 at yahoo.com  Thu Aug 18 01:16:16 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Wed, 17 Aug 2011 14:16:16 -0700 (PDT)
Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
In-Reply-To: <CAKzWOxWhdBGAk13aD9KKKr=LvAXyq0vCo6PkgrLJqWizk7cD3Q@mail.gmail.com>
References: <4E4164C0.8030507@tiendalinux.com>
	<1312937649.7702.YahooMailNeo@web161011.mail.bf1.yahoo.com>
	<CAKbxfG_802p+G1a9HtLEh_i-K1C06fpZgyR4_NP=jZJR68YK8g@mail.gmail.com>
	<1313446639.81086.YahooMailNeo@web161008.mail.bf1.yahoo.com>
	<CAKbxfG8jVqBMWODa72nS692buKjeX_LrAT05zz1HeQqg5MM3Yg@mail.gmail.com>
	<1313558080.89178.YahooMailNeo@web161010.mail.bf1.yahoo.com>
	<CAKzWOxX5rb9BYUAtaW4FjPyq=M1Z9G9QkkdVg+WJS9VMxvDHoQ@mail.gmail.com>
	<1313602322.16798.YahooMailNeo@web161015.mail.bf1.yahoo.com>
	<CAKbxfG-S_QPcOY4Jha9mRg85MW2OEh=U+gaAAFc4L9BQ-YCuaA@mail.gmail.com>
	<1313607184.29837.YahooMailNeo@web161007.mail.bf1.yahoo.com>
	<CAKzWOxWhdBGAk13aD9KKKr=LvAXyq0vCo6PkgrLJqWizk7cD3Q@mail.gmail.com>
Message-ID: <1313615776.59136.YahooMailNeo@web161014.mail.bf1.yahoo.com>

MC,

Here is the pastebin http://pastebin.freeswitch.org/17069 of my perl script. For the current script I just made it display the various channel variables so I can see what values freeswitch will provide me after the call has been 
bridged or not. But the script will be used for a call card app that will just bridge 1 caller to 1 callee, pretty straight forward. So I will need to be able to play back to the caller on certain call states like NO_ANSWER, BUSY, CONGESTION, ETC.



________________________________
From: Michael Collins <msc at freeswitch.org>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, August 17, 2011 1:44 PM
Subject: Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide


Sam,

Did you already pastebin a copy of your script and dialplan? I know we had talked about it. In any case, I'm hoping to see what you're doing so that we can offer you some alternative ideas.?

-MC


On Wed, Aug 17, 2011 at 11:53 AM, Sam <lakersman2006 at yahoo.com> wrote:

I don't see much difference in terms of "originate_disposition" when calling answer explicitly opposed to not calling it, so since it appears there is more issues not calling it I? guess for now I should just call it.
>
>
>
>
>________________________________
>From: Anthony Minessale <anthony.minessale at gmail.com>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>Sent: Wednesday, August 17, 2011 11:00 AM
>
>Subject: Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>
>
>
>This is another problem related to the callflow of the provider that can be fixed.
>
>
>In an ideal world, using the defaults, when the early media comes up on the b leg it will pass to the a leg which also will start sending early media and it will happily pass through.
>
>
>My hunch is they have calls to you set on some kine of LCR hunt that is misconfigured and it's trying to get the answer to stop hunting which is not right.
>
>
>
>On Wed, Aug 17, 2011 at 12:32 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
>I have also found a side effect when I do not explicitly call answer on the inbound leg for b-leg calls that do not return "answer" when using another DID provider (VOIPInnovations). The side effect is that the a-leg can hear the telco network messages from the carrier like "I'm sorry the number you dialed is not a working number ..." or "The user is not accepting calls at the moment." 
>>
>>
>>
>>If I do explicitly call answer, then I cannot hear those telco messages, which would seem to be better fitting for my case.
>>
>>
>>
>>
>>________________________________
>>From: Michael Collins <msc at freeswitch.org>
>>
>>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>Sent: Wednesday, August 17, 2011 9:16 AM
>>
>>Subject: Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>>
>>
>>
>>
>>
>>
>>On Tue, Aug 16, 2011 at 10:14 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>
>>The DID provider I am using is from iCall, and I was searching through their website and noticed that they mentioned a quote with your name on it http://carriers.icall.com/open-source/
>>>so it appears you have had experience with them.
>>>
>>>
>>>
>>We have a lot of experience with iCall. I'm not familiar with any hard requirement to "answer" the inbound leg prior to bridging an outbound leg. What happens in your dialplan if you don't explicitly answer the inbound leg prior to calling the bridge app?
>>-MC
>>?
>>
>>>________________________________
>>> From: Anthony Minessale <anthony.minessale at gmail.com>
>>>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>
>>>Sent: Tuesday, August 16, 2011 5:29 PM
>>>
>>>Subject: Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>>>
>>>
>>>
>>>You should never answer a call before bridging it anyway, it breaks all of the accounting.
>>>It would make sense to find out why the provider is doing that and get it fixed.
>>>
>>>
>>>
>>>On Mon, Aug 15, 2011 at 5:17 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>>
>>>Anthony,
>>>>
>>>>My gripe was not about simply having a DIALSTATUS variable in Freeswitch which copies what is from "originate_disposition" what I wanted is to be able to get the status of the B-Leg because right now when early media is played (which i wanted)? "originate_disposition" shows "ANSWER" which I think is caused by me explitly called the "answer" app in my dialplan before the bridge app, this is because my DID provider requires an answer/sip 200 or else it will keep re-sending the sip invite, therefore causing freeswitch to keep creating new channels. All I want is to be able to get the proper sip/hangup/dial statuses of the B-leg.
>>>>
>>>>
>>>>
>>>>
>>>>________________________________
>>>> From: Anthony Minessale <anthony.minessale at gmail.com>
>>>>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>>Sent: Wednesday, August 10, 2011 8:52 AM
>>>>Subject: Re: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>>>>
>>>>
>>>>
>>>>=D?
>>>>
>>>>
>>>>ok, sure. ?If that's your only complaint.... see commit?9d98d49f0556fb79656c8403f285ae0a615439d3
>>>>
>>>>
>>>>
>>>>Some caveats
>>>>
>>>>
>>>>1) There is actually less?specific, more generalized data in this DIALSTATUS variable than what we already report, when you're ready to move on see the originate_disposition variable: ?It's kind of like going from reporting the precise geo-location of a cafe in Paris to generalizing it to "EUROPE"?
>>>>
>>>>
>>>>We follow the Q.850 standard for call cause codes and follow the SIP RFC to map sip response codes to/from the Q.850?equivalent. ?Also each module has its own version "sip_hangup_disposition" for sip so you have both the real sip response code AND the official Q.850 equiv variables set on each call.
>>>>
>>>>
>>>>
>>>>
>>>>2) We don't have a torture feature so we never return that code.
>>>>
>>>>
>>>>
>>>>
>>>>3) Since our originate can return before a call is answered I added "EARLY" which means the originate succeeded but its still not answered.
>>>>
>>>>
>>>>4) For any others that do not map directly to FreeSWITCH, I just installed a copy of originate_disposition for good measure.
>>>>
>>>>P.S?
>>>>
>>>>
>>>>This email took longer to compose than the patch did while sitting in the middle of a crowded room so you probably could have simply parsed the originate originate_disposition yourself but if it helps people get over being stuck in a?paradigm, it's worth it for me to write........
>>>>?
>>>>
>>>>
>>>>On Tue, Aug 9, 2011 at 7:54 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>>>
>>>>I find that Asterisk's AGI is much easier to use, they allow you to retrieve the dial status much easier than freeswitch's api's. Come on freeswitch, if you want to be better than asterisk, you should make it easier to get the dialstatus, etc. At this point asterisk is still defacto.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>________________________________
>>>>> From: Nestor A Diaz <nestor at tiendalinux.com>
>>>>>To: freeswitch-users at lists.freeswitch.org
>>>>>Sent: Tuesday, August 9, 2011 9:48 AM
>>>>>Subject: [Freeswitch-users] Asterisk to FreeSWITCH migration guide
>>>>>
>>>>>
>>>>>
>>>>>Hi Guys.
>>>>>
>>>>>I am starting to use FreeSWITCH, i am an asterisk user since the
1.0.7
release appears as a package on the debian distribution, at the
beginning i was amazed by the fact i can build a PBX for my own
business
and i did, later i began to install this system for my customers and
sooner i meet the problems, however being the software open source i
always find a way to fix things using patchs from others, sometimes i
felt how my life was at risk when the system stops working and that
usually happens when i have to use queues and dealing with digium
hardware.
>>>>>
>>>>>Fixing those problems either by applying patches or by changing
the
hardware where the digium cards were supposed to be installed helps me,
but that was to much stress for me and seeking for a balance that will
let me invest more time on services, configuration and hoping to have
better hardware
options brings me to freeswitch.
>>>>>
>>>>>I agree with freeswitch philosophy that instead of having
thousands of
modules
that don't work fine i prefer just a few that works the way it should
be, a rock solid system for a corporate pbx and a call center is what i
want.
>>>>>
>>>>>So here i am trying to begin the conversion, and i hope the
information
we can transcript in this list will help others that want to try
another
alternative to asterisk.
>>>>>
>>>>>First of all i think the saner for a migration is to have the two
systems
running either on the same machine or different and use the stable
features of each one.
>>>>>
>>>>>So could you please freeswitch users help me with this rosetta
stone migration guide in order to post it to voip-info.org or
freeswitch wiki (i list only the ones i currently use ):
>>>>>
>>>>>
>>>>>
>>>>>Technology Asterisk Freeswitch 
>>>>>PSTN Connectivity (Digium /
Sangoma) dahdi freetdm 
>>>>>IAX2 mod_iax ?? none stable yet.
>>>>>Use Asterisk to forward traffic via SIP.
>>>>>Enable Hardware HPET for IAX2 trunk if card not available for Asterisk  
>>>>>Bluetooth Channel chan_mobile ??
>>>>>Use asterisk via SIP
>>>>> 
>>>>>Skype Skypeforasterisk (no longer for sale) mod_skypeopen 
>>>>>CDR Stadistics Arternic cdr-stats  ?? 
>>>>>Queue Statistics  Asteriskguru queue-stats ?? 
>>>>>Web Management Freepbx ?? 
>>>>>IVR AGI / AMI Event Socket 
>>>>>Codec G.729 Transcodind Cards
>>>>>G.729 licenses
>>>>>Free G.729 (Intel IPP) Transcodind Cards
>>>>>G.729 licenses
>>>>>fsg729 Intel IPP(any experience with it ? ) 
>>>>>Fax Handling Iaxmodem with Hylafax ??
>>>>>Iaxmodem via asterisk to FS via SIP ?
>>>>> 
>>>>>SIP chan_sip sofia 
>>>>>ACD app_queue mod_callcenter 
>>>>>
>>>>>Thank you all
>>>>>
>>>>>
>>>>>-- 
>>>>>Nestor A. Diaz
>>>>>Ingeniero de Sistemas
>>>>>Tel. +57 1-485-3020 x 211
>>>>>Cel. +57 316-227-3593
>>>>>Tel. SIP: sip:211 at tiendalinux.com
>>>>>Email/MSN: nestor at tiendalinux.com
>>>>>http://www.tiendalinux.com/
>>>>>Bogota, Colombia 
>>>>> 
>>>>>
>>>>>
>>>>>_______________________________________________
>>>>>Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>>FreeSWITCH-users mailing list
>>>>>FreeSWITCH-users at lists.freeswitch.org
>>>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>>_______________________________________________
>>>>>Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>>FreeSWITCH-users mailing list
>>>>>FreeSWITCH-users at lists.freeswitch.org
>>>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>>-- 
>>>>Anthony Minessale II
>>>>
>>>>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
>>>>googletalk:conf+888 at conference.freeswitch.org
>>>>pstn:+19193869900
>>>>
>>>>_______________________________________________
>>>>Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>>FreeSWITCH-users mailing list
>>>>FreeSWITCH-users at lists.freeswitch.org
>>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>>FreeSWITCH-users mailing list
>>>>FreeSWITCH-users at lists.freeswitch.org
>>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>>
>>>-- 
>>>Anthony Minessale II
>>>
>>>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
>>>googletalk:conf+888 at conference.freeswitch.org
>>>pstn:+19193869900
>>>
>>>
>>>FreeSWITCH-users mailing list
>>>FreeSWITCH-users at lists.freeswitch.org
>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>>FreeSWITCH-users mailing list
>>>FreeSWITCH-users at lists.freeswitch.org
>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>http://www.freeswitch.org
>>>
>>>
>>
>>
>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>>
>>
>>
>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>>
>
>
>
>-- 
>Anthony Minessale II
>
>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
>googletalk:conf+888 at conference.freeswitch.org
>pstn:+19193869900
>
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>
>
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>


FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/cc436f54/attachment-0001.html 

From peter.olsson at visionutveckling.se  Thu Aug 18 00:42:33 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 17 Aug 2011 22:42:33 +0200
Subject: [Freeswitch-users] Paging with Polycom Phones
In-Reply-To: <CAKzWOxW27QZx=5MLKxtVZaHCr_y0zWpXUEx59_sfmDBu4C_BTA@mail.gmail.com>
References: <CAC4JBEAF4OMFeD9KV5LSAVZ7DxTG5ceK0Vj3cDEam6kU8hTD3Q@mail.gmail.com>
	<CAKzWOxUMbUYM_+eiQG98OpAYTHdur1GVADHzY1OmfV-E+NBHDw@mail.gmail.com>
	<CAC4JBECX77JNE5htRm2nD8UOtfhVLqK2tnp3N2tnN+ep6DOAqA@mail.gmail.com>,
	<CAKzWOxW27QZx=5MLKxtVZaHCr_y0zWpXUEx59_sfmDBu4C_BTA@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59EBABB8D4@cooper>

I Also tried with default settings, worked fine also..

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Michael Collins [msc at freeswitch.org]
Skickat: den 17 augusti 2011 22:29
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Paging with Polycom Phones

I have plain jane out-of-box Polycoms and they all seem to work. I can't tell you if there are any magic settings.

Does anyone else know if there are specific settings on the Polys that need to be set in order for the mad-boss to work?

-MC

On Wed, Aug 17, 2011 at 11:33 AM, Duvid Rottenberg <adrottenberg at gmail.com<mailto:adrottenberg at gmail.com>> wrote:
The mad boss example looks like it uses the same commands as the conferencing and intercom sample. I copied the  few settings that appear to be different, but I'm still having the issue that the polycom phone answers and hangs up right away. Did you have to make any changes to your polycom config file to get this to work?

Thanks,
Duvid

On Wed, Aug 17, 2011 at 12:51 PM, Michael Collins <msc at freeswitch.org<mailto:msc at freeswitch.org>> wrote:
Try using the "mad boss" example found in the default dialplan. I've tested that with Polycom phones. It's a nice workaround for all these different phone vendors who do things so differently.

-MC

On Wed, Aug 17, 2011 at 8:55 AM, Duvid Rottenberg <adrottenberg at gmail.com<mailto:adrottenberg at gmail.com>> wrote:
Has anyone successfully implemented paging (auto-answer) with a polycom phone?
I am using the Conferencing and Intercom sample which sets the sip_auto_answer variable to true, however on my polycom phone the result is that the phone rings once and hangs up right away (the phone is sending a BYE message). I tried adding an Alert-Info header, however it seems that the Polycom format (Alert-Info: Ring Answer) isn't compliant with the RFC and I couldn't get freeswitch to send the header in this format.

Has anyone else been able to either get polycom phones to work with sip_auto_answer or to get freeswitch to send an Alert-Info header in the polycom format?

Thanks,
Duvid Rottenberg


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




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




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


!DSPAM:4e4c25df32761815520130!


From avi at avimarcus.net  Thu Aug 18 01:47:31 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 18 Aug 2011 00:47:31 +0300
Subject: [Freeswitch-users] Inband DTMF when rfc2833 negotiated?
In-Reply-To: <CAKzWOxVOVRvg3_Tu0e1_dh-8aE3p9pp1QcfUxOxve7OECRmSyw@mail.gmail.com>
References: <CALNZuwkD8q5L7ErTw7rLTTRD=ZB4QNmNF_nbyLuhL5cAwPdFmw@mail.gmail.com>
	<CAKzWOxVOVRvg3_Tu0e1_dh-8aE3p9pp1QcfUxOxve7OECRmSyw@mail.gmail.com>
Message-ID: <CALNZuw=SQop2qxrLKxcLHiTOmt2Tp_3BqzPeoL-2W_HBA5yScA@mail.gmail.com>

The caller, on A leg, is received as INFO, and then FS is sending it as
INBAND on the B leg.
See the trace? It shows sending as inband.

-Avi


On Wed, Aug 17, 2011 at 7:53 PM, Michael Collins <msc at freeswitch.org> wrote:

> Which end is sending inband? Do you have a pcap of this call?
> -MC
>
> On Tue, Aug 16, 2011 at 5:44 AM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> Why is this call using inband when the SDP says rfc2833? I got complaints
>> that the DTMF wasn't working, despite the dtmf numbers in the log being
>> correct.
>>
>> Trace on leg B: http://pastebin.freeswitch.org/17053
>>
>>
>> Thanks,
>> -Avi
>>
>> p.s. I think I've seen this with toll free gateway, too.
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110818/6053a424/attachment.html 

From msc at freeswitch.org  Thu Aug 18 01:53:39 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 17 Aug 2011 14:53:39 -0700
Subject: [Freeswitch-users] Bad port when using rport
In-Reply-To: <CFCBEE64-C506-4A74-8203-7F135AEC248A@5ninesolutions.com>
References: <CFCBEE64-C506-4A74-8203-7F135AEC248A@5ninesolutions.com>
Message-ID: <CAKzWOxVY+iipLRtJuUeor2G8Bo63jDxb-XXF=j2bZFs1pDUofg@mail.gmail.com>

This is what you received from the far end? That looks horribly broken. Time
to find out what device and what firewall.

-MC

On Wed, Aug 17, 2011 at 10:21 AM, Spencer Thomason <
spencer at 5ninesolutions.com> wrote:

> Hello all,
> Our FS server is on a public IP on a dedicated server at one location and
> all clients are behind nat or vpn from other locations.  We have a few "road
> warrior" clients when our employees travel.  I'm seeing the following from
> one of the client devices.  I'm not sure of any of the details of the router
> (or have anyway to control it) but for some reason the rport is sent with an
> invalid port number.  Notice the 8 at the end of the port number is
> duplicated.  I'm seeing the same behavior from both a Bria client and
> Zoiper.  Is there any way to work around this?
>
>
> An excerpt from Wireshark:
>
>    466 2011-08-17 11:36:04.432912 80.34.4.95            206.125.40.171
>    SIP      Request: REGISTER sip:x.x.x.x
>
> Internet Protocol, Src: 80.34.4.95 (80.34.4.95), Dst: 206.125.40.171
> (x.x.x.x)
> User Datagram Protocol, Src Port: 56238 (56238), Dst Port: sip (5060)
> Session Initiation Protocol
>    Request-Line: REGISTER sip:x.x.x.x SIP/2.0
>    Message Header
>        Via: SIP/2.0/UDP 80.34.4.95:562388
> ;rport;branch=z9hG4bKPjSAbzV32ywqJ8xWjWLx81Ml6oQcNlrM.s
>        Max-Forwards: 70
>        From: "XXXX" <sip:1001 at x.x.x.x
> >;tag=rPFQBJFIKAQpE88mHeiZ7ic59suQY70w
>        To: "XXXX" <sip:1001 at x.x.x.x>
>        Call-ID: awr-KAMoJpMYEt2mqkOkknkRrCXUj.Jv
>        CSeq: 27137 REGISTER
>        User-Agent: Bria iPhone 1.2.12
>        Contact: "XXXX" <sip:1001 at 80.34.4.95:562388>
>        Expires: 900
>        Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY,
> REFER, MESSAGE, OPTIONS
>        Content-Length:  0
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/d26231ee/attachment.html 

From msc at freeswitch.org  Thu Aug 18 02:05:26 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 17 Aug 2011 15:05:26 -0700
Subject: [Freeswitch-users] Simulating Asterisk dial(x:y:z)
In-Reply-To: <1313566426.80683.YahooMailNeo@web161019.mail.bf1.yahoo.com>
References: <1313566426.80683.YahooMailNeo@web161019.mail.bf1.yahoo.com>
Message-ID: <CAKzWOxVNM5pqxrD720JJ0p9VDmO1nWC8fyn0CUn36XbdWY1QKA@mail.gmail.com>

You can do multiple api_on_answer calls by using a relatively new syntax:

api_on_answer_1=api(arg),api_on_answer_2=api2(arg2),api_on_answer_3=api3(arg3)

Same syntax applies to the execute_on_xxx variables.

-MC

On Wed, Aug 17, 2011 at 12:33 AM, Sam <lakersman2006 at yahoo.com> wrote:

> I want to simulate the Asterisk dial(x:y:z) app where (x=total call time,
> y=warning time, z=warning interval) in Freeswitch. I know it can be done
> with sched_hangup and sched_broadcast, my question is how can I call a
> series of "api_on_answer" comands to do it?
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/72d99f44/attachment.html 

From lakersman2006 at yahoo.com  Thu Aug 18 03:39:36 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Wed, 17 Aug 2011 16:39:36 -0700 (PDT)
Subject: [Freeswitch-users] Simulating Asterisk dial(x:y:z)
In-Reply-To: <CAKzWOxVNM5pqxrD720JJ0p9VDmO1nWC8fyn0CUn36XbdWY1QKA@mail.gmail.com>
References: <1313566426.80683.YahooMailNeo@web161019.mail.bf1.yahoo.com>
	<CAKzWOxVNM5pqxrD720JJ0p9VDmO1nWC8fyn0CUn36XbdWY1QKA@mail.gmail.com>
Message-ID: <1313624376.73435.YahooMailNeo@web161001.mail.bf1.yahoo.com>

MC,

I tried the syntax you suggested but I does not seem to be working

<action application="export" data="nolocal:api_on_answer_1=sched_broadcast(+1 ${uuid} playback::sound1.wav aleg), nolocal:api_on_answer_2=sched_hangup(+30 ${uuid})"/>


I also tried this syntax and still no luck

<action application="export" data="api_on_answer_1=sched_broadcast +1 ${uuid}?
playback::sound1.wav"/> 
<action application="export" data="api_on_answer_2=sched_hangup +15 ${uuid}"/> 


________________________________
From: Michael Collins <msc at freeswitch.org>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, August 17, 2011 3:05 PM
Subject: Re: [Freeswitch-users] Simulating Asterisk dial(x:y:z)


You can do multiple api_on_answer calls by using a relatively new syntax:

api_on_answer_1=api(arg),api_on_answer_2=api2(arg2),api_on_answer_3=api3(arg3)

Same syntax applies to the execute_on_xxx variables.

-MC


On Wed, Aug 17, 2011 at 12:33 AM, Sam <lakersman2006 at yahoo.com> wrote:

I want to simulate the Asterisk dial(x:y:z) app where (x=total call time, y=warning time, z=warning interval) in Freeswitch. I know it can be done with sched_hangup and sched_broadcast, my question is how can I call a series of "api_on_answer" comands to do it?
>
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>


FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/9bfc016b/attachment-0001.html 

From msc at freeswitch.org  Thu Aug 18 04:11:37 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 17 Aug 2011 17:11:37 -0700
Subject: [Freeswitch-users] Simulating Asterisk dial(x:y:z)
In-Reply-To: <1313624376.73435.YahooMailNeo@web161001.mail.bf1.yahoo.com>
References: <1313566426.80683.YahooMailNeo@web161019.mail.bf1.yahoo.com>
	<CAKzWOxVNM5pqxrD720JJ0p9VDmO1nWC8fyn0CUn36XbdWY1QKA@mail.gmail.com>
	<1313624376.73435.YahooMailNeo@web161001.mail.bf1.yahoo.com>
Message-ID: <CAKzWOxXXv40KF8Jf-tLDYQ9ApxkW0f+4BqNwM=J9VxRFq0Wz-g@mail.gmail.com>

Hmmm,
I wonder if the multiple syntax only made it into the execute_on_xxx vars.
I'll ask Brian about that. In the meantime it is possible to use
execute_on_answer and execute an extension that in turn does all these API
calls on your channel:

<action application="export"
data="nolocal:execute_on_answer=execute_extension do_stuff XML
custom_context"/>

Then have a simple context. Add file conf/dialplan/custom.xml:
<include>
  <context name="custom_context">
    <extension name="do my stuff">
      <condition field="destination_number" expression="^do_stuff$">
        <action application="set" data="res1=${sched_broadcast(+1 ${uuid}
playback::sound1.wav aleg)}"/>
        <action application="set" data="res2=${sched_hangup(+30 ${uuid})}"/>
      </condition>
    </extension>
  </context>
</include>

You can try that one for kicks while we research the api_on_answer question.

-MC

On Wed, Aug 17, 2011 at 4:39 PM, Sam <lakersman2006 at yahoo.com> wrote:

> MC,
>
> I tried the syntax you suggested but I does not seem to be working
>
> <action application="export"
> data="nolocal:api_on_answer_1=sched_broadcast(+1 ${uuid}
> playback::sound1.wav aleg), nolocal:api_on_answer_2=sched_hangup(+30
> ${uuid})"/>
>
>
> I also tried this syntax and still no luck
>
> <action application="export" data="api_on_answer_1=sched_broadcast +1
> ${uuid}  playback::sound1.wav"/>
> <action application="export" data="api_on_answer_2=sched_hangup +15
> ${uuid}"/>
>
> ------------------------------
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Wednesday, August 17, 2011 3:05 PM
> *Subject:* Re: [Freeswitch-users] Simulating Asterisk dial(x:y:z)
>
> You can do multiple api_on_answer calls by using a relatively new syntax:
>
>
> api_on_answer_1=api(arg),api_on_answer_2=api2(arg2),api_on_answer_3=api3(arg3)
>
> Same syntax applies to the execute_on_xxx variables.
>
> -MC
>
> On Wed, Aug 17, 2011 at 12:33 AM, Sam <lakersman2006 at yahoo.com> wrote:
>
> I want to simulate the Asterisk dial(x:y:z) app where (x=total call time,
> y=warning time, z=warning interval) in Freeswitch. I know it can be done
> with sched_hangup and sched_broadcast, my question is how can I call a
> series of "api_on_answer" comands to do it?
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/af5a277e/attachment.html 

From bryan at bryanlemon.com  Thu Aug 18 07:08:41 2011
From: bryan at bryanlemon.com (Bryan Lemon)
Date: Wed, 17 Aug 2011 23:08:41 -0400
Subject: [Freeswitch-users] Question about ext-rtp-ip and ext-sip-ip
In-Reply-To: <7C7183C2-3601-47D4-B8DE-D9E292B592D3@freeswitch.org>
References: <CAEjbKNQ3tgWrVrr5uiTVKzv53avDLp3cRoavLuZsOiJfUaW_fg@mail.gmail.com>
	<65727391-DF08-4074-BB7F-BDB766DF7942@freeswitch.org>
	<CAEjbKNSGi2prQqoeLSGbKjtYbqOsD7-hggPeEUcVEWPgtCp2UQ@mail.gmail.com>
	<CAEjbKNTt8Fd=Q6TnGz407j5N4zfe4X8FSLpf2ttmpUydT9YJpA@mail.gmail.com>
	<7C7183C2-3601-47D4-B8DE-D9E292B592D3@freeswitch.org>
Message-ID: <CAEjbKNT3oQJZ34FZFuq7JRxRV7zo7GaHdaybBRYYrSCKSTY_CQ@mail.gmail.com>

I tried it, and it still does not work.

Relevant details:
freeswitch at MEDIAPC> sofia status profile external

=================================================================================================
Name             external
Domain Name       N/A
Auto-NAT         true
DBName           sofia_reg_external
Pres Hosts
Dialplan         XML
Context           public
Challenge Realm   auto_to
RTP-IP           10.0.10.144
Ext-RTP-IP       204.111.*.*
SIP-IP           10.0.10.144
Ext-SIP-IP       204.111.*.*
URL               sip:mod_sofia at 10.0.10.144:5080
BIND-URL         sip:mod_sofia at 10.0.10.144:5080
HOLD-MUSIC       local_stream://moh


external.xml
<profile name="external">
  <gateways>
    <X-PRE-PROCESS cmd="include" data="external/*.xml"/>
  </gateways>

  <aliases>
    <!--


    <alias name="outbound"/>


    <alias name="nat"/>


    -->
  </aliases>

  <domains>
    <domain name="all" alias="false" parse="true"/>
  </domains>

  <settings>
    <param name="debug" value="0"/>
    <param name="sip-trace" value="no"/>
    <param name="rfc2833-pt" value="101"/>
    <param name="sip-port" value="$${external_sip_port}"/>
    <param name="dialplan" value="XML"/>
    <param name="context" value="public"/>
    <param name="dtmf-duration" value="2000"/>
    <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
    <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
    <param name="hold-music" value="$${hold_music}"/>
    <param name="rtp-timer-name" value="soft"/>
    <param name="local-network-acl" value="localnet.auto"/>
    <param name="manage-presence" value="false"/>
    <!--<param name="aggressive-nat-detection" value="true"/>-->
    <param name="inbound-codec-negotiation" value="generous"/>
    <param name="nonce-ttl" value="60"/>
    <param name="auth-calls" value="false"/>
    <param name="rtp-ip" value="$${local_ip_v4}"/>
    <param name="sip-ip" value="$${local_ip_v4}"/>
    <param name="ext-rtp-ip" value="autonat:204.111.187.118"/>
    <param name="ext-sip-ip" value="autonat:204.111.187.118"/>
    <param name="rtp-timeout-sec" value="300"/>
    <param name="rtp-hold-timeout-sec" value="1800"/>
    <param name="tls" value="$${external_ssl_enable}"/>
    <param name="tls-bind-params" value="transport=tls"/>
    <param name="tls-sip-port" value="$${external_tls_port}"/>
    <param name="tls-cert-dir" value="$${external_ssl_dir}"/>
    <param name="tls-version" value="$${sip_tls_version}"/>

  </settings>
</profile>

Thank you,
Bryan Lemon
(302) 648-2747



On Wed, Aug 17, 2011 at 12:55, Brian West <brian at freeswitch.org> wrote:

> You need to change these to be "autonat:x.x.x.x"  and specify the IP to
> use.
>
> /b
>
> On Aug 17, 2011, at 10:10 AM, Bryan Lemon wrote:
>
> >    <param name="ext-rtp-ip" value="auto-nat"/>
> >    <param name="ext-sip-ip" value="auto-nat"/>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/da0f0766/attachment.html 

From govoiper at gmail.com  Thu Aug 18 08:15:02 2011
From: govoiper at gmail.com (Sam Govind)
Date: Thu, 18 Aug 2011 09:15:02 +0500
Subject: [Freeswitch-users] SIP proxy collect DTMF using FS
In-Reply-To: <4e4b4abb.11a5960a.4abd.6c8dSMTPIN_ADDED@mx.google.com>
References: <CAJUJwtjqMyvVX1eteYBWFf41Mz+fZjnRZA_0BR2DZs0R5zh_tA@mail.gmail.com>
	<0D92C5B0-CD84-47B1-A17C-A2B083B760E2@freeswitch.org>
	<CAJUJwthVmvx3vvHwpO+ZOS0tDaLREp4H3OfX39hBJQ0cJOvp_Q@mail.gmail.com>
	<4e4b4abb.11a5960a.4abd.6c8dSMTPIN_ADDED@mx.google.com>
Message-ID: <CAJUJwtgR0ad5V=G0MEQVYJBtLdrqS+9KOxA35h_gvWMWr1O=Bg@mail.gmail.com>

Thanks for taking your time Gohar, I've looked at the link and this indeed
seems like helping.

Brain, can you suggest any other possibilities for this? I'd definitely like
to go for the simple solution.



On Wed, Aug 17, 2011 at 9:54 AM, Gohar Ahmed <gohar.ahmed at vopium.com> wrote:

> Hey, ****
>
> ** **
>
> Don?t know if this will do the job or not but do take a look at deflect
> application: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect
> ****
>
> This will remove the FS from the call path and tell the originator to go
> somewhere else. ****
>
> ** **
>
> I hope this helps, maybe some other gurus here suggest you anything better
> than this.****
>
> ** **
>
> Regards,****
>
> Gohar A.****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Sam Govind
> *Sent:* Monday, August 15, 2011 10:58 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] SIP proxy collect DTMF using FS****
>
> ** **
>
> Thanks Brian for showing concern. I'm always open for ideas. What I'm
> trying to achieve is collect DTMF from user and then have my SIP Proxy
> verify if a particular caller is allowed to dial that destination(input as
> DTMF).****
>
> ** **
>
> Obviously I could've done the same checks at FS, BUT I'm *required *to let
> SIP Proxy verify instead. FreeSWITCH is only required to get input, release
> the call and if Proxy allows the call only then call be routed to any other
> FreeSWITCH (Pool of FS LoadBalanced).****
>
> ** **
>
> This is supposed to simplify the operations of FS , decrease the load
> volume on FS, and increase the call capacity.****
>
> ** **
>
> If you've any better Ideas do share.****
>
> ** **
>
> On Sun, Aug 14, 2011 at 8:39 AM, Brian West <brian at freeswitch.org> wrote:*
> ***
>
> What are you trying to do exactly because it sounds like you have selected
> the most painful way to accomplish it?****
>
> ** **
>
> /b****
>
> ** **
>
> On Aug 9, 2011, at 2:11 AM, Sam Govind wrote:****
>
>
>
> ****
>
> Hi guys,
>
> I'm looking to establish a scenario like this, any idea how to do it, if
> its
> possible.
>
> 1- SIP proxy send call to FS where DTMF will be collected. (I'm thinking of
> using PlayAndGetDigits)
> 2- DTMF collected be sent back to SIP proxy while FS ends the call
> 3- Call at SIP proxy end keeps running for some other processing.
>
> basically I just need FS to collect DTMF and send those back to SIP Proxy.
>
> Any ideas are welcome.
>
> Thanks.****
>
> ** **
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
> ** **
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110818/67c516d0/attachment-0001.html 

From kbdfck at gmail.com  Thu Aug 18 08:31:57 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Thu, 18 Aug 2011 08:31:57 +0400
Subject: [Freeswitch-users] Dynamically switching DTMF mode?
Message-ID: <CA+GmxbbVCMr7g3o+eDvcMF7tNY=T8BZ0_CyU56mxNbvjPz-2xw@mail.gmail.com>

Hi all

I'm trying to achieve dynamic DTMF mode on incoming/outgoing calls. I have
three profiles - one for outgoing/incoming calls to/from gateways, one for
local registrations, and one for transferred call rerouting from
bind_meta_app att_xfer. I faced some strange problems when tried to maintain
inband DTMF path for call through FS.

How should I set initial DTMF modes on profiles to allow changing DTMF mode
on inbound calls? Is it enough to set dtmf_type to 'inband' and originate
call? On which leg of call I should do this? And what to do with calls going
through several profiles?

As far as I understand, I need late negotiation turned on to disable
automatic RFC2833 negotiation on inbound calls?

-- 
Best regards,

Dmitry Sytchev,
IT Engineer
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110818/9cf2031a/attachment.html 

From ovvenkatesan at gmail.com  Thu Aug 18 08:46:54 2011
From: ovvenkatesan at gmail.com (ovvenkat)
Date: Thu, 18 Aug 2011 10:16:54 +0530
Subject: [Freeswitch-users] [Freeswitch-dev] Meetup Paris?
In-Reply-To: <CAFSQhGWmgZH4QGET7uBujsZvrRsVeubBd+b-W4HL26K99bbKaw@mail.gmail.com>
References: <E58D1B62-B7E9-4D15-A0AF-DC993AA70786@avgs.ca>
	<CAPHrvXHx-3u0hvQ2WzRZ7M1K5OTO6J-Bur=hC1z5wxXN43p2rA@mail.gmail.com>
	<CALXCt0pzErc5beBzNXJ_niDVm+4X5W1CTkV8Ts28vK=hoDD5ww@mail.gmail.com>
	<CAPHrvXG=QQWeu1YHbP-t6X8fpY9c7MCn-bajZou1MOnxkwx_uA@mail.gmail.com>
	<CAFSQhGWmgZH4QGET7uBujsZvrRsVeubBd+b-W4HL26K99bbKaw@mail.gmail.com>
Message-ID: <CAABykDpy=0fgJiKKnoVqc9uO_zB2CyRVF32E7hs8i1MsAuNPsA@mail.gmail.com>

Hi Prashant and guru,

We are dong FS related stuff based in Chennai.
I believe that there are many freeswitchers in india.
All are scattered throughout india.
This is the time to take some initiative for meetup.

Regards,
Venkat.



On Wed, Aug 17, 2011 at 9:49 PM, guru singh <grsingh750 at gmail.com> wrote:

> Hi Prashant,
>
> I'm a FS user based in New Delhi. All the people doing voice related
> stuff I've met or exchanged emails with are Asterisk users, some had
> not even heard of FS. I'd be interested in trying to get some
> initiative/meetup going.
>
> Regards
> guru
>
> On Wed, Aug 17, 2011 at 3:04 PM, Prashant Lamba
> <prashant.lamba at gmail.com> wrote:
> > On Tue, Aug 16, 2011 at 3:23 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> > wrote:
> >>
> >> I have fond memories of a conference I gave couple of years ago in
> >> Hyderabad and the nice reception has had.
> >>
> >> Certainly there are lot of people and companies that can be interested
> >> in FreeSWITCH, and Cluecon India.
> >>
> >> Most people is still on Asterisk, but I believe they'll be very
> >> interested into knowing first hand what are the advantages of using
> >> FS, maybe in powerful combination with OpenSips.
> >>
> >> Let's have the ball rolling, try to gauge what interest you can
> >> gather, and which companies can act as sponsors (money matters ;) ).
> >>
> >> -giovanni
> >
> >
> > Giovanni, I am all for it especially since there is very little awareness
> of
> > FreeSWITCH in India compared to Asterisk. Anyone in?
> >
> > Prashant, Phonologies (India)
> > prashant.lamba at gmail.com
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

If you have come to help me, you are wasting your time.
If you have come to because your liberation is bound up in mine, we can work
together.


Regards
Venkatesan OV.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110818/ce1d7c9b/attachment.html 

From simon0922 at gmail.com  Thu Aug 18 09:11:25 2011
From: simon0922 at gmail.com (Simon Leck)
Date: Thu, 18 Aug 2011 13:11:25 +0800
Subject: [Freeswitch-users] Freeswitch ignoring bye
Message-ID: <035801cc5d65$49127c10$db377430$@gmail.com>

Hi Everyone,

 

Does anybody does why freeswitch is not acknowledging or ignoring BYE
message and if so how do we fix it?

 

Thanks in advanced to anyone who can tell me how to fix this issue. 

 

Thanks again

Simon

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110818/d90e778a/attachment.html 

From msc at freeswitch.org  Thu Aug 18 09:26:44 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 17 Aug 2011 22:26:44 -0700
Subject: [Freeswitch-users] Dynamically switching DTMF mode?
In-Reply-To: <CA+GmxbbVCMr7g3o+eDvcMF7tNY=T8BZ0_CyU56mxNbvjPz-2xw@mail.gmail.com>
References: <CA+GmxbbVCMr7g3o+eDvcMF7tNY=T8BZ0_CyU56mxNbvjPz-2xw@mail.gmail.com>
Message-ID: <CAKzWOxV1Snk9AeU62bCn7vzXDGECpj3+17M3UUm+=SpRtaVGzg@mail.gmail.com>

Are you talking about sending DTMFs or detecting DTMFs?
-MC

On Wed, Aug 17, 2011 at 9:31 PM, Dmitry Sytchev <kbdfck at gmail.com> wrote:

> Hi all
>
> I'm trying to achieve dynamic DTMF mode on incoming/outgoing calls. I have
> three profiles - one for outgoing/incoming calls to/from gateways, one for
> local registrations, and one for transferred call rerouting from
> bind_meta_app att_xfer. I faced some strange problems when tried to maintain
> inband DTMF path for call through FS.
>
> How should I set initial DTMF modes on profiles to allow changing DTMF mode
> on inbound calls? Is it enough to set dtmf_type to 'inband' and originate
> call? On which leg of call I should do this? And what to do with calls going
> through several profiles?
>
> As far as I understand, I need late negotiation turned on to disable
> automatic RFC2833 negotiation on inbound calls?
>
> --
> Best regards,
>
> Dmitry Sytchev,
> IT Engineer
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/bbe7ffbf/attachment.html 

From peter.olsson at visionutveckling.se  Thu Aug 18 00:33:19 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 17 Aug 2011 22:33:19 +0200
Subject: [Freeswitch-users] Multiple Registration
In-Reply-To: <CAAB3jWuzV-m=N6cN+ftSme+Ls9vuteWYMjc8TgKPm17Z3Do-Sg@mail.gmail.com>
References: <CAAB3jWuzV-m=N6cN+ftSme+Ls9vuteWYMjc8TgKPm17Z3Do-Sg@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59EBABB8D3@cooper>

1. Yes, this is normal.

2. You'll see only the last one who registered (if mutliple-registration was not enabled). So if you call out to this number, it will only ring on the phone that last registered.


/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r DJB International [djbinter at gmail.com]
Skickat: den 17 augusti 2011 22:05
Till: FREESWITCH-USERS MAILING LIST
?mne: [Freeswitch-users] Multiple Registration

I have some question regarding the multiple registration.  When I registered 2 devices, even though both of them got registered and I can make calls from both devices, but when I list, I can only see the last device that got registered on the list from sofia status profile internal.

However, when I comment out the multiple-registrations value=true in sofia profile, I can then see 2 devices listed.

Questions:

1)  Is that normal behavior that I need to make multiple-registrations value=true before I can see 2 devices listed when I run sofia status profile internal ?

2)  How come I can only see one device listed even though both devices got registered in the first situation?

Thank you,
Dorn B.


!DSPAM:4e4c20dd32762035271195!


From djbinter at gmail.com  Thu Aug 18 09:42:03 2011
From: djbinter at gmail.com (DJB International)
Date: Wed, 17 Aug 2011 22:42:03 -0700
Subject: [Freeswitch-users] Multiple Registration
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C59EBABB8D3@cooper>
References: <CAAB3jWuzV-m=N6cN+ftSme+Ls9vuteWYMjc8TgKPm17Z3Do-Sg@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59EBABB8D3@cooper>
Message-ID: <CAAB3jWuXkn8HX-jAz0dyEf=wLAtN4suuSikw8Vi5KjSHEoz-Eg@mail.gmail.com>

Thank you Avi and Peter.

Dorn B.

On Wed, Aug 17, 2011 at 1:33 PM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> 1. Yes, this is normal.
>
> 2. You'll see only the last one who registered (if mutliple-registration
> was not enabled). So if you call out to this number, it will only ring on
> the phone that last registered.
>
>
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [
> freeswitch-users-bounces at lists.freeswitch.org] f&#246;r DJB International
> [djbinter at gmail.com]
> Skickat: den 17 augusti 2011 22:05
> Till: FREESWITCH-USERS MAILING LIST
> ?mne: [Freeswitch-users] Multiple Registration
>
> I have some question regarding the multiple registration.  When I
> registered 2 devices, even though both of them got registered and I can make
> calls from both devices, but when I list, I can only see the last device
> that got registered on the list from sofia status profile internal.
>
> However, when I comment out the multiple-registrations value=true in sofia
> profile, I can then see 2 devices listed.
>
> Questions:
>
> 1)  Is that normal behavior that I need to make multiple-registrations
> value=true before I can see 2 devices listed when I run sofia status profile
> internal ?
>
> 2)  How come I can only see one device listed even though both devices got
> registered in the first situation?
>
> Thank you,
> Dorn B.
>
>
> !DSPAM:4e4c20dd32762035271195!
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110817/a29d1dce/attachment-0001.html 

From vermeulen.deon at gmail.com  Thu Aug 18 10:47:43 2011
From: vermeulen.deon at gmail.com (Deon Vermeulen)
Date: Thu, 18 Aug 2011 07:47:43 +0100
Subject: [Freeswitch-users] Sangoma Media Gateway
In-Reply-To: <017601cc5cfb$38b5b180$aa211480$@sirran.com>
References: <017601cc5cfb$38b5b180$aa211480$@sirran.com>
Message-ID: <8238755A-2A09-4C9B-BF9C-C57AEFCB9912@gmail.com>

Hi Ben

I do it as follows.

Go to File Editor -> dialplan -> default.xml

Look for  <extension name="to-sip">

At the very end of the extension there is a part that already shows you how to route to IP.

			<!-- uncomment this if you want to dial to a fixed IP addr -->
			<!-- <action application="bridge" data="sofia/internal/${destnumber}@192.168.1.1"/> -->


So you could have something like this.

			<action application="set" data="sip_contact_user_replacement=${destnumber}"/>			
			<action application="set" data="hangup_after_bridge=yes"/>
			<action application="bridge" data="sofia/internal/${destnumber}@192.168.1.1"/>
			<action application="hangup" data="${originate_disposition}"/>
		
If you need to setup a DID then you could do something like this.

	<extension name="to-switchboard" >
		   <condition field="destination_number" expression="xxxxxxxx">  ( REPLACE xxxxxx with the number dialled on the PSTN)
			<action application="set" data="span=${sip_h_X-FreeTDM-SpanName}"/>
			<action application="log" data="INFO Incoming CALL for switchboard on span=${span}"/>
			<action application="set" data="hangup_after_bridge=yes"/>
			<action application="bridge" data="sofia/internal/yyyyyyyyy at 192.168.1.1"/>   (REPLACE yyyyyy with the destination number of the SIP Phone this call needs to go to)
			<action application="hangup" data="${originate_disposition}"/>
		   </condition>
	</extension>


If your calls need to be routed through the WAN to branch locations etc and you don't want to use the default ulaw/alaw then you could force the codec on the B-Leg of the call to what ever codec you prefer. This will depend on what codecs is supported on the end devices as well as if your card supports transcoding.

It could look something like this:

	<extension name="to-switchboard" >
		   <condition field="destination_number" expression="xxxxxxxx">		
			<action application="set" data="span=${sip_h_X-FreeTDM-SpanName}"/>
			<action application="log" data="INFO Incoming CALL for switchboard on span=${span}"/>
			<action application="set" data="hangup_after_bridge=yes"/>
			<action application="bridge" data="sofia/internal/yyyyyyyyy at 192.168.1.1"/>
			<action application="hangup" data="{absolute_codec_string=GSM} ${originate_disposition}"/>
			<action application="hangup" data="${originate_disposition}"/>
		   </condition>
	</extension>


These are just simple examples you can use.

Hope this is helpful.


Regards
Deon


On Aug 17, 2011, at 5:32 PM, Ben Naylor wrote:

> Hello
>  
> Apologies if this is in the wrong place, I am a complete beginner at Freeswitch!
>  
> Has anyone had much joy with using the above to route calls from ISDN to a SIP provider?  I have setup the SMG which is a stripped down version of Freeswitch, but am struggling to work out what to configure to get this working.
> So far I have tried to set up an external gateway to my provider, but this hasn?t appeared in the SMG Web-gui as a SIP profile.  My provider doesn?t required auth by the way, they have just given me an IP to connect to.
>  
> Any help is greatly appreciated
>  
> Kind regards
>  
> Ben
>  
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110818/113ad65f/attachment.html 

From bnaylor at sirran.com  Thu Aug 18 11:37:08 2011
From: bnaylor at sirran.com (Ben Naylor)
Date: Thu, 18 Aug 2011 08:37:08 +0100
Subject: [Freeswitch-users] Sangoma Media Gateway
In-Reply-To: <8238755A-2A09-4C9B-BF9C-C57AEFCB9912@gmail.com>
References: <017601cc5cfb$38b5b180$aa211480$@sirran.com>
	<8238755A-2A09-4C9B-BF9C-C57AEFCB9912@gmail.com>
Message-ID: <003c01cc5d79$a394dd90$eabe98b0$@sirran.com>

Hi Deon

 

This is great, just what I was looking for!

 

I am in Germany implementing this over the next few days (or however long it
takes), so will start to play around with it today.

 

Many thanks for your help, very much appreciated!

 

Regards

 

Ben

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Deon
Vermeulen
Sent: 18 August 2011 07:48
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Sangoma Media Gateway

 

Hi Ben

 

I do it as follows.

 

Go to File Editor -> dialplan -> default.xml

 

Look for  <extension name="to-sip">

 

At the very end of the extension there is a part that already shows you how
to route to IP.

 

                                    <!-- uncomment this if you want to dial
to a fixed IP addr -->

                                    <!-- <action application="bridge"
data="sofia/internal/${destnumber}@192.168.1.1
<mailto:sofia/internal/$%7bdestnumber%7d at 192.168.1.1> "/> -->

 

 

So you could have something like this.

 

                                    <action application="set"
data="sip_contact_user_replacement=${destnumber}"/>


                                    <action application="set"
data="hangup_after_bridge=yes"/>

                                    <action application="bridge"
data="sofia/internal/${destnumber}@192.168.1.1
<mailto:sofia/internal/$%7bdestnumber%7d at 192.168.1.1> "/>

                                    <action application="hangup"
data="${originate_disposition}"/>

                        

If you need to setup a DID then you could do something like this.

 

            <extension name="to-switchboard" >

                          <condition field="destination_number"
expression="xxxxxxxx">  ( REPLACE xxxxxx with the number dialled on the
PSTN)

                                    <action application="set"
data="span=${sip_h_X-FreeTDM-SpanName}"/>

                                    <action application="log" data="INFO
Incoming CALL for switchboard on span=${span}"/>

                                    <action application="set"
data="hangup_after_bridge=yes"/>

                                    <action application="bridge"
data="sofia/internal/yyyyyyyyy at 192.168.1.1"/>   (REPLACE yyyyyy with the
destination number of the SIP Phone this call needs to go to)

                                    <action application="hangup"
data="${originate_disposition}"/>

                          </condition>

            </extension>

 

 

If your calls need to be routed through the WAN to branch locations etc and
you don't want to use the default ulaw/alaw then you could force the codec
on the B-Leg of the call to what ever codec you prefer. This will depend on
what codecs is supported on the end devices as well as if your card supports
transcoding.

 

It could look something like this:

 

            <extension name="to-switchboard" >

                           <condition field="destination_number"
expression="xxxxxxxx">             

                                    <action application="set"
data="span=${sip_h_X-FreeTDM-SpanName}"/>

                                    <action application="log" data="INFO
Incoming CALL for switchboard on span=${span}"/>

                                    <action application="set"
data="hangup_after_bridge=yes"/>

                                    <action application="bridge"
data="sofia/internal/yyyyyyyyy at 192.168.1.1"/>

                                    <action application="hangup"
data="{absolute_codec_string=GSM} ${originate_disposition}"/>

                                    <action application="hangup"
data="${originate_disposition}"/>

                           </condition>

            </extension>

 

 

These are just simple examples you can use.

 

Hope this is helpful.

 

 

Regards

Deon

 

 

On Aug 17, 2011, at 5:32 PM, Ben Naylor wrote:





Hello

 

Apologies if this is in the wrong place, I am a complete beginner at
Freeswitch!

 

Has anyone had much joy with using the above to route calls from ISDN to a
SIP provider?  I have setup the SMG which is a stripped down version of
Freeswitch, but am struggling to work out what to configure to get this
working.

So far I have tried to set up an external gateway to my provider, but this
hasn't appeared in the SMG Web-gui as a SIP profile.  My provider doesn't
required auth by the way, they have just given me an IP to connect to.

 

Any help is greatly appreciated

 

Kind regards

 

Ben

 


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

 

  _____  

No virus found in this message.
Checked by AVG - www.avg.com
Version: 10.0.1392 / Virus Database: 1520/3840 - Release Date: 08/17/11

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110818/0b582cde/attachment-0001.html 

From covici at ccs.covici.com  Thu Aug 18 11:37:29 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 18 Aug 2011 03:37:29 -0400
Subject: [Freeswitch-users] having problems building mono 2.10
Message-ID: <24703.1313653049@ccs.covici.com>

Hi.  I am using gentoo, so I have to build mono by compiling and when I
try to do this I get the following error:

make[8]: Entering directory
`/var/tmp/portage/dev-lang/mono-2.10.2-r1/work/mono-2.10.2/mcs/tools/gacutil'
MCS     [basic] gacutil.exe
Inconsistency detected by ld.so: dl-deps.c: 622: _dl_map_object_deps:
Assertion `nlist > 1' failed!
make[8]: *** [../../class/lib/basic/gacutil.exe] Error 127

Any assistance on this would be appreciated.

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From kbdfck at gmail.com  Thu Aug 18 12:00:11 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Thu, 18 Aug 2011 12:00:11 +0400
Subject: [Freeswitch-users] Dynamically switching DTMF mode?
In-Reply-To: <CAKzWOxV1Snk9AeU62bCn7vzXDGECpj3+17M3UUm+=SpRtaVGzg@mail.gmail.com>
References: <CA+GmxbbVCMr7g3o+eDvcMF7tNY=T8BZ0_CyU56mxNbvjPz-2xw@mail.gmail.com>
	<CAKzWOxV1Snk9AeU62bCn7vzXDGECpj3+17M3UUm+=SpRtaVGzg@mail.gmail.com>
Message-ID: <CA+GmxbaaFGchwuciBruR1Hy1r3Ay7PK2Oww48OtQzDDYm=LS1A@mail.gmail.com>

About passing it through, and detecting.

We use Linksys PAP2T which has troubles with RFC2833 conversion. Small parts
of inband DTMF leaked into audio channel when it converts DTMF to 2833. And
then, when RFC2833 part is delayed by bind_digit_action or bind_meta_app in
FS, PSTN gateway connected via external profile converts it back to inband
from rfc2833, but inband parts are already passed through to PSTN. This
leads to double/triple/nightmare DTMF on PSTN side. Situation getting worse
if there were several transfers with att_xfer, so multiple channels are in
chain and delay is getting bigger and bigger.

To solve this, we will use inband DTMF on customer devices, detect it in FS
with start_dtmf and pass to PSTN gateway. But in order to accomplish this, I
need to maintain clear inband DTMF path between multipel channels and
profiles in FS, including channels created by calls to att_xfer.

So I need to disable RFC2833 for external gateway according to internal
endpoint type on inbound calls from external profile and conversely.

I'm looking for right way to do this, and will appreciate any help.


2011/8/18 Michael Collins <msc at freeswitch.org>

> Are you talking about sending DTMFs or detecting DTMFs?
> -MC
>
> On Wed, Aug 17, 2011 at 9:31 PM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>
>> Hi all
>>
>> I'm trying to achieve dynamic DTMF mode on incoming/outgoing calls. I have
>> three profiles - one for outgoing/incoming calls to/from gateways, one for
>> local registrations, and one for transferred call rerouting from
>> bind_meta_app att_xfer. I faced some strange problems when tried to maintain
>> inband DTMF path for call through FS.
>>
>> How should I set initial DTMF modes on profiles to allow changing DTMF
>> mode on inbound calls? Is it enough to set dtmf_type to 'inband' and
>> originate call? On which leg of call I should do this? And what to do with
>> calls going through several profiles?
>>
>> As far as I understand, I need late negotiation turned on to disable
>> automatic RFC2833 negotiation on inbound calls?
>>
>> --
>> Best regards,
>>
>> Dmitry Sytchev,
>> IT Engineer
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Best regards,

Dmitry Sytchev,
IT Engineer
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110818/7832d67b/attachment.html 

From vermeulen.deon at gmail.com  Thu Aug 18 12:58:21 2011
From: vermeulen.deon at gmail.com (Deon Vermeulen)
Date: Thu, 18 Aug 2011 09:58:21 +0100
Subject: [Freeswitch-users] Sangoma Media Gateway
In-Reply-To: <003c01cc5d79$a394dd90$eabe98b0$@sirran.com>
References: <017601cc5cfb$38b5b180$aa211480$@sirran.com>
	<8238755A-2A09-4C9B-BF9C-C57AEFCB9912@gmail.com>
	<003c01cc5d79$a394dd90$eabe98b0$@sirran.com>
Message-ID: <2BB14827-43BB-4F4C-9B0F-1A38395F4C5F@gmail.com>

Hi Ben

Glad I could help.
I was also very new to FS when I setup SMG and took me quite a while to figure out how SMG is put together, etc...

Thanks to the excellent help and guidance from the Sangoma Support Team I could get my system up and running to my requirements.


Regards
Deon


On Aug 18, 2011, at 8:37 AM, Ben Naylor wrote:

> Hi Deon
>  
> This is great, just what I was looking for!
>  
> I am in Germany implementing this over the next few days (or however long it takes), so will start to play around with it today.
>  
> Many thanks for your help, very much appreciated!
>  
> Regards
>  
> Ben
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Deon Vermeulen
> Sent: 18 August 2011 07:48
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Sangoma Media Gateway
>  
> Hi Ben
>  
> I do it as follows.
>  
> Go to File Editor -> dialplan -> default.xml
>  
> Look for  <extension name="to-sip">
>  
> At the very end of the extension there is a part that already shows you how to route to IP.
>  
>                                     <!-- uncomment this if you want to dial to a fixed IP addr -->
>                                     <!-- <action application="bridge" data="sofia/internal/${destnumber}@192.168.1.1"/> -->
>  
>  
> So you could have something like this.
>  
>                                     <action application="set" data="sip_contact_user_replacement=${destnumber}"/>                                   
>                                     <action application="set" data="hangup_after_bridge=yes"/>
>                                     <action application="bridge" data="sofia/internal/${destnumber}@192.168.1.1"/>
>                                     <action application="hangup" data="${originate_disposition}"/>
>                        
> If you need to setup a DID then you could do something like this.
>  
>             <extension name="to-switchboard" >
>                           <condition field="destination_number" expression="xxxxxxxx">  ( REPLACE xxxxxx with the number dialled on the PSTN)
>                                     <action application="set" data="span=${sip_h_X-FreeTDM-SpanName}"/>
>                                     <action application="log" data="INFO Incoming CALL for switchboard on span=${span}"/>
>                                     <action application="set" data="hangup_after_bridge=yes"/>
>                                     <action application="bridge" data="sofia/internal/yyyyyyyyy at 192.168.1.1"/>   (REPLACE yyyyyy with the destination number of the SIP Phone this call needs to go to)
>                                     <action application="hangup" data="${originate_disposition}"/>
>                           </condition>
>             </extension>
>  
>  
> If your calls need to be routed through the WAN to branch locations etc and you don't want to use the default ulaw/alaw then you could force the codec on the B-Leg of the call to what ever codec you prefer. This will depend on what codecs is supported on the end devices as well as if your card supports transcoding.
>  
> It could look something like this:
>  
>             <extension name="to-switchboard" >
>                            <condition field="destination_number" expression="xxxxxxxx">            
>                                     <action application="set" data="span=${sip_h_X-FreeTDM-SpanName}"/>
>                                     <action application="log" data="INFO Incoming CALL for switchboard on span=${span}"/>
>                                     <action application="set" data="hangup_after_bridge=yes"/>
>                                     <action application="bridge" data="sofia/internal/yyyyyyyyy at 192.168.1.1"/>
>                                     <action application="hangup" data="{absolute_codec_string=GSM} ${originate_disposition}"/>
>                                     <action application="hangup" data="${originate_disposition}"/>
>                            </condition>
>             </extension>
>  
>  
> These are just simple examples you can use.
>  
> Hope this is helpful.
>  
>  
> Regards
> Deon
>  
>  
> On Aug 17, 2011, at 5:32 PM, Ben Naylor wrote:
> 
> 
> Hello
>  
> Apologies if this is in the wrong place, I am a complete beginner at Freeswitch!
>  
> Has anyone had much joy with using the above to route calls from ISDN to a SIP provider?  I have setup the SMG which is a stripped down version of Freeswitch, but am struggling to work out what to configure to get this working.
> So far I have tried to set up an external gateway to my provider, but this hasn?t appeared in the SMG Web-gui as a SIP profile.  My provider doesn?t required auth by the way, they have just given me an IP to connect to.
>  
> Any help is greatly appreciated
>  
> Kind regards
>  
> Ben
>  
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>  
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 10.0.1392 / Virus Database: 1520/3840 - Release Date: 08/17/11
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110818/6e67d899/attachment-0001.html 

From avi at avimarcus.net  Thu Aug 18 13:10:25 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 18 Aug 2011 12:10:25 +0300
Subject: [Freeswitch-users] Freeswitch ignoring bye
In-Reply-To: <035801cc5d65$49127c10$db377430$@gmail.com>
References: <035801cc5d65$49127c10$db377430$@gmail.com>
Message-ID: <CALNZuwnrrC7V0TUMuxm4_Tg=mOakO+YB8xAvZ5bFamqHH-SjnA@mail.gmail.com>

Can you pastebin a siptrace with the fs_cli log?
-Avi


On Thu, Aug 18, 2011 at 8:11 AM, Simon Leck <simon0922 at gmail.com> wrote:

> Hi Everyone,****
>
> ** **
>
> Does anybody does why freeswitch is not acknowledging or ignoring BYE
> message and if so how do we fix it?****
>
> ** **
>
> Thanks in advanced to anyone who can tell me how to fix this issue. ****
>
> ** **
>
> Thanks again****
>
> Simon****
>
> ** **
>
> ** **
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110818/656f1e9f/attachment.html 

From jakub.ouhrabka at gmail.com  Thu Aug 18 14:51:29 2011
From: jakub.ouhrabka at gmail.com (Jakub Ouhrabka)
Date: Thu, 18 Aug 2011 12:51:29 +0200
Subject: [Freeswitch-users] Freetdm bridge to SIP DTMF problem
In-Reply-To: <4E4CD94C.3030403@comgate.cz>
References: <4E4CD94C.3030403@comgate.cz>
Message-ID: <4E4CEEB1.3040202@gmail.com>

Hi,

we've changed OpenZAP to FreeTDM sometime ago and from that time we're
experiencing problems with DTMF.

We're bridging incoming ISDN30 calls from Sangoma A108DE card + libri +
FreeTDM to SIP outgoing calls. Freeswitch is recognizing DTMF correctly 
but DTMF is not send to outgoing SIP call - it's not present in RTP 
stream trace (we've setup RFC2833 DTMF signalling). When bridging the 
call back to ISDN30 DTMF works correctly.

We're using latest versions of all software as of 1 month ago.

Below are attached two snippets of logs: first is bridging back to 
ISDN30 where DTMF is sent to called party. Second is bridging to SIP 
where DTMF is not sent to the called party.

We've tried it with both <action application="start_dtmf" /> on and off 
with no success.

Any pointers how to investigate the issue?

Thanks,

Jakub

Log snippets:

ISDN30 to ISDN30 bridge - DTMF ok

2011-08-18 12:31:45.950158 [DEBUG] ftmod_wanpipe.c:1545 [s1c11][1:11] 
read wanpipe event 3
2011-08-18 12:31:45.950158 [DEBUG] ftmod_wanpipe.c:1415 [s1c11][1:11] 
Queuing wanpipe DTMF: 5
2011-08-18 12:31:45.950158 [DEBUG] ftdm_io.c:3504 [s1c11][1:11] Queuing 
DTMF 5 (debug = 0)
2011-08-18 12:31:45.950158 [DEBUG] mod_freetdm.c:733 Queuing DTMF [5] in 
channel FreeTDM/1:11/XXXXXXXXX
2011-08-18 12:31:45.970623 [DEBUG] switch_ivr_bridge.c:391 Send signal 
FreeTDM/1:5/XXXXXXXXX [BREAK]
2011-08-18 12:31:45.991084 [DEBUG] ftdm_io.c:3694 [s1c5][1:5] Generating 
DTMF [5]

ISDN30 to SIP bridge - DTMF not forwarded

2011-08-18 12:34:05.287480 [DEBUG] ftmod_wanpipe.c:1545 [s1c29][1:29] 
read wanpipe event 3
2011-08-18 12:34:05.607708 [DEBUG] ftmod_wanpipe.c:1545 [s1c29][1:29] 
read wanpipe event 3
2011-08-18 12:34:05.607708 [DEBUG] ftmod_wanpipe.c:1415 [s1c29][1:29] 
Queuing wanpipe DTMF: 6
2011-08-18 12:34:05.607708 [DEBUG] ftdm_io.c:3504 [s1c29][1:29] Queuing 
DTMF 6 (debug = 0)
2011-08-18 12:34:05.607708 [DEBUG] ftmod_libpri.c:1586 -- Caught Event 
span 1 18 (KEYPAD_DIGIT)
2011-08-18 12:34:05.628171 [DEBUG] mod_freetdm.c:733 Queuing DTMF [6] in 
channel FreeTDM/1:29/XXXXXXXXXXXXXXX
2011-08-18 12:34:05.628171 [DEBUG] switch_ivr_bridge.c:391 Send signal 
sofia/external/XXXXXXX at XXXXXXX:5060 [BREAK]



From tomasz at hyziak.pl  Thu Aug 18 13:13:04 2011
From: tomasz at hyziak.pl (Tomasz Hyziak)
Date: Thu, 18 Aug 2011 11:13:04 +0200
Subject: [Freeswitch-users] Problem with FIFO music and chime
In-Reply-To: <CAKbxfG-GaiSSEzqAZoXADQzR10Ui_jqVw=1LwCmqkUgMjCHEyg@mail.gmail.com>
References: <CAFv7ub1-7K0v5Fe7kLeSv_1NF1GW3m9zv5+bPJSBm6Q0G4Mm0Q@mail.gmail.com>
	<CAKbxfG-GaiSSEzqAZoXADQzR10Ui_jqVw=1LwCmqkUgMjCHEyg@mail.gmail.com>
Message-ID: <CAFv7ub2vMQF3dW4m6UPP9hN3oK927v66HGOSUDWw25ovfFdbYA@mail.gmail.com>

Hello Anthony.

Before a moment I recompiled FS to .version git-cd31633 2011-08-17
19-34-22 -0500 . It's not working. Shall I send my full configuration
to you ?

--
pozdrawiam - Tomasz Hyziak



2011/8/17 Anthony Minessale <anthony.minessale at gmail.com>:
> Can you update again and confirm that is still the case. ?We tested it
> and it seems to work.
>
>
> On Wed, Aug 17, 2011 at 3:40 AM, Tomasz Hyziak <tomasz at hyziak.pl> wrote:
>> Hi
>>
>> I created simple IVR. When caller presses 2, he is redirected to FIFO.
>> But i've got a problem with fifo_music and fifo_chime_* variables...
>>
>> When I set only fifo_music - music plays.
>> When I set fifo_music and fifo_chime_freq (set to 10) and
>> fifo_chime_list - music play ONLY. There are no chime every 10
>> seconds.
>> When I set only fifo_chime_* variables (without fifo_music) - chime
>> plays every 10 seconds...
>>
>> I've got no idea why it happend - it was working about 2 weeks ago...
>>
>> I use FreeSwitch from git (downloaded @ 4 July).
>>
>> Dialplan:
>>
>> ?<extension name="in_dialog_1">
>> ? ?<condition field="destination_number" expression="^(hidden)$">
>> ? ? ?<action application="set" data="RECORD_STEREO=true"/>
>> ? ? ?<action application="record_session"
>> data="/srv/nagrania/aktualne/IN_${strftime(%Y%m%d-%H%M%S)}_${destination_number}_${caller_id_number}.wav"/>
>>
>> ? ? ?<action application="set" data="media_bug_answer_req=true"/>
>> ? ? ?<action application="set" data="domain_name=$${domain}"/>
>> ? ? ?<action application="export" data="absolute_codec_string=PCMU,PCMA"/>
>> ? ? ?<action application="export" data="codec_string=PCMU,PCMA"/>
>> ? ? ?<action application="set" data="jitterbuffer_msec=180"/>
>> ? ? ?<action application="set" data="hangup_after_bridge=true"/>
>>
>> ? ? ?<action application="answer"/>
>> ? ? ?<action application="sleep" data="2000"/>
>>
>> ? ? ?<action application="set"
>> data="fifo_chime_list=/srv/nagrania/ivr/prosze_czekac.wav"/>
>> ? ? ?<action application="set" data="fifo_chime_freq=10"/>
>> ? ? ?<action application="set"
>> data="fifo_music=/usr/local/freeswitch/sounds/music/8000/ponce-preludio-in-e-major.wav"/>
>>
>> ? ? ?<action application="ivr" data="ivr"/>
>>
>> ? ? ?<action application="fifo" data="fifo_ogolne in"/>
>>
>> ? ? ?<action application="hangup"/>
>> ? ?</condition>
>> ?</extension>
>>
>>
>> IVR:
>>
>> <include>
>> ?<menu name="ivr"
>> ? ? ?greet-long="/srv/nagrania/ivr/powitanie_pelne.wav"
>> ? ? ?greet-short="/srv/nagrania/ivr/powitanie_skrocone.wav"
>> ? ? ?invalid-sound="/srv/nagrania/ivr/zla_opcja.wav"
>> ? ? ?exit-sound="/srv/nagrania/ivr/laczenie_z_operatorem.wav"
>> ? ? ?confirm-macro="" confirm-key="" tts-engine="flite"
>> tts-voice="rms" confirm-attempts="2"
>> ? ? ?timeout="5000" inter-digit-timeout="2000" max-failures="2"
>> max-timeouts="2" digit-len="1">
>>
>> ? ?<entry action="menu-exec-app" digits="1" param="bridge user/1100"/>
>> ? ?<entry action="menu-exec-app" digits="2" param="fifo fifo_ogolne in"/>
>> ? ?<entry action="menu-top" digits="9"/>
>> ?</menu>
>>
>> </include>
>>
>>
>> FIFO:
>>
>> <configuration name="fifo.conf" description="FIFO Configuration">
>> ?<settings>
>> ? ?<param name="delete-all-outbound-member-on-startup" value="false"/>
>> ?</settings>
>> ?<fifos>
>> ? ?<fifo name="fifo_ogolne" importance="0">
>> ? ? ?<member timeout="10" simo="1"
>> lag="2">{fifo_member_wait=nowait}user/1110</member>
>> ? ? ?<member timeout="10" simo="1"
>> lag="2">{fifo_member_wait=nowait}user/1111</member>
>> ? ? ?<member timeout="10" simo="1"
>> lag="2">{fifo_member_wait=nowait}user/1112</member>
>> ? ? ?...other users...
>> ? ?</fifo>
>> ?</fifos>
>> </configuration>
>>
>>
>> --
>> Greetings - Tomasz Hyziak
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From nagalenoj at gmail.com  Thu Aug 18 16:09:11 2011
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Thu, 18 Aug 2011 17:39:11 +0530
Subject: [Freeswitch-users] DTMF issue when using execute_extension with
	play_and_get_digits
In-Reply-To: <CAKzWOxX9CEbvti8pQCRBgSipyW4y-Lx43RDQ0QTVoS2QwqbK9g@mail.gmail.com>
References: <CAA1Qw2sLHtu8ZHQn7GJFOs=f2k9H2pLa+mgu6Kt+FEtGcwWBOw@mail.gmail.com>
	<CAKzWOxX9CEbvti8pQCRBgSipyW4y-Lx43RDQ0QTVoS2QwqbK9g@mail.gmail.com>
Message-ID: <CAA1Qw2vOrZupmM7Amh9Co3esZe51_qRTO_5ir0A-Sr152C5cmA@mail.gmail.com>

I couldn't find why it's not working. I don't know, whether this is problem
specific to freetdm??

I've gone ahead and did the below experimentation after posting the
question. I tried bridging with a SIP extension instead of a mobile number,
and it works well this time. Extension 4567 is able to collect the DTMFs
given.

<action application="bridge" data="user/1010"/>

So, what is going wrong when I bridge with freetdm?

On Mon, Aug 15, 2011 at 9:32 PM, Michael Collins <msc at freeswitch.org> wrote:

> Were you able to resolve this yet?
> -MC
>
> On Tue, Aug 9, 2011 at 10:07 AM, Nagalenoj H. <nagalenoj at gmail.com> wrote:
>
>> Hi Friends,
>>      Facing an issue when using bind_meta_app and execute_extension(with
>> play_and_get_digits) combined.
>>
>> Here is my dialplan,
>> <condition field="destination_number" expression="^4567$" break="on-true">
>> <action application="play_and_get_digits" data="1 5 3 7000 #
>> '/voice_files/A.wav' '/voice_files/failed.wav' dtmf_received \d+"/>
>> </condition>
>> <condition field="destination_number" expression="^.*$" break="on-true">
>> <action application="answer"/>
>> <action application="bind_meta_app" data="5 b o execute_extension::4567
>> XML default"/>
>> <action application="bridge" data="freetdm/1/a/9952248266"/>
>> <action application="park"/>
>> </condition>
>>
>> So, when callee enters *5,  I want the caller to enter a number. I get the
>> extension executed as expected. The caller is able to hear the voice file
>> played and when he enters the digits, it is not received. Digits are not
>> even present in FS log.
>>
>> In the normal cases, there is no issues in getting DTMFs. I don't know,
>> what am I doing wrong here. Kindly, help me to resolve this.
>>
>> --
>> Regards,
>> Nagalenoj H.
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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,
Nagalenoj H.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110818/847bb07f/attachment.html 

From Stefan.Weigel at allianz-warranty.com  Thu Aug 18 16:32:50 2011
From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan)
Date: Thu, 18 Aug 2011 14:32:50 +0200
Subject: [Freeswitch-users] Original Caller ID/number after attended transfer
Message-ID: <5003D7D3E06F514E8C682F18D223265C04D3B36D6D@AZWSMS03.azwarranty.int>

A non-text attachment was scrubbed...
Name: smime.p7m
Type: application/x-pkcs7-mime
Size: 12561 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110818/17d95fa4/attachment-0001.bin 

From Stefan.Weigel at allianz-warranty.com  Thu Aug 18 16:36:17 2011
From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan)
Date: Thu, 18 Aug 2011 14:36:17 +0200
Subject: [Freeswitch-users] Original Caller ID/number after attended transfer
Message-ID: <5003D7D3E06F514E8C682F18D223265C04D3B36D6E@AZWSMS03.azwarranty.int>

Hi all,



is there a possibility to display the original caller ID & number after doing a attended transfer. External call to phone A -> calls phone B (I see caller ID & number of phone A) -> doing an attended transfer of external call to phone B (still caller ID & number of phone A).





Thanks in advance and best regards,



Stefan





Stefan Weigel
System Specialist AITP



Allianz Automotive Services GmbH

Einsteinring 28
85609 Aschheim
Germany

Tel.: +49 89 2000 48 975
Fax: +49 89 2000 48 566
eMail: Stefan.Weigel at allianz-warranty.com<mailto:Stefan.Weigel at allianz-warranty.com>


http://www.allianz-warranty.com
Gesch?ftsf?hrung: Andreas R?sing, Horst Ziegler
Amtsgericht M?nchen, HRB 175682
F?r Umsatzsteuerzwecke: Ust-ID-Nr.: DE 262 617 720



-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110818/e9be4c6d/attachment.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.gif
Type: image/gif
Size: 4415 bytes
Desc: image001.gif
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110818/e9be4c6d/attachment.gif 

From jakub.ouhrabka at gmail.com  Thu Aug 18 16:53:51 2011
From: jakub.ouhrabka at gmail.com (Jakub Ouhrabka)
Date: Thu, 18 Aug 2011 14:53:51 +0200
Subject: [Freeswitch-users] Freetdm bridge to SIP DTMF problem
In-Reply-To: <4E4CEEB1.3040202@gmail.com>
References: <4E4CD94C.3030403@comgate.cz> <4E4CEEB1.3040202@gmail.com>
Message-ID: <4E4D0B5F.2060004@gmail.com>

Hi,

I've tried further: both dtmf-type info and rfc2833 are not working 
(can't see it in packet dump). Only dtmf-type set to "none" together 
with start_dtmf_generate is working as expected for me - dtmf is 
generated inband.

How to force freeswitch to forward dtmf from freetdm to sofia preferably 
using rfc2833?

Thank you for any hints,

Jakub

Dne 18.8.2011 12:51, Jakub Ouhrabka napsal(a):
> Hi,
>
> we've changed OpenZAP to FreeTDM sometime ago and from that time we're
> experiencing problems with DTMF.
>
> We're bridging incoming ISDN30 calls from Sangoma A108DE card + libri +
> FreeTDM to SIP outgoing calls. Freeswitch is recognizing DTMF correctly
> but DTMF is not send to outgoing SIP call - it's not present in RTP
> stream trace (we've setup RFC2833 DTMF signalling). When bridging the
> call back to ISDN30 DTMF works correctly.
>
> We're using latest versions of all software as of 1 month ago.
>
> Below are attached two snippets of logs: first is bridging back to
> ISDN30 where DTMF is sent to called party. Second is bridging to SIP
> where DTMF is not sent to the called party.
>
> We've tried it with both <action application="start_dtmf" /> on and off
> with no success.
>
> Any pointers how to investigate the issue?
>
> Thanks,
>
> Jakub
>
> Log snippets:
>
> ISDN30 to ISDN30 bridge - DTMF ok
>
> 2011-08-18 12:31:45.950158 [DEBUG] ftmod_wanpipe.c:1545 [s1c11][1:11]
> read wanpipe event 3
> 2011-08-18 12:31:45.950158 [DEBUG] ftmod_wanpipe.c:1415 [s1c11][1:11]
> Queuing wanpipe DTMF: 5
> 2011-08-18 12:31:45.950158 [DEBUG] ftdm_io.c:3504 [s1c11][1:11] Queuing
> DTMF 5 (debug = 0)
> 2011-08-18 12:31:45.950158 [DEBUG] mod_freetdm.c:733 Queuing DTMF [5] in
> channel FreeTDM/1:11/XXXXXXXXX
> 2011-08-18 12:31:45.970623 [DEBUG] switch_ivr_bridge.c:391 Send signal
> FreeTDM/1:5/XXXXXXXXX [BREAK]
> 2011-08-18 12:31:45.991084 [DEBUG] ftdm_io.c:3694 [s1c5][1:5] Generating
> DTMF [5]
>
> ISDN30 to SIP bridge - DTMF not forwarded
>
> 2011-08-18 12:34:05.287480 [DEBUG] ftmod_wanpipe.c:1545 [s1c29][1:29]
> read wanpipe event 3
> 2011-08-18 12:34:05.607708 [DEBUG] ftmod_wanpipe.c:1545 [s1c29][1:29]
> read wanpipe event 3
> 2011-08-18 12:34:05.607708 [DEBUG] ftmod_wanpipe.c:1415 [s1c29][1:29]
> Queuing wanpipe DTMF: 6
> 2011-08-18 12:34:05.607708 [DEBUG] ftdm_io.c:3504 [s1c29][1:29] Queuing
> DTMF 6 (debug = 0)
> 2011-08-18 12:34:05.607708 [DEBUG] ftmod_libpri.c:1586 -- Caught Event
> span 1 18 (KEYPAD_DIGIT)
> 2011-08-18 12:34:05.628171 [DEBUG] mod_freetdm.c:733 Queuing DTMF [6] in
> channel FreeTDM/1:29/XXXXXXXXXXXXXXX
> 2011-08-18 12:34:05.628171 [DEBUG] switch_ivr_bridge.c:391 Send signal
> sofia/external/XXXXXXX at XXXXXXX:5060 [BREAK]
>


From peter.olsson at visionutveckling.se  Thu Aug 18 17:27:05 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 18 Aug 2011 15:27:05 +0200
Subject: [Freeswitch-users] Great ClueCon!
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59EBCDEE6A@cooper>

Hi all,

I just wanted to thank everyone involved in the ClueCon conference. It was the first time for me, but definately not the last one :) It was very well organized, and it was really great to meet some of the people I usually just "talk" to on the mailing list and Jira etc.

/Peter
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110818/2f359f19/attachment-0001.html 

From roger.castaldo at gmail.com  Thu Aug 18 17:34:50 2011
From: roger.castaldo at gmail.com (Roger Castaldo)
Date: Thu, 18 Aug 2011 09:34:50 -0400
Subject: [Freeswitch-users] having problems building mono 2.10
In-Reply-To: <24703.1313653049@ccs.covici.com>
References: <24703.1313653049@ccs.covici.com>
Message-ID: <CAAYf2LvLPw-s+09z13OCyrxVXq_hU9BQon4fvr_XPYNwvBPJRA@mail.gmail.com>

I think you are looking for help in the wrong list, you might want to check
with the mono development teams mailing list.

On Thu, Aug 18, 2011 at 3:37 AM, <covici at ccs.covici.com> wrote:

> Hi.  I am using gentoo, so I have to build mono by compiling and when I
> try to do this I get the following error:
>
> make[8]: Entering directory
>
> `/var/tmp/portage/dev-lang/mono-2.10.2-r1/work/mono-2.10.2/mcs/tools/gacutil'
> MCS     [basic] gacutil.exe
> Inconsistency detected by ld.so: dl-deps.c: 622: _dl_map_object_deps:
> Assertion `nlist > 1' failed!
> make[8]: *** [../../class/lib/basic/gacutil.exe] Error 127
>
> Any assistance on this would be appreciated.
>
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>         John Covici
>         covici at ccs.covici.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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110818/d7e6332c/attachment.html 

From yungwei at resolvity.com  Thu Aug 18 17:53:20 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Thu, 18 Aug 2011 09:53:20 -0400
Subject: [Freeswitch-users] voicemail doesn't work
In-Reply-To: <CAKzWOxX9rrQ_PJBBhhT+vc4Es4yq-U=q1rB7Zm43kFQHxsOreA@mail.gmail.com>
References: <33095823FD21DF429B481B5163264B7950FF12FD15@VMBX102.ihostexchange.net>
	<CAHKqMb6=znNGaG0pDOj_AbaZM-P1=JOJ2hCH3PDZ0Pqwm5CUEg@mail.gmail.com>
	<33095823FD21DF429B481B5163264B7950FF130083@VMBX102.ihostexchange.net>
	<CAKzWOxX9rrQ_PJBBhhT+vc4Es4yq-U=q1rB7Zm43kFQHxsOreA@mail.gmail.com>
Message-ID: <33095823FD21DF429B481B5163264B7950FF130274@VMBX102.ihostexchange.net>

Here's the logs for the voicemail that works: http://pastebin.freeswitch.org/17079
Here's the logs for the other case: http://pastebin.freeswitch.org/17080

Thanks.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Wednesday, August 17, 2011 3:38 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] voicemail doesn't work

Use pastebin.freeswitch.org<http://pastebin.freeswitch.org> and put the console debug output there. Capture the traffic for both the working and non-working dialplans. Hopefully there will be an error or warning that gives a clue as to what is happening. Hint: use "FreeSWITCH Log" as the syntax highlighting and it will be much easier to read.

-MC
On Wed, Aug 17, 2011 at 8:55 AM, Yungwei Chen  wrote:
If I change my dialplan to the following, voicemail will work properly. What am I missing here? Thanks.
   <extension name="main_number_2" continue="true">
     <condition field="destination_number" expression="^1112223333$">
<!--
       <action application="set" bluebox="settingEndBridge" data="hangup_after_bridge=true"/>
       <action application="set" bluebox="settingFail" data="continue_on_fail=true"/>
       <action application="set" bluebox="autoPlay" data="vm_auto_play=false"/>
       <action application="set" bluebox="settingTimeout" data="call_timeout=30"/>
       <action application="set" bluebox="settingRing" data="ringback=${us-ring}"/>
       <action application="set" bluebox="settingRingXfer" data="transfer_ringback=${us-ring}"/>
       <action application="export" bluebox="sipCalleeIdName" data="sip_callee_id_name=m1"/>
       <action application="export" bluebox="sipCalleeIdNumber" data="sip_callee_id_number=1112223333"/>
-->
       <action application="answer"/>
       <action application="voicemail" data="default voicemail_2 91000"/>
       <action application="hangup"/>
     </condition>
   </extension>

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Nandy Dagondon
Sent: Monday, August 15, 2011 6:52 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] voicemail is not saved

check the directory/file permissions
-nandy
On Tue, Aug 16, 2011 at 3:33 AM, Yungwei Chen wrote:
Hi,

I left several voicemails (Each is longer than 3 sec) to a user account, but none is available when I check the mailbox.
Relevant settings are listed below. What am I missing here? Thanks.

In conf/autoload_configs/modules.conf.xml, mod_voicemail is already loaded.
freeswitch at internal> load mod_voicemail
+OK Reloading XML
-ERR [Module already loaded]
freeswitch at internal> 2011-08-15 14:32:10.666978 [WARNING] switch_loadable_module.c:998 Module mod_voicemail Already Loaded!

Here's the content of conf/autoload_configs/voicemail.conf.xml:
<configuration name="voicemail.conf" description="Voicemail">
 <settings>
 </settings>
 <profiles>
   <profile name="default">
     <param name="file-extension" value="wav"/>
     <param name="terminator-key" value="#"/>
     <param name="max-login-attempts" value="3"/>
     <param name="digit-timeout" value="10000"/>
     <param name="min-record-len" value="3"/>
     <param name="max-record-len" value="300"/>
     <param name="max-retries" value="3"/>
     <param name="tone-spec" value="%(1000, 0, 640)"/>
     <param name="callback-dialplan" value="XML"/>
     <param name="callback-context" value="default"/>
     <param name="play-new-messages-key" value="1"/>
     <param name="play-saved-messages-key" value="2"/>
     <param name="login-keys" value="0"/>
     <param name="main-menu-key" value="0"/>
     <param name="config-menu-key" value="5"/>
     <param name="record-greeting-key" value="1"/>
     <param name="choose-greeting-key" value="2"/>
     <param name="change-pass-key" value="6"/>
     <param name="record-name-key" value="3"/>
     <param name="record-file-key" value="3"/>
     <param name="listen-file-key" value="1"/>
     <param name="save-file-key" value="2"/>
     <param name="delete-file-key" value="7"/>
     <param name="undelete-file-key" value="8"/>
     <param name="email-key" value="4"/>
     <param name="pause-key" value="0"/>
     <param name="restart-key" value="1"/>
     <param name="ff-key" value="6"/>
     <param name="rew-key" value="4"/>
     <param name="skip-greet-key" value="#"/>
     <param name="previous-message-key" value="1"/>
     <param name="next-message-key" value="3"/>
     <param name="skip-info-key" value="*"/>
     <param name="repeat-message-key" value="0"/>
     <param name="record-silence-threshold" value="200"/>
     <param name="record-silence-hits" value="2"/>
     <param name="web-template-file" value="web-vm.tpl"/>
     <!-- if you need to change the sample rate of the recorded files e.g. gmail voicemail player -->
     <!--<param name="record-sample-rate" value="11025"/>-->
     <!-- the next two both must be set for this to be enabled
          the extension is in the format of <dest> [<dialplan>] [<context>]
      -->
     <param name="operator-extension" value="operator XML default"/>
     <param name="operator-key" value="9"/>
     <param name="vmain-extension" value="vmain XML default"/>
     <param name="vmain-key" value="*"/>
     <!-- playback created files as soon as they were recorded by default -->
     <!--<param name="auto-playback-recordings" value="true"/>-->
     <email>
       <param name="template-file" value="voicemail.tpl"/>
       <param name="notify-template-file" value="notify-voicemail.tpl"/>
       <!-- this is the format voicemail_time will have -->
       <param name="date-fmt" value="%A, %B %d %Y, %I %M %p"/>
       <param name="email-from" value="${voicemail_account}@${voicemail_domain}"/>
     </email>
     <!--<param name="storage-dir" value="/tmp"/>-->
     <!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
     <!--<param name="record-comment" value="Your Comment"/>-->
     <!--<param name="record-title" value="Your Title"/>-->
     <!--<param name="record-copyright" value="Your Copyright"/>-->
   </profile>
 </profiles>
</configuration>

In conf/directory/default.xml, user 91000 is defined in domain voicemail_2.
 <domain name="voicemail_2">
   <groups>
     <group name="default">
       <users>
         <user bluebox_id="2" id="91000">
           <params>
             <param name="sip-forbid-register" value="true"/>
             <param name="vm-password" value="xxxx"/>
             <param name="vm-message-ext" value="wav"/>
             <param name="vm-mailto" value="xxx at xxx.com<mailto:xxx at xxx.com>"/>
             <param name="email-addr" value="xxx at xxx.com<mailto:xxx at xxx.com>"/>
             <param name="vm-delete-file" value="false"/>
             <param name="vm-attach-file" value="true"/>
             <param name="vm-email-all-messages" value="true"/>
           </params>
           <variables>
             <variable name="timezone" value="America/Chicago"/>
           </variables>
         </user>
       </users>
     </group>
   </groups>
 </domain>

In my dialplan, calls to 1112223333 will be sent to user 91000's voicemail box if they are not answered.
   <extension name="main_number_2" continue="true">
     <condition field="destination_number" expression="^1112223333$">
       <action application="set" bluebox="settingEndBridge" data="hangup_after_bridge=true"/>
       <action application="set" bluebox="settingFail" data="continue_on_fail=true"/>
       <action application="set" bluebox="autoPlay" data="vm_auto_play=false"/>
       <action application="set" bluebox="settingTimeout" data="call_timeout=30"/>
       <action application="set" bluebox="settingRing" data="ringback=${us-ring}"/>
       <action application="set" bluebox="settingRingXfer" data="transfer_ringback=${us-ring}"/>
       <action application="export" bluebox="sipCalleeIdName" data="sip_callee_id_name=m1"/>
       <action application="export" bluebox="sipCalleeIdNumber" data="sip_callee_id_number=1112223333"/>
       <action application="bridge" data="user/91000@$${location_3}"/>
       <action application="answer"/>
       <action application="voicemail" data="default voicemail_2 91000"/>
       <action application="hangup"/>
     </condition>
   </extension>


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



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

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110818/2df54649/attachment-0001.html 

From covici at ccs.covici.com  Thu Aug 18 18:04:16 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 18 Aug 2011 10:04:16 -0400
Subject: [Freeswitch-users] having problems building mono 2.10
In-Reply-To: <CAAYf2LvLPw-s+09z13OCyrxVXq_hU9BQon4fvr_XPYNwvBPJRA@mail.gmail.com>
References: <24703.1313653049@ccs.covici.com>
	<CAAYf2LvLPw-s+09z13OCyrxVXq_hU9BQon4fvr_XPYNwvBPJRA@mail.gmail.com>
Message-ID: <6595.1313676256@ccs.covici.com>

I will try that, but in case someone here had built it, I was hoping.

Thanks.

Roger Castaldo <roger.castaldo at gmail.com> wrote:

> I think you are looking for help in the wrong list, you might want to check
> with the mono development teams mailing list.
> 
> On Thu, Aug 18, 2011 at 3:37 AM, <covici at ccs.covici.com> wrote:
> 
> > Hi.  I am using gentoo, so I have to build mono by compiling and when I
> > try to do this I get the following error:
> >
> > make[8]: Entering directory
> >
> > `/var/tmp/portage/dev-lang/mono-2.10.2-r1/work/mono-2.10.2/mcs/tools/gacutil'
> > MCS     [basic] gacutil.exe
> > Inconsistency detected by ld.so: dl-deps.c: 622: _dl_map_object_deps:
> > Assertion `nlist > 1' failed!
> > make[8]: *** [../../class/lib/basic/gacutil.exe] Error 127
> >
> > Any assistance on this would be appreciated.
> >
> > --
> > Your life is like a penny.  You're going to lose it.  The question is:
> > How do
> > you spend it?
> >
> >         John Covici
> >         covici at ccs.covici.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
> >
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From erin.omeara at salmonbaytechnology.com  Thu Aug 18 18:35:56 2011
From: erin.omeara at salmonbaytechnology.com (Erin O'Meara)
Date: Thu, 18 Aug 2011 07:35:56 -0700
Subject: [Freeswitch-users] Audio Problem - one way
Message-ID: <CAB4Oy0XDz5xcBfVbpzu_gxHhakZ2-+j8OdCgHG5GScqPppZ50A@mail.gmail.com>

I have to VOIP servers in the cloud(same provider), one is FusionPBX running
Freeswitch the other is Elastix with Asterisk.  I have two phones(same
Grandstream GXP280) in my office each registered to one of the servers.  I
have one phone number and when I route it thru the Elastix server, I get no
complaints about the audio. When I route the number thru the FusionPBX the
person on the other end complains of choppy audio but I can't tell
the difference on my end. But when I call from my cell I can tell that the
Freeswitch server audio is not as good on the outgoing.

Both Virtual servers are the same specs, memory and proc. The elastix box
has more overhead with all the extra crap it runs and I can look at
the performance metric's on the virtual servers and confirm that the
fusionpbx has less proc usage and memory usage. The calls are using the same
codec (ulaw). Currently the call volume is just my one call for the Fusion
server, and at most 2 for the Elastix.

Any thoughts on with the same setup and internet routes, audio in one
direction is preforming so poorly with the freeswitch?


Regards,

206.905.9520
http://salmonbaytechnology.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110818/2d163311/attachment.html 

From stkn at freeswitch.org  Thu Aug 18 18:45:56 2011
From: stkn at freeswitch.org (Stefan Knoblich)
Date: Thu, 18 Aug 2011 16:45:56 +0200
Subject: [Freeswitch-users] having problems building mono 2.10
In-Reply-To: <24703.1313653049@ccs.covici.com>
References: <24703.1313653049@ccs.covici.com>
Message-ID: <1656439.BN3H24Bjmd@tsukasa>

On Thursday 18 August 2011 03:37:29 covici at ccs.covici.com wrote:
> Hi.  I am using gentoo, so I have to build mono by compiling and when I
> try to do this I get the following error:
> 
> make[8]: Entering directory
> `/var/tmp/portage/dev-lang/mono-2.10.2-r1/work/mono-2.10.2/mcs/tools/gacutil'
> MCS     [basic] gacutil.exe
> Inconsistency detected by ld.so: dl-deps.c: 622: _dl_map_object_deps:
> Assertion `nlist > 1' failed!
> make[8]: *** [../../class/lib/basic/gacutil.exe] Error 127
> 
> Any assistance on this would be appreciated.


not a mono bug:  https://bugs.gentoo.org/show_bug.cgi?id=374107

-- 
-------------------------------------------------------------------------------
Stefan Knoblich                        | Web:   http://www.axsentis.de/
axsentis GmbH                          |        http://oss.axsentis.de/
Eupener Str. 74, 50933 Koeln, Germany  |
Amtsgericht Koeln: HR B 56238          | Email: s.knoblich at axsentis.de
UST-ID: DE244977565                    | JID:   s.knoblich at jabber.axsentis.de
-------------------------------------------------------------------------------
Web:   http://stkn.techmage.de/
Email: stkn at freeswitch.org
IRC:   #freeswitch-de @ irc.freenode.net
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: application/pgp-signature
Size: 198 bytes
Desc: This is a digitally signed message part.
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110818/676ff76c/attachment.bin 

From dujinfang at gmail.com  Thu Aug 18 18:52:58 2011
From: dujinfang at gmail.com (Seven Du)
Date: Thu, 18 Aug 2011 22:52:58 +0800
Subject: [Freeswitch-users] strdup error on solaris
Message-ID: <7E3EF9F827D54F628C130D0AC650BAD0@gmail.com>

Hi,

I'm new to solaris

SunOS solaris 5.11 snv_134 i86pc i386 i86pc

I didn't follow the wiki about installing on solaris but installed building tool chain with pkg install SUNWgcc etc. 
bootstrap and configure was ok, however, I got error on gmake. Any highlight on this? Thanks.

libs/stfu/stfu.c: In function `stfu_n_debug':
libs/stfu/stfu.c:224: warning: implicit declaration of function `strdup'
libs/stfu/stfu.c:224: warning: assignment makes pointer from integer without a cast
libs/stfu/stfu.c:227: warning: assignment makes pointer from integer without a cast
libs/stfu/stfu.c: In function `stfu_n_init':
libs/stfu/stfu.c:302: warning: assignment makes pointer from integer without a cast
gmake[1]: *** [libfreeswitch_la-stfu.lo] #### 1
gmake: *** [all] #### 2
link at solaris:~/seven/freeswitch# vi libs/stfu/stfu.c:224
link at solaris:~/seven/freeswitch# vi libs/stfu/stfu.c


link at solaris:~/seven/freeswitch# gcc -v
Reading specs from /usr/sfw/lib/gcc/i386-pc-solaris2.11/3.4.3/specs
Configured with: /builds2/sfwnv-gate/usr/src/cmd/gcc/gcc-3.4.3/configure --prefix=/usr/sfw --with-as=/usr/sfw/bin/gas --with-gnu-as --with-ld=/usr/ccs/bin/ld --without-gnu-ld --enable-languages=c,c++,f77,objc --enable-shared
Thread model: posix
gcc version 3.4.3 (csl-sol210-3_4-20050802)

-- 
Seven Du
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj: http://www.freeswitch.org.cn
Sent with Sparrow (http://www.sparrowmailapp.com)
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110818/1a018e0a/attachment.html 

From anthony.minessale at gmail.com  Thu Aug 18 18:56:06 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 18 Aug 2011 09:56:06 -0500
Subject: [Freeswitch-users] strdup error on solaris
In-Reply-To: <7E3EF9F827D54F628C130D0AC650BAD0@gmail.com>
References: <7E3EF9F827D54F628C130D0AC650BAD0@gmail.com>
Message-ID: <CAKbxfG8BUmhTfr0GEoRrb=g1LcaeRLasmf+Mxt24xgPaG6Eiag@mail.gmail.com>

hmm, did solaris stop putting strdup in string.h ?


On Thu, Aug 18, 2011 at 9:52 AM, Seven Du <dujinfang at gmail.com> wrote:
> Hi,
> I'm new to solaris
> SunOS solaris 5.11 snv_134 i86pc i386 i86pc
> I didn't follow the wiki about installing on solaris but installed building
> tool chain with pkg install SUNWgcc etc.
> bootstrap and configure was ok, however, I got error on gmake. Any highlight
> on this? ?Thanks.
> libs/stfu/stfu.c: In function `stfu_n_debug':
> libs/stfu/stfu.c:224: warning: implicit declaration of function `strdup'
> libs/stfu/stfu.c:224: warning: assignment makes pointer from integer without
> a cast
> libs/stfu/stfu.c:227: warning: assignment makes pointer from integer without
> a cast
> libs/stfu/stfu.c: In function `stfu_n_init':
> libs/stfu/stfu.c:302: warning: assignment makes pointer from integer without
> a cast
> gmake[1]: *** [libfreeswitch_la-stfu.lo] #### 1
> gmake: *** [all] #### 2
> link at solaris:~/seven/freeswitch# vi libs/stfu/stfu.c:224
> link at solaris:~/seven/freeswitch# vi libs/stfu/stfu.c
>
> link at solaris:~/seven/freeswitch# gcc -v
> Reading specs from /usr/sfw/lib/gcc/i386-pc-solaris2.11/3.4.3/specs
> Configured with: /builds2/sfwnv-gate/usr/src/cmd/gcc/gcc-3.4.3/configure
> --prefix=/usr/sfw --with-as=/usr/sfw/bin/gas --with-gnu-as
> --with-ld=/usr/ccs/bin/ld --without-gnu-ld --enable-languages=c,c++,f77,objc
> --enable-shared
> Thread model: posix
> gcc version 3.4.3 (csl-sol210-3_4-20050802)
> --
> Seven Du
> About: http://about.me/dujinfang
> Blog: http://www.dujinfang.com
> Proj: ?http://www.freeswitch.org.cn
> Sent with Sparrow
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From covici at ccs.covici.com  Thu Aug 18 18:56:33 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 18 Aug 2011 10:56:33 -0400
Subject: [Freeswitch-users] having problems building mono 2.10
In-Reply-To: <1656439.BN3H24Bjmd@tsukasa>
References: <24703.1313653049@ccs.covici.com> <1656439.BN3H24Bjmd@tsukasa>
Message-ID: <13751.1313679393@ccs.covici.com>

Thanks -- my google search did not find this at all.

Stefan Knoblich <stkn at freeswitch.org> wrote:

> On Thursday 18 August 2011 03:37:29 covici at ccs.covici.com wrote:
> > Hi.  I am using gentoo, so I have to build mono by compiling and when I
> > try to do this I get the following error:
> > 
> > make[8]: Entering directory
> > `/var/tmp/portage/dev-lang/mono-2.10.2-r1/work/mono-2.10.2/mcs/tools/gacutil'
> > MCS     [basic] gacutil.exe
> > Inconsistency detected by ld.so: dl-deps.c: 622: _dl_map_object_deps:
> > Assertion `nlist > 1' failed!
> > make[8]: *** [../../class/lib/basic/gacutil.exe] Error 127
> > 
> > Any assistance on this would be appreciated.
> 
> 
> not a mono bug:  https://bugs.gentoo.org/show_bug.cgi?id=374107
> 
> -- 
> -------------------------------------------------------------------------------
> Stefan Knoblich                        | Web:   http://www.axsentis.de/
> axsentis GmbH                          |        http://oss.axsentis.de/
> Eupener Str. 74, 50933 Koeln, Germany  |
> Amtsgericht Koeln: HR B 56238          | Email: s.knoblich at axsentis.de
> UST-ID: DE244977565                    | JID:   s.knoblich at jabber.axsentis.de
> -------------------------------------------------------------------------------
> Web:   http://stkn.techmage.de/
> Email: stkn at freeswitch.org
> IRC:   #freeswitch-de @ irc.freenode.net
-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From anthony.minessale at gmail.com  Thu Aug 18 19:04:09 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 18 Aug 2011 10:04:09 -0500
Subject: [Freeswitch-users] Problem with FIFO music and chime
In-Reply-To: <CAFv7ub2vMQF3dW4m6UPP9hN3oK927v66HGOSUDWw25ovfFdbYA@mail.gmail.com>
References: <CAFv7ub1-7K0v5Fe7kLeSv_1NF1GW3m9zv5+bPJSBm6Q0G4Mm0Q@mail.gmail.com>
	<CAKbxfG-GaiSSEzqAZoXADQzR10Ui_jqVw=1LwCmqkUgMjCHEyg@mail.gmail.com>
	<CAFv7ub2vMQF3dW4m6UPP9hN3oK927v66HGOSUDWw25ovfFdbYA@mail.gmail.com>
Message-ID: <CAKbxfG_oMtNAqzAncL6zbeDefqZP9xNVx=9C6Nu14BWNOYocZQ@mail.gmail.com>

yes please

On Thu, Aug 18, 2011 at 4:13 AM, Tomasz Hyziak <tomasz at hyziak.pl> wrote:
> Hello Anthony.
>
> Before a moment I recompiled FS to .version git-cd31633 2011-08-17
> 19-34-22 -0500 . It's not working. Shall I send my full configuration
> to you ?
>
> --
> pozdrawiam - Tomasz Hyziak
>
>
>
> 2011/8/17 Anthony Minessale <anthony.minessale at gmail.com>:
>> Can you update again and confirm that is still the case. ?We tested it
>> and it seems to work.
>>
>>
>> On Wed, Aug 17, 2011 at 3:40 AM, Tomasz Hyziak <tomasz at hyziak.pl> wrote:
>>> Hi
>>>
>>> I created simple IVR. When caller presses 2, he is redirected to FIFO.
>>> But i've got a problem with fifo_music and fifo_chime_* variables...
>>>
>>> When I set only fifo_music - music plays.
>>> When I set fifo_music and fifo_chime_freq (set to 10) and
>>> fifo_chime_list - music play ONLY. There are no chime every 10
>>> seconds.
>>> When I set only fifo_chime_* variables (without fifo_music) - chime
>>> plays every 10 seconds...
>>>
>>> I've got no idea why it happend - it was working about 2 weeks ago...
>>>
>>> I use FreeSwitch from git (downloaded @ 4 July).
>>>
>>> Dialplan:
>>>
>>> ?<extension name="in_dialog_1">
>>> ? ?<condition field="destination_number" expression="^(hidden)$">
>>> ? ? ?<action application="set" data="RECORD_STEREO=true"/>
>>> ? ? ?<action application="record_session"
>>> data="/srv/nagrania/aktualne/IN_${strftime(%Y%m%d-%H%M%S)}_${destination_number}_${caller_id_number}.wav"/>
>>>
>>> ? ? ?<action application="set" data="media_bug_answer_req=true"/>
>>> ? ? ?<action application="set" data="domain_name=$${domain}"/>
>>> ? ? ?<action application="export" data="absolute_codec_string=PCMU,PCMA"/>
>>> ? ? ?<action application="export" data="codec_string=PCMU,PCMA"/>
>>> ? ? ?<action application="set" data="jitterbuffer_msec=180"/>
>>> ? ? ?<action application="set" data="hangup_after_bridge=true"/>
>>>
>>> ? ? ?<action application="answer"/>
>>> ? ? ?<action application="sleep" data="2000"/>
>>>
>>> ? ? ?<action application="set"
>>> data="fifo_chime_list=/srv/nagrania/ivr/prosze_czekac.wav"/>
>>> ? ? ?<action application="set" data="fifo_chime_freq=10"/>
>>> ? ? ?<action application="set"
>>> data="fifo_music=/usr/local/freeswitch/sounds/music/8000/ponce-preludio-in-e-major.wav"/>
>>>
>>> ? ? ?<action application="ivr" data="ivr"/>
>>>
>>> ? ? ?<action application="fifo" data="fifo_ogolne in"/>
>>>
>>> ? ? ?<action application="hangup"/>
>>> ? ?</condition>
>>> ?</extension>
>>>
>>>
>>> IVR:
>>>
>>> <include>
>>> ?<menu name="ivr"
>>> ? ? ?greet-long="/srv/nagrania/ivr/powitanie_pelne.wav"
>>> ? ? ?greet-short="/srv/nagrania/ivr/powitanie_skrocone.wav"
>>> ? ? ?invalid-sound="/srv/nagrania/ivr/zla_opcja.wav"
>>> ? ? ?exit-sound="/srv/nagrania/ivr/laczenie_z_operatorem.wav"
>>> ? ? ?confirm-macro="" confirm-key="" tts-engine="flite"
>>> tts-voice="rms" confirm-attempts="2"
>>> ? ? ?timeout="5000" inter-digit-timeout="2000" max-failures="2"
>>> max-timeouts="2" digit-len="1">
>>>
>>> ? ?<entry action="menu-exec-app" digits="1" param="bridge user/1100"/>
>>> ? ?<entry action="menu-exec-app" digits="2" param="fifo fifo_ogolne in"/>
>>> ? ?<entry action="menu-top" digits="9"/>
>>> ?</menu>
>>>
>>> </include>
>>>
>>>
>>> FIFO:
>>>
>>> <configuration name="fifo.conf" description="FIFO Configuration">
>>> ?<settings>
>>> ? ?<param name="delete-all-outbound-member-on-startup" value="false"/>
>>> ?</settings>
>>> ?<fifos>
>>> ? ?<fifo name="fifo_ogolne" importance="0">
>>> ? ? ?<member timeout="10" simo="1"
>>> lag="2">{fifo_member_wait=nowait}user/1110</member>
>>> ? ? ?<member timeout="10" simo="1"
>>> lag="2">{fifo_member_wait=nowait}user/1111</member>
>>> ? ? ?<member timeout="10" simo="1"
>>> lag="2">{fifo_member_wait=nowait}user/1112</member>
>>> ? ? ?...other users...
>>> ? ?</fifo>
>>> ?</fifos>
>>> </configuration>
>>>
>>>
>>> --
>>> Greetings - Tomasz Hyziak
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> 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
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From x.liu at hw.ac.uk  Thu Aug 18 19:50:44 2011
From: x.liu at hw.ac.uk (xl127)
Date: Thu, 18 Aug 2011 16:50:44 +0100
Subject: [Freeswitch-users] mod_spidermonkey loading error
In-Reply-To: <CAKzWOxVyTnNMeHvxAWviHNM5MfPFW8XQoF_M0X=0ghzLv3Zhpw@mail.gmail.com>
References: <20110817091244.5582.qmail@community37.interfree.it>	<4E4B9352.4010006@hw.ac.uk>
	<CAKzWOxVyTnNMeHvxAWviHNM5MfPFW8XQoF_M0X=0ghzLv3Zhpw@mail.gmail.com>
Message-ID: <4E4D34D4.8000401@hw.ac.uk>

It works on Fedora 14 now, thanks Michael!

I am wondering if I need to give the flag "--without-libcurl" every time 
when I add a new module to FS and do "configure"?

Cheers,

Xing


On 17/08/11 21:33, Michael Collins wrote:
> This is an odd duck. Most likely you're on a 64 bit platform. There's 
> an issue with CentOS6 and evidently Fedora14. Try re-running configure 
> like this:
>
> ./configure --without-libcurl
>
> See http://jira.freeswitch.org/browse/FS-3393 for a discussion of this 
> bug.
>
> -MC
>
> On Wed, Aug 17, 2011 at 3:09 AM, xl127 <x.liu at hw.ac.uk 
> <mailto:x.liu at hw.ac.uk>> wrote:
>
>     Hi,
>
>     I can run the FreeSwitch on CentOS 5 without any problem.
>     When I run it on Fedora 14 (tried latest git version and latest
>     snapshot
>     version), I got following error
>
>     011-08-16 18:19:52.928857 [CRIT] switch_loadable_module.c:929 Error
>     Loading module /usr/local/freeswitch/mod/mod_spidermonkey.so
>     **/usr/lib/libldap-2.4.so.2: undefined symbol:
>     PR_GetDirectorySeparator**
>
>     I'm stuck here for a while, googled around but didn't figure out a
>     solution.
>
>     Any suggestions?
>
>     Thanks,
>
>     Xing
>
>
>
>     --
>     Heriot-Watt University is a Scottish charity
>     registered under charity number SC000278.
>
>
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Heriot-Watt University is a Scottish charity
registered under charity number SC000278.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110818/e3370078/attachment.html 

From lfurrea at gmail.com  Thu Aug 18 19:50:49 2011
From: lfurrea at gmail.com (Luis F Urrea)
Date: Thu, 18 Aug 2011 09:50:49 -0600
Subject: [Freeswitch-users] Best tabletop conference phone
Message-ID: <CAB2KLydFq-UmZfbOaypj5uWoFyFHjtsLZ+po-H4P3Jja+RTPqw@mail.gmail.com>

Hi all,

After using a Snom Meeting Point for 9 months that suddenly went dead while
connected to PoE which in turn is properly connected to UPS my customer is
extremely upset and frustrated to that experience.

He is asking me for a quote on the best possible tabletop conference phone
that we can use with FS since he is not even interested in processing the
Meeting Point warranty.

Please please please,your input is appreciated!!!

Regards,

Luis
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110818/fc256fd0/attachment.html 

From john at millican.us  Thu Aug 18 20:15:42 2011
From: john at millican.us (john Millican)
Date: Thu, 18 Aug 2011 12:15:42 -0400
Subject: [Freeswitch-users] Best tabletop conference phone
In-Reply-To: <CAB2KLydFq-UmZfbOaypj5uWoFyFHjtsLZ+po-H4P3Jja+RTPqw@mail.gmail.com>
References: <CAB2KLydFq-UmZfbOaypj5uWoFyFHjtsLZ+po-H4P3Jja+RTPqw@mail.gmail.com>
Message-ID: <4E4D3AAE.2090402@millican.us>

On 8/18/2011 11:50 AM, Luis F Urrea wrote:
> Hi all,
>
> After using a Snom Meeting Point for 9 months that suddenly went dead 
> while connected to PoE which in turn is properly connected to UPS my 
> customer is extremely upset and frustrated to that experience.
>
> He is asking me for a quote on the best possible tabletop conference 
> phone that we can use with FS since he is not even interested in 
> processing the Meeting Point warranty.
>
> Please please please,your input is appreciated!!!
>
> Regards,
>
> Luis
>
>
I have had good luck with the polycom soundstation ip conference 
phones.  Good sound quality and they have options for extra mics to be 
placed around the room or table.  Easy to configure if you are at all 
familiar with the polycom config's.  I have several of these in place 
going back about 4 years.

JohnM
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110818/fd1f9221/attachment.html 

From spencer at 5ninesolutions.com  Thu Aug 18 20:29:52 2011
From: spencer at 5ninesolutions.com (Spencer Thomason)
Date: Thu, 18 Aug 2011 11:29:52 -0500
Subject: [Freeswitch-users] Best tabletop conference phone
In-Reply-To: <4E4D3AAE.2090402@millican.us>
References: <CAB2KLydFq-UmZfbOaypj5uWoFyFHjtsLZ+po-H4P3Jja+RTPqw@mail.gmail.com>
	<4E4D3AAE.2090402@millican.us>
Message-ID: <D0D2D7FF-B584-4926-810B-9DB5E34612A3@5ninesolutions.com>

+1 for the Polycoms.  They sound great and are very reliable.

Spencer


On Aug 18, 2011, at 11:15 AM, john Millican wrote:

> On 8/18/2011 11:50 AM, Luis F Urrea wrote:
>> 
>> Hi all,
>> 
>> After using a Snom Meeting Point for 9 months that suddenly went dead while connected to PoE which in turn is properly connected to UPS my customer is extremely upset and frustrated to that experience.
>> 
>> He is asking me for a quote on the best possible tabletop conference phone that we can use with FS since he is not even interested in processing the Meeting Point warranty.
>> 
>> Please please please,your input is appreciated!!!
>> 
>> Regards,
>> 
>> Luis
>> 
>> 
> I have had good luck with the polycom soundstation ip conference phones.  Good sound quality and they have options for extra mics to be placed around the room or table.  Easy to configure if you are at all familiar with the polycom config's.  I have several of these in place going back about 4 years.
> 
> JohnM
>> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From msc at freeswitch.org  Thu Aug 18 20:34:13 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 18 Aug 2011 09:34:13 -0700
Subject: [Freeswitch-users] mod_spidermonkey loading error
In-Reply-To: <4E4D34D4.8000401@hw.ac.uk>
References: <20110817091244.5582.qmail@community37.interfree.it>
	<4E4B9352.4010006@hw.ac.uk>
	<CAKzWOxVyTnNMeHvxAWviHNM5MfPFW8XQoF_M0X=0ghzLv3Zhpw@mail.gmail.com>
	<4E4D34D4.8000401@hw.ac.uk>
Message-ID: <CAKzWOxV6nM0dnar6kZThiakpv3gCpjzMAQXr6+P2WBuYncC8-Q@mail.gmail.com>

On Thu, Aug 18, 2011 at 8:50 AM, xl127 <x.liu at hw.ac.uk> wrote:

> **
> It works on Fedora 14 now, thanks Michael!
>
> I am wondering if I need to give the flag "--without-libcurl" every time
> when I add a new module to FS and do "configure"?
>

Only when you need to actually re-run the configure script. Most of the time
when you do "git pull && make install" or "make current" you're fine.

-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110818/39990ac4/attachment.html 

From anthony.minessale at gmail.com  Thu Aug 18 20:34:33 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 18 Aug 2011 11:34:33 -0500
Subject: [Freeswitch-users] Simulating Asterisk dial(x:y:z)
In-Reply-To: <CAKzWOxXXv40KF8Jf-tLDYQ9ApxkW0f+4BqNwM=J9VxRFq0Wz-g@mail.gmail.com>
References: <1313566426.80683.YahooMailNeo@web161019.mail.bf1.yahoo.com>
	<CAKzWOxVNM5pqxrD720JJ0p9VDmO1nWC8fyn0CUn36XbdWY1QKA@mail.gmail.com>
	<1313624376.73435.YahooMailNeo@web161001.mail.bf1.yahoo.com>
	<CAKzWOxXXv40KF8Jf-tLDYQ9ApxkW0f+4BqNwM=J9VxRFq0Wz-g@mail.gmail.com>
Message-ID: <CAKbxfG9QSr=_y+-AEt_NzyMbknSj_SCcZHnaL9Aaqo9JuZs2ZA@mail.gmail.com>

latest git all of the above should work including pushing many vars
into execute_on_answer/api_on_answer referenced as an array.

On Wed, Aug 17, 2011 at 7:11 PM, Michael Collins <msc at freeswitch.org> wrote:
> Hmmm,
> I wonder if the multiple syntax only made it into the execute_on_xxx vars.
> I'll ask Brian about that. In the meantime it is possible to use
> execute_on_answer and execute an extension that in turn does all these API
> calls on your channel:
> <action application="export"
> data="nolocal:execute_on_answer=execute_extension do_stuff XML
> custom_context"/>
> Then have a simple context. Add file conf/dialplan/custom.xml:
> <include>
> ? <context name="custom_context">
> ? ? <extension name="do my stuff">
> ? ? ? <condition field="destination_number"?expression="^do_stuff$">
> ? ? ? ? <action application="set" data="res1=${sched_broadcast(+1 ${uuid}
> playback::sound1.wav aleg)}"/>
> ? ? ? ? <action application="set" data="res2=${sched_hangup(+30 ${uuid})}"/>
> ? ? ? </condition>
> ? ? </extension>
> ? </context>
> </include>
> You can try that one for kicks while we research the api_on_answer question.
> -MC
>
> On Wed, Aug 17, 2011 at 4:39 PM, Sam <lakersman2006 at yahoo.com> wrote:
>>
>> MC,
>> I tried the syntax you suggested but I does not seem to be working
>> <action application="export"
>> data="nolocal:api_on_answer_1=sched_broadcast(+1 ${uuid}
>> playback::sound1.wav aleg), nolocal:api_on_answer_2=sched_hangup(+30
>> ${uuid})"/>
>>
>> I also tried this syntax and still no luck
>> <action application="export" data="api_on_answer_1=sched_broadcast +1
>> ${uuid}? playback::sound1.wav"/>
>> <action application="export" data="api_on_answer_2=sched_hangup +15
>> ${uuid}"/>
>> ________________________________
>> From: Michael Collins <msc at freeswitch.org>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Sent: Wednesday, August 17, 2011 3:05 PM
>> Subject: Re: [Freeswitch-users] Simulating Asterisk dial(x:y:z)
>>
>> You can do multiple api_on_answer calls by using a relatively new syntax:
>>
>> api_on_answer_1=api(arg),api_on_answer_2=api2(arg2),api_on_answer_3=api3(arg3)
>> Same syntax applies to the execute_on_xxx variables.
>> -MC
>>
>> On Wed, Aug 17, 2011 at 12:33 AM, Sam <lakersman2006 at yahoo.com> wrote:
>>
>> I want to simulate the Asterisk dial(x:y:z) app where (x=total call time,
>> y=warning time, z=warning interval) in Freeswitch. I know it can be done
>> with sched_hangup and sched_broadcast, my question is how can I call a
>> series of "api_on_answer" comands to do it?
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From lfurrea at gmail.com  Thu Aug 18 20:37:53 2011
From: lfurrea at gmail.com (Luis F Urrea)
Date: Thu, 18 Aug 2011 10:37:53 -0600
Subject: [Freeswitch-users] Best tabletop conference phone
In-Reply-To: <D0D2D7FF-B584-4926-810B-9DB5E34612A3@5ninesolutions.com>
References: <CAB2KLydFq-UmZfbOaypj5uWoFyFHjtsLZ+po-H4P3Jja+RTPqw@mail.gmail.com>
	<4E4D3AAE.2090402@millican.us>
	<D0D2D7FF-B584-4926-810B-9DB5E34612A3@5ninesolutions.com>
Message-ID: <CAB2KLycjbLg2Je7aAxyTnnpGmHOrk8zvbyV_uAOTkxrZE-7V-g@mail.gmail.com>

Yeah I thought so, seems that you cannot go wrong with the Polycoms.

I'll go for the Soundstation IP 7000

On Thu, Aug 18, 2011 at 10:29 AM, Spencer Thomason <
spencer at 5ninesolutions.com> wrote:

> +1 for the Polycoms.  They sound great and are very reliable.
>
> Spencer
>
>
> On Aug 18, 2011, at 11:15 AM, john Millican wrote:
>
> > On 8/18/2011 11:50 AM, Luis F Urrea wrote:
> >>
> >> Hi all,
> >>
> >> After using a Snom Meeting Point for 9 months that suddenly went dead
> while connected to PoE which in turn is properly connected to UPS my
> customer is extremely upset and frustrated to that experience.
> >>
> >> He is asking me for a quote on the best possible tabletop conference
> phone that we can use with FS since he is not even interested in processing
> the Meeting Point warranty.
> >>
> >> Please please please,your input is appreciated!!!
> >>
> >> Regards,
> >>
> >> Luis
> >>
> >>
> > I have had good luck with the polycom soundstation ip conference phones.
>  Good sound quality and they have options for extra mics to be placed around
> the room or table.  Easy to configure if you are at all familiar with the
> polycom config's.  I have several of these in place going back about 4
> years.
> >
> > JohnM
> >>
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110818/c6af27ef/attachment-0001.html 

From msc at freeswitch.org  Thu Aug 18 20:39:00 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 18 Aug 2011 09:39:00 -0700
Subject: [Freeswitch-users] Great ClueCon!
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C59EBCDEE6A@cooper>
References: <549CFEF87AEDE841A38E9D15EAB4C04C59EBCDEE6A@cooper>
Message-ID: <CAKzWOxUq3ksh_Qv90KQaxza0N+BN_tfkQBkje0rzcUVurL-W+w@mail.gmail.com>

On Thu, Aug 18, 2011 at 6:27 AM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> Hi all,****
>
> ** **
>
> I just wanted to thank everyone involved in the ClueCon conference. It was
> the first time for me, but definately not the last one :) It was very well
> organized, and it was really great to meet some of the people I usually just
> ?talk? to on the mailing list and Jira etc.****
>
> ** **
>
> /Peter****
>
>
Thank you for saying so! It was a busy time for me but I really enjoyed
seeing everyone. Every year leading up to ClueCon Brian West and I are
pulling our hair out because it's so much work, but in the end it's *always*
worth it! Thanks to everyone who came out and helped us have such a great
event.

See you all in 2012! ;)

-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110818/f8b8bfde/attachment.html 

From msc at freeswitch.org  Thu Aug 18 20:42:27 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 18 Aug 2011 09:42:27 -0700
Subject: [Freeswitch-users] Best tabletop conference phone
In-Reply-To: <CAB2KLycjbLg2Je7aAxyTnnpGmHOrk8zvbyV_uAOTkxrZE-7V-g@mail.gmail.com>
References: <CAB2KLydFq-UmZfbOaypj5uWoFyFHjtsLZ+po-H4P3Jja+RTPqw@mail.gmail.com>
	<4E4D3AAE.2090402@millican.us>
	<D0D2D7FF-B584-4926-810B-9DB5E34612A3@5ninesolutions.com>
	<CAB2KLycjbLg2Je7aAxyTnnpGmHOrk8zvbyV_uAOTkxrZE-7V-g@mail.gmail.com>
Message-ID: <CAKzWOxX+bgYYDwnEvKjhO_q9K9JiJ5O1tR9pGGJtFTVZ00vbsA@mail.gmail.com>

On Thu, Aug 18, 2011 at 9:37 AM, Luis F Urrea <lfurrea at gmail.com> wrote:

> Yeah I thought so, seems that you cannot go wrong with the Polycoms.
>
> I'll go for the Soundstation IP 7000
>

Don't forget to teach the people who use the speaker phone that if they are
in a room with solid walls that cause the sound to bounce all over the place
that the other party will hear "funny echo" and stuff like that. I can't
tell you how many times we've had to educate people on the fact that these
phones, while awesome, simply cannot magically filter out every little sound
that you don't want transmitted. :)

-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110818/49e6ed44/attachment.html 

From anthony.minessale at gmail.com  Thu Aug 18 20:45:03 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 18 Aug 2011 11:45:03 -0500
Subject: [Freeswitch-users] Original Caller ID/number after attended
	transfer
In-Reply-To: <5003D7D3E06F514E8C682F18D223265C04D3B36D6E@AZWSMS03.azwarranty.int>
References: <5003D7D3E06F514E8C682F18D223265C04D3B36D6E@AZWSMS03.azwarranty.int>
Message-ID: <CAKbxfG-6a26_NqN0UJ46sXXEbr2JHbF76kX5SUmkda7wgRWLZQ@mail.gmail.com>

if you get phones that support display updates (polycom, snom, cisco and a
few others) yes. It already works.
What phones do you have?


On Thu, Aug 18, 2011 at 7:36 AM, Weigel, Stefan <
Stefan.Weigel at allianz-warranty.com> wrote:

>  Hi all,****
>
> ** **
>
> is there a possibility to display the original caller ID & number after
> doing a attended transfer. External call to phone A -> calls phone B (I see
> caller ID & number of phone A) -> doing an attended transfer of external
> call to phone B (still caller ID & number of phone A).****
>
> ** **
>
> ** **
>
> Thanks in advance and best regards,****
>
> ** **
>
> Stefan****
>
> ** **
>
> ** **
>
> *Stefan Weigel*
> System Specialist AITP****
>
> **
>
> *Allianz Automotive Services GmbH*****
>
> Einsteinring 28
> 85609 Aschheim
> Germany****
>
> Tel.: +49 89 2000 48 975
> Fax: +49 89 2000 48 566
> eMail: Stefan.Weigel at allianz-warranty.com ****
>
>
> http://www.allianz-warranty.com
> Gesch?ftsf?hrung: Andreas R?sing, Horst Ziegler
> Amtsgericht M?nchen, HRB 175682
> F?r Umsatzsteuerzwecke: Ust-ID-Nr.: DE 262 617 720****
>
> ** **
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110818/99ea7857/attachment.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: image/gif
Size: 4415 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110818/99ea7857/attachment.gif 

From robert.hadley at teotech.com  Thu Aug 18 20:47:19 2011
From: robert.hadley at teotech.com (Robert Hadley)
Date: Thu, 18 Aug 2011 09:47:19 -0700
Subject: [Freeswitch-users] Freetdm bridge to SIP DTMF problem
In-Reply-To: <4E4D0B5F.2060004@gmail.com>
References: <4E4CD94C.3030403@comgate.cz> <4E4CEEB1.3040202@gmail.com>
	<4E4D0B5F.2060004@gmail.com>
Message-ID: <E826C94A3C6BE44198FAF4D66B5AB4D81BDC3249F3@VA3DIAXVS351.RED001.local>

Hi,

I also had an incoming PRI DTMF not working issue using Freeswitch with Freetdm and libsng_isdn in July.  I demonstrated this issue to Sangoma.  The problem was resolved in a FS update I pulled July 26th.

Regards,
Robert


-----Original Message-----
From: Jakub Ouhrabka [mailto:jakub.ouhrabka at gmail.com] 
Sent: Thursday, August 18, 2011 5:54 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Freetdm bridge to SIP DTMF problem

Hi,

I've tried further: both dtmf-type info and rfc2833 are not working (can't see it in packet dump). Only dtmf-type set to "none" together with start_dtmf_generate is working as expected for me - dtmf is generated inband.

How to force freeswitch to forward dtmf from freetdm to sofia preferably using rfc2833?

Thank you for any hints,

Jakub

Dne 18.8.2011 12:51, Jakub Ouhrabka napsal(a):
> Hi,
>
> we've changed OpenZAP to FreeTDM sometime ago and from that time we're 
> experiencing problems with DTMF.
>
> We're bridging incoming ISDN30 calls from Sangoma A108DE card + libri 
> + FreeTDM to SIP outgoing calls. Freeswitch is recognizing DTMF 
> correctly but DTMF is not send to outgoing SIP call - it's not present 
> in RTP stream trace (we've setup RFC2833 DTMF signalling). When 
> bridging the call back to ISDN30 DTMF works correctly.
>
> We're using latest versions of all software as of 1 month ago.
>
> Below are attached two snippets of logs: first is bridging back to
> ISDN30 where DTMF is sent to called party. Second is bridging to SIP 
> where DTMF is not sent to the called party.
>
> We've tried it with both <action application="start_dtmf" /> on and 
> off with no success.
>
> Any pointers how to investigate the issue?
>
> Thanks,
>
> Jakub
>
> Log snippets:
>
> ISDN30 to ISDN30 bridge - DTMF ok
>
> 2011-08-18 12:31:45.950158 [DEBUG] ftmod_wanpipe.c:1545 [s1c11][1:11] 
> read wanpipe event 3
> 2011-08-18 12:31:45.950158 [DEBUG] ftmod_wanpipe.c:1415 [s1c11][1:11] 
> Queuing wanpipe DTMF: 5
> 2011-08-18 12:31:45.950158 [DEBUG] ftdm_io.c:3504 [s1c11][1:11] 
> Queuing DTMF 5 (debug = 0)
> 2011-08-18 12:31:45.950158 [DEBUG] mod_freetdm.c:733 Queuing DTMF [5] 
> in channel FreeTDM/1:11/XXXXXXXXX
> 2011-08-18 12:31:45.970623 [DEBUG] switch_ivr_bridge.c:391 Send signal 
> FreeTDM/1:5/XXXXXXXXX [BREAK]
> 2011-08-18 12:31:45.991084 [DEBUG] ftdm_io.c:3694 [s1c5][1:5] 
> Generating DTMF [5]
>
> ISDN30 to SIP bridge - DTMF not forwarded
>
> 2011-08-18 12:34:05.287480 [DEBUG] ftmod_wanpipe.c:1545 [s1c29][1:29] 
> read wanpipe event 3
> 2011-08-18 12:34:05.607708 [DEBUG] ftmod_wanpipe.c:1545 [s1c29][1:29] 
> read wanpipe event 3
> 2011-08-18 12:34:05.607708 [DEBUG] ftmod_wanpipe.c:1415 [s1c29][1:29] 
> Queuing wanpipe DTMF: 6
> 2011-08-18 12:34:05.607708 [DEBUG] ftdm_io.c:3504 [s1c29][1:29] 
> Queuing DTMF 6 (debug = 0)
> 2011-08-18 12:34:05.607708 [DEBUG] ftmod_libpri.c:1586 -- Caught Event 
> span 1 18 (KEYPAD_DIGIT)
> 2011-08-18 12:34:05.628171 [DEBUG] mod_freetdm.c:733 Queuing DTMF [6] 
> in channel FreeTDM/1:29/XXXXXXXXXXXXXXX
> 2011-08-18 12:34:05.628171 [DEBUG] switch_ivr_bridge.c:391 Send signal
> sofia/external/XXXXXXX at XXXXXXX:5060 [BREAK]
>



From steveu at coppice.org  Thu Aug 18 20:50:42 2011
From: steveu at coppice.org (Steve Underwood)
Date: Fri, 19 Aug 2011 00:50:42 +0800
Subject: [Freeswitch-users] Best tabletop conference phone
In-Reply-To: <CAB2KLydFq-UmZfbOaypj5uWoFyFHjtsLZ+po-H4P3Jja+RTPqw@mail.gmail.com>
References: <CAB2KLydFq-UmZfbOaypj5uWoFyFHjtsLZ+po-H4P3Jja+RTPqw@mail.gmail.com>
Message-ID: <4E4D42E2.7050909@coppice.org>

On 08/18/2011 11:50 PM, Luis F Urrea wrote:
> Hi all,
>
> After using a Snom Meeting Point for 9 months that suddenly went dead 
> while connected to PoE which in turn is properly connected to UPS my 
> customer is extremely upset and frustrated to that experience.
>
> He is asking me for a quote on the best possible tabletop conference 
> phone that we can use with FS since he is not even interested in 
> processing the Meeting Point warranty.
>
> Please please please,your input is appreciated!!!
>
> Regards,
>
> Luis
>
Are you really saying they had 9 months of satisfactory service before 
an unfortunate breakdown, and the customer's faith in the product is 
totally destroyed? They seem more than a little easy to annoy. :-\

Steve


From a.afzali2003 at gmail.com  Thu Aug 18 21:50:17 2011
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Thu, 18 Aug 2011 22:20:17 +0430
Subject: [Freeswitch-users] Exchanging IMs via RTMP
Message-ID: <CABK0fC7jR2sdpjZF8pEUP7MsaJomnOqkDUZ55xYNEFppNq+3XQ@mail.gmail.com>

Hi FreeSWITCHers,

I think about a chat service which an anonymous be able to send and receive
IM to / from a selected user. Based on the flex client and the server side
code it seems that sending SEND_MESSAGE event can handle half of this
scenario. My questions is : if the anonymous, registers itself as a fake /
temporary user, will be able to receive IM as well ? It seems that onEvent
function does not invoke in netStatus fuction !

BEST,
-- afshin
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110818/39f29045/attachment.html 

From lfurrea at gmail.com  Thu Aug 18 21:57:47 2011
From: lfurrea at gmail.com (Luis F Urrea)
Date: Thu, 18 Aug 2011 11:57:47 -0600
Subject: [Freeswitch-users] Best tabletop conference phone
In-Reply-To: <4E4D42E2.7050909@coppice.org>
References: <CAB2KLydFq-UmZfbOaypj5uWoFyFHjtsLZ+po-H4P3Jja+RTPqw@mail.gmail.com>
	<4E4D42E2.7050909@coppice.org>
Message-ID: <CAB2KLyeEZShKZiMz0U-PjNuNdK2sVv_SSny+LLvqWVDomwRh-w@mail.gmail.com>

Well unfortunately the 9 months were not that satisfactory. We had issues
with the phone dialing digits twice and no firmware image was able to
correct that.

For a  $600+ phone I guess they indeed would expect to be able to dial
without hassles and something more that 1 year of use, but yes they may
indeed be a little picky.

As a systems integrator I only had 1 conference phone deployed and be
hearing from your customer every other week that the phone is dialing digits
twice and get no support at all from the manufacturer and  after 9 months of
use be dealing with RMA is not exactly thrilling.

Now Snom 300 series, rock solid! I have 150> deployed for more than 2 years
and so far just a couple of RMAs top and a couple of handsets replaced, that
is the kind of experience that I would mention as enjoyable.

The 800 series is still giving me headaches but I assume the firmware will
get better with time.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110818/c24c5d1e/attachment.html 

From freeswitch-list at puzzled.xs4all.nl  Thu Aug 18 22:15:26 2011
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Thu, 18 Aug 2011 20:15:26 +0200
Subject: [Freeswitch-users] Best tabletop conference phone
In-Reply-To: <CAB2KLydFq-UmZfbOaypj5uWoFyFHjtsLZ+po-H4P3Jja+RTPqw@mail.gmail.com>
References: <CAB2KLydFq-UmZfbOaypj5uWoFyFHjtsLZ+po-H4P3Jja+RTPqw@mail.gmail.com>
Message-ID: <4E4D56BE.9030105@puzzled.xs4all.nl>

On 08/18/2011 05:50 PM, Luis F Urrea wrote:
> Hi all,
>
> After using a Snom Meeting Point for 9 months that suddenly went dead
> while connected to PoE which in turn is properly connected to UPS my
> customer is extremely upset and frustrated to that experience.
>
> He is asking me for a quote on the best possible tabletop conference
> phone that we can use with FS since he is not even interested in
> processing the Meeting Point warranty.
>
> Please please please,your input is appreciated!!!

The Polycom Soundstation IP7000 works very well.

Regards,
Patrick


From covici at ccs.covici.com  Thu Aug 18 22:17:47 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 18 Aug 2011 14:17:47 -0400
Subject: [Freeswitch-users] having problems building mono 2.10
In-Reply-To: <13751.1313679393@ccs.covici.com>
References: <24703.1313653049@ccs.covici.com> <1656439.BN3H24Bjmd@tsukasa>
	<13751.1313679393@ccs.covici.com>
Message-ID: <24482.1313691467@ccs.covici.com>

OK, so I have mono built, but mod_managed blows up -- at least with this
version of mono-- I wonder have you or anyone been successful in getting
that combination to work?

covici at ccs.covici.com wrote:

> Thanks -- my google search did not find this at all.
> 
> Stefan Knoblich <stkn at freeswitch.org> wrote:
> 
> > On Thursday 18 August 2011 03:37:29 covici at ccs.covici.com wrote:
> > > Hi.  I am using gentoo, so I have to build mono by compiling and when I
> > > try to do this I get the following error:
> > > 
> > > make[8]: Entering directory
> > > `/var/tmp/portage/dev-lang/mono-2.10.2-r1/work/mono-2.10.2/mcs/tools/gacutil'
> > > MCS     [basic] gacutil.exe
> > > Inconsistency detected by ld.so: dl-deps.c: 622: _dl_map_object_deps:
> > > Assertion `nlist > 1' failed!
> > > make[8]: *** [../../class/lib/basic/gacutil.exe] Error 127
> > > 
> > > Any assistance on this would be appreciated.
> > 
> > 
> > not a mono bug:  https://bugs.gentoo.org/show_bug.cgi?id=374107
> > 
> > -- 
> > -------------------------------------------------------------------------------
> > Stefan Knoblich                        | Web:   http://www.axsentis.de/
> > axsentis GmbH                          |        http://oss.axsentis.de/
> > Eupener Str. 74, 50933 Koeln, Germany  |
> > Amtsgericht Koeln: HR B 56238          | Email: s.knoblich at axsentis.de
> > UST-ID: DE244977565                    | JID:   s.knoblich at jabber.axsentis.de
> > -------------------------------------------------------------------------------
> > Web:   http://stkn.techmage.de/
> > Email: stkn at freeswitch.org
> > IRC:   #freeswitch-de @ irc.freenode.net
> -- 
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
> 
>          John Covici
>          covici at ccs.covici.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

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From lakersman2006 at yahoo.com  Thu Aug 18 22:27:27 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Thu, 18 Aug 2011 11:27:27 -0700 (PDT)
Subject: [Freeswitch-users] RFC2833 and Inband DTMF
Message-ID: <1313692047.359.YahooMailNeo@web161001.mail.bf1.yahoo.com>

Hi,

I wanted to know if Freeswitch offered a dtmf mode to negotiate whether the inbound/A-leg's dtmf is rfc2833 or inband compatible? Just like how Asterisk has "dtmfmode=auto". 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110818/1d88e623/attachment.html 

From msc at freeswitch.org  Thu Aug 18 23:08:41 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 18 Aug 2011 12:08:41 -0700
Subject: [Freeswitch-users] VegaStream Contacts
Message-ID: <CAKzWOxVTW-Y=Y4hCgTC6QDXLb_WxPsvt5a_+fOsd3Ty7=kr5Jw@mail.gmail.com>

Hello all,

If you know someone who works at VegaStream please contact me off list. I
need to contact VegaStream and I prefer not to go in cold if I can help it.

Thanks,
Michael
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110818/fc076257/attachment.html 

From mgg at giagnocavo.net  Thu Aug 18 23:23:14 2011
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Thu, 18 Aug 2011 15:23:14 -0400
Subject: [Freeswitch-users] having problems building mono 2.10
In-Reply-To: <24482.1313691467@ccs.covici.com>
References: <24703.1313653049@ccs.covici.com> <1656439.BN3H24Bjmd@tsukasa>
	<13751.1313679393@ccs.covici.com> <24482.1313691467@ccs.covici.com>
Message-ID: <03351FCC6082174C8534AB714B8258A5E416C495@mse17be1.mse17.exchange.ms>

Can you provide details on "blows up"?

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of covici at ccs.covici.com
Sent: Thursday, August 18, 2011 12:18 PM
To: FreeSWITCH Users Help
Cc: Stefan Knoblich
Subject: Re: [Freeswitch-users] having problems building mono 2.10

OK, so I have mono built, but mod_managed blows up -- at least with this version of mono-- I wonder have you or anyone been successful in getting that combination to work?

covici at ccs.covici.com wrote:

> Thanks -- my google search did not find this at all.
> 
> Stefan Knoblich <stkn at freeswitch.org> wrote:
> 
> > On Thursday 18 August 2011 03:37:29 covici at ccs.covici.com wrote:
> > > Hi.  I am using gentoo, so I have to build mono by compiling and 
> > > when I try to do this I get the following error:
> > > 
> > > make[8]: Entering directory
> > > `/var/tmp/portage/dev-lang/mono-2.10.2-r1/work/mono-2.10.2/mcs/tools/gacutil'
> > > MCS     [basic] gacutil.exe
> > > Inconsistency detected by ld.so: dl-deps.c: 622: _dl_map_object_deps:
> > > Assertion `nlist > 1' failed!
> > > make[8]: *** [../../class/lib/basic/gacutil.exe] Error 127
> > > 
> > > Any assistance on this would be appreciated.
> > 
> > 
> > not a mono bug:  https://bugs.gentoo.org/show_bug.cgi?id=374107
> > 
> > --
> > -------------------------------------------------------------------------------
> > Stefan Knoblich                        | Web:   http://www.axsentis.de/
> > axsentis GmbH                          |        http://oss.axsentis.de/
> > Eupener Str. 74, 50933 Koeln, Germany  |
> > Amtsgericht Koeln: HR B 56238          | Email: s.knoblich at axsentis.de
> > UST-ID: DE244977565                    | JID:   s.knoblich at jabber.axsentis.de
> > -------------------------------------------------------------------------------
> > Web:   http://stkn.techmage.de/
> > Email: stkn at freeswitch.org
> > IRC:   #freeswitch-de @ irc.freenode.net
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
> 
>          John Covici
>          covici at ccs.covici.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-use
> rs
> http://www.freeswitch.org

--
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


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


From covici at ccs.covici.com  Thu Aug 18 23:50:19 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 18 Aug 2011 15:50:19 -0400
Subject: [Freeswitch-users] having problems building mono 2.10
In-Reply-To: <03351FCC6082174C8534AB714B8258A5E416C495@mse17be1.mse17.exchange.ms>
References: <24703.1313653049@ccs.covici.com> <1656439.BN3H24Bjmd@tsukasa>
	<13751.1313679393@ccs.covici.com> <24482.1313691467@ccs.covici.com>
	<03351FCC6082174C8534AB714B8258A5E416C495@mse17be1.mse17.exchange.ms>
Message-ID: <4501.1313697019@ccs.covici.com>

When I load the module, from the cli, it just says socket interrupt, bye
and fs shuts down.  The last line before that is:
2011-08-18 13:30:50.392788 [DEBUG] switch_cpp.cpp:1197
FreeSWITCH.Managed loader is starting with directory
'/usr/local/freeswitch/mod/managed'.
It does not seem to be even a seg fault.


Michael Giagnocavo <mgg at giagnocavo.net> wrote:

> Can you provide details on "blows up"?
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of covici at ccs.covici.com
> Sent: Thursday, August 18, 2011 12:18 PM
> To: FreeSWITCH Users Help
> Cc: Stefan Knoblich
> Subject: Re: [Freeswitch-users] having problems building mono 2.10
> 
> OK, so I have mono built, but mod_managed blows up -- at least with this version of mono-- I wonder have you or anyone been successful in getting that combination to work?
> 
> covici at ccs.covici.com wrote:
> 
> > Thanks -- my google search did not find this at all.
> > 
> > Stefan Knoblich <stkn at freeswitch.org> wrote:
> > 
> > > On Thursday 18 August 2011 03:37:29 covici at ccs.covici.com wrote:
> > > > Hi.  I am using gentoo, so I have to build mono by compiling and 
> > > > when I try to do this I get the following error:
> > > > 
> > > > make[8]: Entering directory
> > > > `/var/tmp/portage/dev-lang/mono-2.10.2-r1/work/mono-2.10.2/mcs/tools/gacutil'
> > > > MCS     [basic] gacutil.exe
> > > > Inconsistency detected by ld.so: dl-deps.c: 622: _dl_map_object_deps:
> > > > Assertion `nlist > 1' failed!
> > > > make[8]: *** [../../class/lib/basic/gacutil.exe] Error 127
> > > > 
> > > > Any assistance on this would be appreciated.
> > > 
> > > 
> > > not a mono bug:  https://bugs.gentoo.org/show_bug.cgi?id=374107
> > > 
> > > --
> > > -------------------------------------------------------------------------------
> > > Stefan Knoblich                        | Web:   http://www.axsentis.de/
> > > axsentis GmbH                          |        http://oss.axsentis.de/
> > > Eupener Str. 74, 50933 Koeln, Germany  |
> > > Amtsgericht Koeln: HR B 56238          | Email: s.knoblich at axsentis.de
> > > UST-ID: DE244977565                    | JID:   s.knoblich at jabber.axsentis.de
> > > -------------------------------------------------------------------------------
> > > Web:   http://stkn.techmage.de/
> > > Email: stkn at freeswitch.org
> > > IRC:   #freeswitch-de @ irc.freenode.net
> > --
> > Your life is like a penny.  You're going to lose it.  The question is:
> > How do
> > you spend it?
> > 
> >          John Covici
> >          covici at ccs.covici.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-use
> > rs
> > http://www.freeswitch.org
> 
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
> 
>          John Covici
>          covici at ccs.covici.com
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From mgg at giagnocavo.net  Fri Aug 19 00:02:08 2011
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Thu, 18 Aug 2011 16:02:08 -0400
Subject: [Freeswitch-users] having problems building mono 2.10
In-Reply-To: <4501.1313697019@ccs.covici.com>
References: <24703.1313653049@ccs.covici.com> <1656439.BN3H24Bjmd@tsukasa>
	<13751.1313679393@ccs.covici.com> <24482.1313691467@ccs.covici.com>
	<03351FCC6082174C8534AB714B8258A5E416C495@mse17be1.mse17.exchange.ms>
	<4501.1313697019@ccs.covici.com>
Message-ID: <03351FCC6082174C8534AB714B8258A5E416C4B8@mse17be1.mse17.exchange.ms>

Are there any managed modules in that directory, and does that directory exist? Could you pastebin the last 20 lines or so before death?	

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of covici at ccs.covici.com
Sent: Thursday, August 18, 2011 1:50 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] having problems building mono 2.10

When I load the module, from the cli, it just says socket interrupt, bye and fs shuts down.  The last line before that is:
2011-08-18 13:30:50.392788 [DEBUG] switch_cpp.cpp:1197 FreeSWITCH.Managed loader is starting with directory '/usr/local/freeswitch/mod/managed'.
It does not seem to be even a seg fault.


Michael Giagnocavo <mgg at giagnocavo.net> wrote:

> Can you provide details on "blows up"?
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org 
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of 
> covici at ccs.covici.com
> Sent: Thursday, August 18, 2011 12:18 PM
> To: FreeSWITCH Users Help
> Cc: Stefan Knoblich
> Subject: Re: [Freeswitch-users] having problems building mono 2.10
> 
> OK, so I have mono built, but mod_managed blows up -- at least with this version of mono-- I wonder have you or anyone been successful in getting that combination to work?
> 
> covici at ccs.covici.com wrote:
> 
> > Thanks -- my google search did not find this at all.
> > 
> > Stefan Knoblich <stkn at freeswitch.org> wrote:
> > 
> > > On Thursday 18 August 2011 03:37:29 covici at ccs.covici.com wrote:
> > > > Hi.  I am using gentoo, so I have to build mono by compiling and 
> > > > when I try to do this I get the following error:
> > > > 
> > > > make[8]: Entering directory
> > > > `/var/tmp/portage/dev-lang/mono-2.10.2-r1/work/mono-2.10.2/mcs/tools/gacutil'
> > > > MCS     [basic] gacutil.exe
> > > > Inconsistency detected by ld.so: dl-deps.c: 622: _dl_map_object_deps:
> > > > Assertion `nlist > 1' failed!
> > > > make[8]: *** [../../class/lib/basic/gacutil.exe] Error 127
> > > > 
> > > > Any assistance on this would be appreciated.
> > > 
> > > 
> > > not a mono bug:  https://bugs.gentoo.org/show_bug.cgi?id=374107
> > > 
> > > --
> > > -------------------------------------------------------------------------------
> > > Stefan Knoblich                        | Web:   http://www.axsentis.de/
> > > axsentis GmbH                          |        http://oss.axsentis.de/
> > > Eupener Str. 74, 50933 Koeln, Germany  |
> > > Amtsgericht Koeln: HR B 56238          | Email: s.knoblich at axsentis.de
> > > UST-ID: DE244977565                    | JID:   s.knoblich at jabber.axsentis.de
> > > -------------------------------------------------------------------------------
> > > Web:   http://stkn.techmage.de/
> > > Email: stkn at freeswitch.org
> > > IRC:   #freeswitch-de @ irc.freenode.net
> > --
> > Your life is like a penny.  You're going to lose it.  The question is:
> > How do
> > you spend it?
> > 
> >          John Covici
> >          covici at ccs.covici.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-u
> > se
> > rs
> > http://www.freeswitch.org
> 
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
> 
>          John Covici
>          covici at ccs.covici.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-use
> rs
> http://www.freeswitch.org
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org

--
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


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


From covici at ccs.covici.com  Fri Aug 19 00:12:16 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 18 Aug 2011 16:12:16 -0400
Subject: [Freeswitch-users] having problems building mono 2.10
In-Reply-To: <03351FCC6082174C8534AB714B8258A5E416C4B8@mse17be1.mse17.exchange.ms>
References: <24703.1313653049@ccs.covici.com> <1656439.BN3H24Bjmd@tsukasa>
	<13751.1313679393@ccs.covici.com> <24482.1313691467@ccs.covici.com>
	<03351FCC6082174C8534AB714B8258A5E416C495@mse17be1.mse17.exchange.ms>
	<4501.1313697019@ccs.covici.com>
	<03351FCC6082174C8534AB714B8258A5E416C4B8@mse17be1.mse17.exchange.ms>
Message-ID: <7495.1313698336@ccs.covici.com>

The managed directory is empty -- here are the last lines before dying.
2011-08-18 13:30:50.352782 [DEBUG] switch_loadable_module.c:890 Loading
module with global namespace at request of module
2011-08-18 13:30:50.352782 [INFO] mod_managed.cpp:311 Loading
mod_managed (Common Language Infrastructure), Mono Version
2011-08-18 13:30:50.352782 [INFO] mod_managed.cpp:215 Calling
mono_assembly_loaded.
2011-08-18 13:30:50.352782 [INFO] mod_managed.cpp:219 Calling
mono_domain_assembly_open.
2011-08-18 13:30:50.352782 [DEBUG] mod_managed.cpp:261 Found all loader
functions.
2011-08-18 13:30:50.392788 [DEBUG] switch_cpp.cpp:1197
FreeSWITCH.Managed loader is starting with directory
'/usr/local/freeswitch/mod/managed'.

Have you been using 2.10?


Michael Giagnocavo <mgg at giagnocavo.net> wrote:

> Are there any managed modules in that directory, and does that directory exist? Could you pastebin the last 20 lines or so before death?	
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of covici at ccs.covici.com
> Sent: Thursday, August 18, 2011 1:50 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] having problems building mono 2.10
> 
> When I load the module, from the cli, it just says socket interrupt, bye and fs shuts down.  The last line before that is:
> 2011-08-18 13:30:50.392788 [DEBUG] switch_cpp.cpp:1197 FreeSWITCH.Managed loader is starting with directory '/usr/local/freeswitch/mod/managed'.
> It does not seem to be even a seg fault.
> 
> 
> Michael Giagnocavo <mgg at giagnocavo.net> wrote:
> 
> > Can you provide details on "blows up"?
> > 
> > -----Original Message-----
> > From: freeswitch-users-bounces at lists.freeswitch.org 
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of 
> > covici at ccs.covici.com
> > Sent: Thursday, August 18, 2011 12:18 PM
> > To: FreeSWITCH Users Help
> > Cc: Stefan Knoblich
> > Subject: Re: [Freeswitch-users] having problems building mono 2.10
> > 
> > OK, so I have mono built, but mod_managed blows up -- at least with this version of mono-- I wonder have you or anyone been successful in getting that combination to work?
> > 
> > covici at ccs.covici.com wrote:
> > 
> > > Thanks -- my google search did not find this at all.
> > > 
> > > Stefan Knoblich <stkn at freeswitch.org> wrote:
> > > 
> > > > On Thursday 18 August 2011 03:37:29 covici at ccs.covici.com wrote:
> > > > > Hi.  I am using gentoo, so I have to build mono by compiling and 
> > > > > when I try to do this I get the following error:
> > > > > 
> > > > > make[8]: Entering directory
> > > > > `/var/tmp/portage/dev-lang/mono-2.10.2-r1/work/mono-2.10.2/mcs/tools/gacutil'
> > > > > MCS     [basic] gacutil.exe
> > > > > Inconsistency detected by ld.so: dl-deps.c: 622: _dl_map_object_deps:
> > > > > Assertion `nlist > 1' failed!
> > > > > make[8]: *** [../../class/lib/basic/gacutil.exe] Error 127
> > > > > 
> > > > > Any assistance on this would be appreciated.
> > > > 
> > > > 
> > > > not a mono bug:  https://bugs.gentoo.org/show_bug.cgi?id=374107
> > > > 
> > > > --
> > > > -------------------------------------------------------------------------------
> > > > Stefan Knoblich                        | Web:   http://www.axsentis.de/
> > > > axsentis GmbH                          |        http://oss.axsentis.de/
> > > > Eupener Str. 74, 50933 Koeln, Germany  |
> > > > Amtsgericht Koeln: HR B 56238          | Email: s.knoblich at axsentis.de
> > > > UST-ID: DE244977565                    | JID:   s.knoblich at jabber.axsentis.de
> > > > -------------------------------------------------------------------------------
> > > > Web:   http://stkn.techmage.de/
> > > > Email: stkn at freeswitch.org
> > > > IRC:   #freeswitch-de @ irc.freenode.net
> > > --
> > > Your life is like a penny.  You're going to lose it.  The question is:
> > > How do
> > > you spend it?
> > > 
> > >          John Covici
> > >          covici at ccs.covici.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-u
> > > se
> > > rs
> > > http://www.freeswitch.org
> > 
> > --
> > Your life is like a penny.  You're going to lose it.  The question is:
> > How do
> > you spend it?
> > 
> >          John Covici
> >          covici at ccs.covici.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-use
> > rs
> > http://www.freeswitch.org
> > 
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> > rs
> > http://www.freeswitch.org
> 
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
> 
>          John Covici
>          covici at ccs.covici.com
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From jchavanton at gmail.com  Thu Aug 18 22:37:42 2011
From: jchavanton at gmail.com (Julien Chavanton)
Date: Thu, 18 Aug 2011 14:37:42 -0400
Subject: [Freeswitch-users] g.729 licence usage
Message-ID: <CAKmcL2mJCvDUG4UbAQdL9aMjOFAajSQnKNGAu3PZPAVSjEVfYg@mail.gmail.com>

It seems 2 calls bridged with recording consume 3 licences ? (2 encoders and
3 decoders)

Is there a way to minimize the usage, I do not want DTMF capture (inband)
but only need to record.

freeswitch at internal> show calls
call_uuid,call_created,call_created_epoch,function,caller_cid_name,caller_cid_num,caller_dest_num,caller_chan_name,caller_uuid,callee_cid_name,callee_cid_num,callee_dest_num,callee_chan_name,callee_uuid,hostname
e0567680-8c11-4cce-8462-b277f8632d0e,2011-08-18
18:31:12,1313692272,switch_ivr_multi_threaded_bridge,Outbound
Call,1514xxxxxxx,xxxxxxxxx,sofia/external/1514xxxxxxx,e0567680-8c11-4cce-8462-b277f8632d0e,Outbound
Call,2xxxxxxxxxx,b_leg_answer,sofia/external/2xxxxxxxxxx,419da0d6-7723-4f2c-b856-20c73cbd91c1,voxcast01

1 total.

freeswitch at internal> g729_used
2:3
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110818/1157f34e/attachment.html 

From dujinfang at gmail.com  Fri Aug 19 01:41:33 2011
From: dujinfang at gmail.com (Seven Du)
Date: Fri, 19 Aug 2011 05:41:33 +0800
Subject: [Freeswitch-users] strdup error on solaris
In-Reply-To: <CAKbxfG8BUmhTfr0GEoRrb=g1LcaeRLasmf+Mxt24xgPaG6Eiag@mail.gmail.com>
References: <7E3EF9F827D54F628C130D0AC650BAD0@gmail.com>
	<CAKbxfG8BUmhTfr0GEoRrb=g1LcaeRLasmf+Mxt24xgPaG6Eiag@mail.gmail.com>
Message-ID: <DA68975C70B74A1399C5B8BBE649A6F3@gmail.com>

On Thursday, August 18, 2011 at 10:56 PM, Anthony Minessale wrote: 
> hmm, did solaris stop putting strdup in string.h ?
> 
> 
looks it's still there. Do you think I should report a jira?



#if defined(__EXTENSIONS__) || \
 (!defined(_STRICT_STDC) && !defined(__XOPEN_OR_POSIX)) || \
 defined(_XPG4_2)
extern char *strdup(const char *);
#endif


#if defined(__EXTENSIONS__) || !defined(__XOPEN_OR_POSIX) || defined(_XPG4_2)
extern char *strdup();
#endif


http://pastebin.freeswitch.org/17088

> On Thu, Aug 18, 2011 at 9:52 AM, Seven Du <dujinfang at gmail.com (mailto:dujinfang at gmail.com)> wrote:
> > Hi,
> > I'm new to solaris
> > SunOS solaris 5.11 snv_134 i86pc i386 i86pc
> > I didn't follow the wiki about installing on solaris but installed building
> > tool chain with pkg install SUNWgcc etc.
> > bootstrap and configure was ok, however, I got error on gmake. Any highlight
> > on this? Thanks.
> > libs/stfu/stfu.c: In function `stfu_n_debug':
> > libs/stfu/stfu.c:224: warning: implicit declaration of function `strdup'
> > libs/stfu/stfu.c:224: warning: assignment makes pointer from integer without
> > a cast
> > libs/stfu/stfu.c:227: warning: assignment makes pointer from integer without
> > a cast
> > libs/stfu/stfu.c: In function `stfu_n_init':
> > libs/stfu/stfu.c:302: warning: assignment makes pointer from integer without
> > a cast
> > gmake[1]: *** [libfreeswitch_la-stfu.lo] #### 1
> > gmake: *** [all] #### 2
> > link at solaris:~/seven/freeswitch# vi libs/stfu/stfu.c:224
> > link at solaris:~/seven/freeswitch# vi libs/stfu/stfu.c
> > 
> > link at solaris:~/seven/freeswitch# gcc -v
> > Reading specs from /usr/sfw/lib/gcc/i386-pc-solaris2.11/3.4.3/specs
> > Configured with: /builds2/sfwnv-gate/usr/src/cmd/gcc/gcc-3.4.3/configure
> > --prefix=/usr/sfw --with-as=/usr/sfw/bin/gas --with-gnu-as
> > --with-ld=/usr/ccs/bin/ld --without-gnu-ld --enable-languages=c,c++,f77,objc
> > --enable-shared
> > Thread model: posix
> > gcc version 3.4.3 (csl-sol210-3_4-20050802)
> > --
> > Seven Du
> > About: http://about.me/dujinfang
> > Blog: http://www.dujinfang.com
> > Proj: http://www.freeswitch.org.cn
> > Sent with Sparrow
> > 
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org)
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com (mailto:anthony_minessale at hotmail.com)
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com (mailto:anthony.minessale at gmail.com)
> IRC: irc.freenode.net (http://irc.freenode.net) #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org (mailto:888 at conference.freeswitch.org)
> googletalk:conf+888 at conference.freeswitch.org (mailto:conf+888 at conference.freeswitch.org)
> pstn:+19193869900
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org)
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110819/9e01fc3b/attachment.html 

From avi at avimarcus.net  Fri Aug 19 02:18:27 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 19 Aug 2011 01:18:27 +0300
Subject: [Freeswitch-users] Goip GSM Gateway works great with FreeSwitch!
In-Reply-To: <1313582063738-6695132.post@n2.nabble.com>
References: <1313496506598-6691087.post@n2.nabble.com>
	<CALNZuwnvSNaTD4LVwikb71YW4t2aa6=XWYyXaE17oWimrZ4_eQ@mail.gmail.com>
	<1313582063738-6695132.post@n2.nabble.com>
Message-ID: <CALNZuwnBhiQBTMBuD6sVWM87yDPgYz7ZDFZ2qdyNjha8UW+TVw@mail.gmail.com>

Thank you for posting, I made a link from the interop page.

-Avi



On Wed, Aug 17, 2011 at 2:54 PM, Dissident <jcgpoza at gmail.com> wrote:

> Hello Marcus,
>
> here is the link http://wiki.freeswitch.org/wiki/Goip_FreeSwitch_HowTo
>
> Yes, I think it should be linked from the Interop List.
>
> Best regards.
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Goip-GSM-Gateway-works-great-with-FreeSwitch-tp6691087p6695132.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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110819/638d3d57/attachment-0001.html 

From jakub.ouhrabka at gmail.com  Fri Aug 19 02:29:31 2011
From: jakub.ouhrabka at gmail.com (Jakub Ouhrabka)
Date: Fri, 19 Aug 2011 00:29:31 +0200
Subject: [Freeswitch-users] Freetdm bridge to SIP DTMF problem
In-Reply-To: <E826C94A3C6BE44198FAF4D66B5AB4D81BDC3249F3@VA3DIAXVS351.RED001.local>
References: <4E4CD94C.3030403@comgate.cz> <4E4CEEB1.3040202@gmail.com>
	<4E4D0B5F.2060004@gmail.com>
	<E826C94A3C6BE44198FAF4D66B5AB4D81BDC3249F3@VA3DIAXVS351.RED001.local>
Message-ID: <4E4D924B.90406@gmail.com>

Hi,

 > I also had an incoming PRI DTMF not working issue using Freeswitch
 > with Freetdm and libsng_isdn in July. [...]

Robert: thank you for the info.

I upgraded to latest FreeSWITCH Version 1.0.head (git-92d2999 2011-08-18 
16-00-19 -0400) with libpri 1.4.12 and latest wanpipe (3.5.20) but the 
problem still persists.

Anyone has any suggestions how to debug the issue, please?

Any help would be greatly appreciated!

Best regards,

Jakub

Dne 18.8.2011 18:47, Robert Hadley napsal(a):
> Hi,
>
> I also had an incoming PRI DTMF not working issue using Freeswitch with Freetdm and libsng_isdn in July.  I demonstrated this issue to Sangoma.  The problem was resolved in a FS update I pulled July 26th.
>
> Regards,
> Robert
>
>
> -----Original Message-----
> From: Jakub Ouhrabka [mailto:jakub.ouhrabka at gmail.com]
> Sent: Thursday, August 18, 2011 5:54 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Freetdm bridge to SIP DTMF problem
>
> Hi,
>
> I've tried further: both dtmf-type info and rfc2833 are not working (can't see it in packet dump). Only dtmf-type set to "none" together with start_dtmf_generate is working as expected for me - dtmf is generated inband.
>
> How to force freeswitch to forward dtmf from freetdm to sofia preferably using rfc2833?
>
> Thank you for any hints,
>
> Jakub
>
> Dne 18.8.2011 12:51, Jakub Ouhrabka napsal(a):
>> Hi,
>>
>> we've changed OpenZAP to FreeTDM sometime ago and from that time we're
>> experiencing problems with DTMF.
>>
>> We're bridging incoming ISDN30 calls from Sangoma A108DE card + libri
>> + FreeTDM to SIP outgoing calls. Freeswitch is recognizing DTMF
>> correctly but DTMF is not send to outgoing SIP call - it's not present
>> in RTP stream trace (we've setup RFC2833 DTMF signalling). When
>> bridging the call back to ISDN30 DTMF works correctly.
>>
>> We're using latest versions of all software as of 1 month ago.
>>
>> Below are attached two snippets of logs: first is bridging back to
>> ISDN30 where DTMF is sent to called party. Second is bridging to SIP
>> where DTMF is not sent to the called party.
>>
>> We've tried it with both<action application="start_dtmf" />  on and
>> off with no success.
>>
>> Any pointers how to investigate the issue?
>>
>> Thanks,
>>
>> Jakub
>>
>> Log snippets:
>>
>> ISDN30 to ISDN30 bridge - DTMF ok
>>
>> 2011-08-18 12:31:45.950158 [DEBUG] ftmod_wanpipe.c:1545 [s1c11][1:11]
>> read wanpipe event 3
>> 2011-08-18 12:31:45.950158 [DEBUG] ftmod_wanpipe.c:1415 [s1c11][1:11]
>> Queuing wanpipe DTMF: 5
>> 2011-08-18 12:31:45.950158 [DEBUG] ftdm_io.c:3504 [s1c11][1:11]
>> Queuing DTMF 5 (debug = 0)
>> 2011-08-18 12:31:45.950158 [DEBUG] mod_freetdm.c:733 Queuing DTMF [5]
>> in channel FreeTDM/1:11/XXXXXXXXX
>> 2011-08-18 12:31:45.970623 [DEBUG] switch_ivr_bridge.c:391 Send signal
>> FreeTDM/1:5/XXXXXXXXX [BREAK]
>> 2011-08-18 12:31:45.991084 [DEBUG] ftdm_io.c:3694 [s1c5][1:5]
>> Generating DTMF [5]
>>
>> ISDN30 to SIP bridge - DTMF not forwarded
>>
>> 2011-08-18 12:34:05.287480 [DEBUG] ftmod_wanpipe.c:1545 [s1c29][1:29]
>> read wanpipe event 3
>> 2011-08-18 12:34:05.607708 [DEBUG] ftmod_wanpipe.c:1545 [s1c29][1:29]
>> read wanpipe event 3
>> 2011-08-18 12:34:05.607708 [DEBUG] ftmod_wanpipe.c:1415 [s1c29][1:29]
>> Queuing wanpipe DTMF: 6
>> 2011-08-18 12:34:05.607708 [DEBUG] ftdm_io.c:3504 [s1c29][1:29]
>> Queuing DTMF 6 (debug = 0)
>> 2011-08-18 12:34:05.607708 [DEBUG] ftmod_libpri.c:1586 -- Caught Event
>> span 1 18 (KEYPAD_DIGIT)
>> 2011-08-18 12:34:05.628171 [DEBUG] mod_freetdm.c:733 Queuing DTMF [6]
>> in channel FreeTDM/1:29/XXXXXXXXXXXXXXX
>> 2011-08-18 12:34:05.628171 [DEBUG] switch_ivr_bridge.c:391 Send signal
>> sofia/external/XXXXXXX at XXXXXXX:5060 [BREAK]
>>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From msc at freeswitch.org  Fri Aug 19 04:02:25 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 18 Aug 2011 17:02:25 -0700
Subject: [Freeswitch-users] INFO: Wiki update
Message-ID: <CAKzWOxWAojX3A2vfA4+_+yB7niSsxnVbokcvnhTR2J8AVLe5hw@mail.gmail.com>

Hello all,

We have been experimenting with some access controls on the wiki. If you get
a dialog box asking for a password then use the same credentials as you
would for pastebin.freeswitch.org. You should only see the dialog box when
logging in or editing pages. Just searching the wiki does not cause the
dialog to show up. We are trying to cut down on the spammers.

Let me know if you run into any weirdness.

Thanks,
MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110818/636ddccb/attachment.html 

From covici at ccs.covici.com  Fri Aug 19 04:50:39 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Thu, 18 Aug 2011 20:50:39 -0400
Subject: [Freeswitch-users] INFO: Wiki update
In-Reply-To: <CAKzWOxWAojX3A2vfA4+_+yB7niSsxnVbokcvnhTR2J8AVLe5hw@mail.gmail.com>
References: <CAKzWOxWAojX3A2vfA4+_+yB7niSsxnVbokcvnhTR2J8AVLe5hw@mail.gmail.com>
Message-ID: <13889.1313715039@ccs.covici.com>

I got the dialog just by going to wiki.freeswitch.org and the user name
/password did not work -- but when I escaped out of the dialog, I was
able to get in to search the wiki.  Very strange.

Michael Collins <msc at freeswitch.org> wrote:

> Hello all,
> 
> We have been experimenting with some access controls on the wiki. If you get
> a dialog box asking for a password then use the same credentials as you
> would for pastebin.freeswitch.org. You should only see the dialog box when
> logging in or editing pages. Just searching the wiki does not cause the
> dialog to show up. We are trying to cut down on the spammers.
> 
> Let me know if you run into any weirdness.
> 
> Thanks,
> MC
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From steveu at coppice.org  Fri Aug 19 05:14:47 2011
From: steveu at coppice.org (Steve Underwood)
Date: Fri, 19 Aug 2011 09:14:47 +0800
Subject: [Freeswitch-users] Best tabletop conference phone
In-Reply-To: <CAB2KLyeEZShKZiMz0U-PjNuNdK2sVv_SSny+LLvqWVDomwRh-w@mail.gmail.com>
References: <CAB2KLydFq-UmZfbOaypj5uWoFyFHjtsLZ+po-H4P3Jja+RTPqw@mail.gmail.com>	<4E4D42E2.7050909@coppice.org>
	<CAB2KLyeEZShKZiMz0U-PjNuNdK2sVv_SSny+LLvqWVDomwRh-w@mail.gmail.com>
Message-ID: <4E4DB907.6000806@coppice.org>

On 08/19/2011 01:57 AM, Luis F Urrea wrote:
> Well unfortunately the 9 months were not that satisfactory. We had 
> issues with the phone dialing digits twice and no firmware image was 
> able to correct that.
>
> For a  $600+ phone I guess they indeed would expect to be able to dial 
> without hassles and something more that 1 year of use, but yes they 
> may indeed be a little picky.
>
> As a systems integrator I only had 1 conference phone deployed and be 
> hearing from your customer every other week that the phone is dialing 
> digits twice and get no support at all from the manufacturer and 
>  after 9 months of use be dealing with RMA is not exactly thrilling.
>
> Now Snom 300 series, rock solid! I have 150> deployed for more than 2 
> years and so far just a couple of RMAs top and a couple of handsets 
> replaced, that is the kind of experience that I would mention as 
> enjoyable.
>
> The 800 series is still giving me headaches but I assume the firmware 
> will get better with time.
>
That makes more sense.

Others have mentioned the Polycom IP7000. There is also an IP6000. These 
are pretty much in the industry standard for desktop conferencing 
phones. If you are not looking for something real cheap, they are the 
ones to look at. Some other makes of conferencing phone (e.g. Cisco) are 
just the Polycoms relabelled.

Steve



From u2nsam at gmail.com  Fri Aug 19 09:31:07 2011
From: u2nsam at gmail.com (Sam)
Date: Fri, 19 Aug 2011 11:01:07 +0530
Subject: [Freeswitch-users] play
Message-ID: <CAE2ghw6OOA8d55tSXR4bj8QdZFcsKjfvvfz9HqnK0Svy0EjsWw@mail.gmail.com>

Hello,

What could be the reason of the below error , i am trying to pass command
from the console . I see the file on the system located
/usr/local/freeswitch/sounds/en/us/callie/ivr-call.wav .


freeswitch at internal> conference conference_1 play ivr-call.wav
(play) File: ivr-call.wav not found.

2011-08-19 05:22:39.455963 [ERR] mod_sndfile.c:194 Error Opening File
[/usr/local/freeswitch/sounds/en/us/callie/ivr-call.wav] [System error : No
such file or directory.]
freeswitch at internal>



freeswitch at internal> conference conference_1 play ivr-call
(play) File: ivr-call not found.

2011-08-19 05:28:00.259488 [ERR] switch_core_file.c:112 Unknown file Format
[/usr/local/freeswitch/sounds/en/us/callie/ivr-call]



Regards
Sam
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110819/e176d565/attachment.html 

From u2nsam at gmail.com  Fri Aug 19 09:59:55 2011
From: u2nsam at gmail.com (Sam)
Date: Fri, 19 Aug 2011 11:29:55 +0530
Subject: [Freeswitch-users] play
In-Reply-To: <CAE2ghw6OOA8d55tSXR4bj8QdZFcsKjfvvfz9HqnK0Svy0EjsWw@mail.gmail.com>
References: <CAE2ghw6OOA8d55tSXR4bj8QdZFcsKjfvvfz9HqnK0Svy0EjsWw@mail.gmail.com>
Message-ID: <CAE2ghw6irUQMubVQ6Z6V8NpoQwGgc=Pd_h+ErW8zwj9ChKX4kQ@mail.gmail.com>

I got the it, it should had been conference conference_1 play
ivr/ivr-call.wav

Regards
Sam

On Fri, Aug 19, 2011 at 11:01 AM, Sam <u2nsam at gmail.com> wrote:

> Hello,
>
> What could be the reason of the below error , i am trying to pass command
> from the console . I see the file on the system located
> /usr/local/freeswitch/sounds/en/us/callie/ivr-call.wav .
>
>
> freeswitch at internal> conference conference_1 play ivr-call.wav
> (play) File: ivr-call.wav not found.
>
> 2011-08-19 05:22:39.455963 [ERR] mod_sndfile.c:194 Error Opening File
> [/usr/local/freeswitch/sounds/en/us/callie/ivr-call.wav] [System error : No
> such file or directory.]
> freeswitch at internal>
>
>
>
> freeswitch at internal> conference conference_1 play ivr-call
> (play) File: ivr-call not found.
>
> 2011-08-19 05:28:00.259488 [ERR] switch_core_file.c:112 Unknown file Format
> [/usr/local/freeswitch/sounds/en/us/callie/ivr-call]
>
>
>
> Regards
> Sam
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110819/72be336e/attachment.html 

From Stefan.Weigel at allianz-warranty.com  Fri Aug 19 11:56:38 2011
From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan)
Date: Fri, 19 Aug 2011 09:56:38 +0200
Subject: [Freeswitch-users] Original Caller ID/number after
	attended	transfer
In-Reply-To: <CAKbxfG-6a26_NqN0UJ46sXXEbr2JHbF76kX5SUmkda7wgRWLZQ@mail.gmail.com>
References: <5003D7D3E06F514E8C682F18D223265C04D3B36D6E@AZWSMS03.azwarranty.int>
	<CAKbxfG-6a26_NqN0UJ46sXXEbr2JHbF76kX5SUmkda7wgRWLZQ@mail.gmail.com>
Message-ID: <5003D7D3E06F514E8C682F18D223265C04D3B36D76@AZWSMS03.azwarranty.int>

Hi Anthony, list,



we're working with Polycom Soundpoint 560 phones. But meanwhile I could solve this problem.

A 'global_getvar' showed me that 'ignore_display_updates' was 'true'. After setting it to 'false' I now get the original caller ID displayed when doing a transfer.



One point is left that I'm currently working on:



on incoming calls I add a leading 0 to the original number. I need to do this because if a internal member wants to call outside he/she needs to dial with leading 0.



[..]



        <action application="export" data="caller_id_number=0$1"/>

        <action application="export" data="effective_caller_id_number=0$1"/>

        <action application="set" data="ignore_display_updates=true"/>



[..]



The initial incoming call is working, the number has a leading zero. Doing an attended transfer to a phone I now have the original number (without leading 0). With a blind transfer it's working.



Any suggestions ?







Thanks in advance and best regards



Stefan





Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Anthony Minessale
Gesendet: Donnerstag, 18. August 2011 18:45
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] Original Caller ID/number after attended transfer



if you get phones that support display updates (polycom, snom, cisco and a few others) yes. It already works.

What phones do you have?



On Thu, Aug 18, 2011 at 7:36 AM, Weigel, Stefan <Stefan.Weigel at allianz-warranty.com<mailto:Stefan.Weigel at allianz-warranty.com>> wrote:

Hi all,



is there a possibility to display the original caller ID & number after doing a attended transfer. External call to phone A -> calls phone B (I see caller ID & number of phone A) -> doing an attended transfer of external call to phone B (still caller ID & number of phone A).





Thanks in advance and best regards,



Stefan





Stefan Weigel
System Specialist AITP



Allianz Automotive Services GmbH

Einsteinring 28
85609 Aschheim
Germany

Tel.: +49 89 2000 48 975<tel:%2B49%2089%202000%2048%20975>
Fax: +49 89 2000 48 566<tel:%2B49%2089%202000%2048%20566>
eMail: Stefan.Weigel at allianz-warranty.com<mailto:Stefan.Weigel at allianz-warranty.com>


http://www.allianz-warranty.com
Gesch?ftsf?hrung: Andreas R?sing, Horst Ziegler
Amtsgericht M?nchen, HRB 175682
F?r Umsatzsteuerzwecke: Ust-ID-Nr.: DE 262 617 720





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







--
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110819/cb336fc3/attachment-0001.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.gif
Type: image/gif
Size: 4415 bytes
Desc: image001.gif
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110819/cb336fc3/attachment-0001.gif 

From jcgpoza at gmail.com  Fri Aug 19 12:10:32 2011
From: jcgpoza at gmail.com (jcgpoza gonzalez)
Date: Fri, 19 Aug 2011 10:10:32 +0200
Subject: [Freeswitch-users] Goip GSM Gateway works great with FreeSwitch!
In-Reply-To: <CALNZuwnBhiQBTMBuD6sVWM87yDPgYz7ZDFZ2qdyNjha8UW+TVw@mail.gmail.com>
References: <1313496506598-6691087.post@n2.nabble.com>
	<CALNZuwnvSNaTD4LVwikb71YW4t2aa6=XWYyXaE17oWimrZ4_eQ@mail.gmail.com>
	<1313582063738-6695132.post@n2.nabble.com>
	<CALNZuwnBhiQBTMBuD6sVWM87yDPgYz7ZDFZ2qdyNjha8UW+TVw@mail.gmail.com>
Message-ID: <CAMDVdehLfMSEOG+M+Ungk-B5tq8WHzh_AniKpPd+kvck6kKbRA@mail.gmail.com>

OK, cheers mate

2011/8/19 Avi Marcus <avi at avimarcus.net>

> Thank you for posting, I made a link from the interop page.
>
> -Avi
>
>
>
>
> On Wed, Aug 17, 2011 at 2:54 PM, Dissident <jcgpoza at gmail.com> wrote:
>
>> Hello Marcus,
>>
>> here is the link http://wiki.freeswitch.org/wiki/Goip_FreeSwitch_HowTo
>>
>> Yes, I think it should be linked from the Interop List.
>>
>> Best regards.
>>
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/Goip-GSM-Gateway-works-great-with-FreeSwitch-tp6691087p6695132.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
>>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110819/451f70c8/attachment.html 

From jakub.ouhrabka at gmail.com  Fri Aug 19 15:17:39 2011
From: jakub.ouhrabka at gmail.com (Jakub Ouhrabka)
Date: Fri, 19 Aug 2011 13:17:39 +0200
Subject: [Freeswitch-users] Freetdm bridge to SIP DTMF problem
In-Reply-To: <4E4D0B5F.2060004@gmail.com>
References: <4E4CD94C.3030403@comgate.cz> <4E4CEEB1.3040202@gmail.com>
	<4E4D0B5F.2060004@gmail.com>
Message-ID: <4E4E4653.3090309@gmail.com>

Hi,

another self follow-up: It seems that I have problem even when bridging 
SIP channel to SIP channel - DTMF is not sent to the called party. So I 
guess it's not Sangoma/Wanpipe/libpri/FreeTDM related. DTMF is correctly 
decoded but not forwarded (according to tcpdump/wireshark).

 From the log:

2011-08-19 11:12:41.757636 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF 5:880
2011-08-19 11:12:41.757636 [DEBUG] switch_ivr_bridge.c:391 Send signal 
sofia/external/XXXXXX at XXXXX.XX:5060 [BREAK]

It must some stupid error in configuration - is there anything I have to 
set in SIP profile, or in bridge or somewhere else? Or must the called 
party somehow advertise that it is capable of receiving DTMF?

Thanks,

Jakub

Dne 18.8.2011 14:53, Jakub Ouhrabka napsal(a):
> Hi,
>
> I've tried further: both dtmf-type info and rfc2833 are not working
> (can't see it in packet dump). Only dtmf-type set to "none" together
> with start_dtmf_generate is working as expected for me - dtmf is
> generated inband.
>
> How to force freeswitch to forward dtmf from freetdm to sofia preferably
> using rfc2833?
>
> Thank you for any hints,
>
> Jakub
>
> Dne 18.8.2011 12:51, Jakub Ouhrabka napsal(a):
>> Hi,
>>
>> we've changed OpenZAP to FreeTDM sometime ago and from that time we're
>> experiencing problems with DTMF.
>>
>> We're bridging incoming ISDN30 calls from Sangoma A108DE card + libri +
>> FreeTDM to SIP outgoing calls. Freeswitch is recognizing DTMF correctly
>> but DTMF is not send to outgoing SIP call - it's not present in RTP
>> stream trace (we've setup RFC2833 DTMF signalling). When bridging the
>> call back to ISDN30 DTMF works correctly.
>>
>> We're using latest versions of all software as of 1 month ago.
>>
>> Below are attached two snippets of logs: first is bridging back to
>> ISDN30 where DTMF is sent to called party. Second is bridging to SIP
>> where DTMF is not sent to the called party.
>>
>> We've tried it with both <action application="start_dtmf" /> on and off
>> with no success.
>>
>> Any pointers how to investigate the issue?
>>
>> Thanks,
>>
>> Jakub
>>
>> Log snippets:
>>
>> ISDN30 to ISDN30 bridge - DTMF ok
>>
>> 2011-08-18 12:31:45.950158 [DEBUG] ftmod_wanpipe.c:1545 [s1c11][1:11]
>> read wanpipe event 3
>> 2011-08-18 12:31:45.950158 [DEBUG] ftmod_wanpipe.c:1415 [s1c11][1:11]
>> Queuing wanpipe DTMF: 5
>> 2011-08-18 12:31:45.950158 [DEBUG] ftdm_io.c:3504 [s1c11][1:11] Queuing
>> DTMF 5 (debug = 0)
>> 2011-08-18 12:31:45.950158 [DEBUG] mod_freetdm.c:733 Queuing DTMF [5] in
>> channel FreeTDM/1:11/XXXXXXXXX
>> 2011-08-18 12:31:45.970623 [DEBUG] switch_ivr_bridge.c:391 Send signal
>> FreeTDM/1:5/XXXXXXXXX [BREAK]
>> 2011-08-18 12:31:45.991084 [DEBUG] ftdm_io.c:3694 [s1c5][1:5] Generating
>> DTMF [5]
>>
>> ISDN30 to SIP bridge - DTMF not forwarded
>>
>> 2011-08-18 12:34:05.287480 [DEBUG] ftmod_wanpipe.c:1545 [s1c29][1:29]
>> read wanpipe event 3
>> 2011-08-18 12:34:05.607708 [DEBUG] ftmod_wanpipe.c:1545 [s1c29][1:29]
>> read wanpipe event 3
>> 2011-08-18 12:34:05.607708 [DEBUG] ftmod_wanpipe.c:1415 [s1c29][1:29]
>> Queuing wanpipe DTMF: 6
>> 2011-08-18 12:34:05.607708 [DEBUG] ftdm_io.c:3504 [s1c29][1:29] Queuing
>> DTMF 6 (debug = 0)
>> 2011-08-18 12:34:05.607708 [DEBUG] ftmod_libpri.c:1586 -- Caught Event
>> span 1 18 (KEYPAD_DIGIT)
>> 2011-08-18 12:34:05.628171 [DEBUG] mod_freetdm.c:733 Queuing DTMF [6] in
>> channel FreeTDM/1:29/XXXXXXXXXXXXXXX
>> 2011-08-18 12:34:05.628171 [DEBUG] switch_ivr_bridge.c:391 Send signal
>> sofia/external/XXXXXXX at XXXXXXX:5060 [BREAK]
>>


From benkokakao at gmail.com  Fri Aug 19 17:11:05 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Fri, 19 Aug 2011 15:11:05 +0200
Subject: [Freeswitch-users] Fetching user-variables
Message-ID: <CAAMQ8bRTgb_Dxn_L=C4KEeoNM+WTsy6hJKdQp7d7EjdarLqBjA@mail.gmail.com>

Hi!

I'm working on a redirect script and ran into a problem with toll_allow:
A tries to reach B but B has redirected to C
B is allowed to make external calls(via the user-specific-variable
"toll_allow"), but A is not(Could be a external caller or a
directory-user with restrictions).
B's redirection will not work as she intended, as the dialplan is
restricted through toll_allow.
B is responsible for the external call of A to C, therefore her
accountcode and her toll_allow should apply - unfortunately the
dialplan doesn't know that the call has been redirected by B.
Right now i don't know how to solve this with FreeSWITCH.

Is there a way to fetch user-specific-variables of B, like
"toll_allow", in a call that was not directly initiated by B?

Best regards
Christian


From vipkilla at gmail.com  Fri Aug 19 17:54:56 2011
From: vipkilla at gmail.com (vip killa)
Date: Fri, 19 Aug 2011 09:54:56 -0400
Subject: [Freeswitch-users] enterprise deployment not working
In-Reply-To: <CAC-LwPP+oNWNu-5=-Y2KQwGkn0qaaf7Bq59CcXTo_Ura+AZwBA@mail.gmail.com>
References: <CAC-LwPPSAei6EyeQBtrHoin0DtN73f2zfd6GU5JyQm0Z8_rXPw@mail.gmail.com>
	<CAC-LwPP+oNWNu-5=-Y2KQwGkn0qaaf7Bq59CcXTo_Ura+AZwBA@mail.gmail.com>
Message-ID: <CAC-LwPMAT1zsohk-hCSwpSLetui8-_figFGjDxztB3Z9FFxfZg@mail.gmail.com>

I managed to get this working and i wanted to contribute how... i edited
http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS#OpenSIPS
but the format is all screwed up and i'd like to add an init.d startup
script
perhaps someone could format that wiki page better and add an init.d script?

On Tue, Aug 16, 2011 at 3:22 PM, vip killa <vipkilla at gmail.com> wrote:

> I did an ngrep trace and opensips is returning:
>
> SIP/2.0 503 Service Unavailable
> CSeq: 54 REGISTER
>
> before it even tries the FS server...
> perhaps this wiki page needs to be updated...
>
>
> On Tue, Aug 16, 2011 at 2:03 PM, vip killa <vipkilla at gmail.com> wrote:
>
>> I'm following tutorial at
>> http://wiki.freeswitch.org/wiki/Enterprise_deployment_<http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS>
>> OpenSIPS <http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS> and
>> opensips is not forwarding the registration requests...
>> Running opensips-1.6.4-2-tls,
>> Here is what the debugging looks like when i try to register with the
>> opensip box
>> 10.20.30.17 is the opensips box.
>> 10.20.30.18 is the FreeSWITCH box.
>>
>>
>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>> DBG:core:parse_msg: SIP Request:
>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>> DBG:core:parse_msg:  method:  <REGISTER>
>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>> DBG:core:parse_msg:  uri:     <sip:10.20.30.17>
>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>> DBG:core:parse_msg:  version: <SIP/2.0>
>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>> DBG:core:parse_headers: flags=2
>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>> DBG:core:get_hdr_field: cseq <CSeq>: <43> <REGISTER>
>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>> DBG:core:parse_via_param: found param type 232, <branch> =
>> <z9hG4bKda029da0-e604-1910-80bc-002622a67db9>; state=6
>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>> DBG:core:parse_via_param: found param type 235, <rport> = <n/a>; state=17
>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>> DBG:core:parse_via: end of header reached, state=5
>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>> DBG:core:parse_headers: via found, flags=2
>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>> DBG:core:parse_headers: this is the first via
>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>> DBG:core:receive_msg: After parse_msg...
>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>> DBG:core:receive_msg: preparing to run routing scripts...
>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>> DBG:core:parse_headers: flags=100
>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>> DBG:core:parse_to: end of header reached, state=10
>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>> DBG:core:parse_to: display={}, ruri={sip:1000 at 10.20.30.17}
>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>> DBG:core:get_hdr_field: <To> [24]; uri=[sip:1000 at 10.20.30.17]
>>  ]  16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>> DBG:core:get_hdr_field: to body [<sip:1000 at 10.20.30.17>
>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>> DBG:core:get_hdr_field: content_length=0
>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>> DBG:maxfwd:is_maxfwd_present: value = 70
>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>> DBG:uri:has_totag: no totag
>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>> DBG:core:parse_to_param: tag=70ea9ca0-e604-1910-80bc-002622a67db9
>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>> DBG:core:parse_to: end of header reached, state=29
>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>> DBG:core:parse_to: display={}, ruri={sip:1000 at 10.20.30.17}
>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>> DBG:dispatcher:ds_select_dst: set [1]
>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>> DBG:dispatcher:ds_select_dst: alg hash [1], id [1]
>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>> DBG:core:parse_headers: flags=ffffffffffffffff
>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>> DBG:core:get_hdr_field: found end of header
>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>> DBG:core:check_ip_address: params 72.237.213.162, 72.237.213.162, 0
>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>> DBG:core:destroy_avp_list: destroying list (nil)
>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>> DBG:core:receive_msg: cleaning up
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110819/00c4f085/attachment.html 

From tomasz at hyziak.pl  Fri Aug 19 17:03:52 2011
From: tomasz at hyziak.pl (Tomasz Hyziak)
Date: Fri, 19 Aug 2011 15:03:52 +0200
Subject: [Freeswitch-users] Cannot transfer incoming call
Message-ID: <CAFv7ub3brR6jJRfTXjqhZtFFmHOt6Zd1ZXtOU-pshvLNLF4Z_A@mail.gmail.com>

Hi

Transfers from local to local user work without problem, but when a
person call in via sip provider (selects call type via ivr and he is
put into fifo) and somebody answers this call, he is unable to
transfer this call to another person (via *1/*3/*4). When he presses
*, he is disconnected. Any ideas why ?

2011-08-19 14:55:17.475054 [DEBUG] switch_rtp.c:3303 RTP RECV DTMF *:800
2011-08-19 14:55:17.475054 [DEBUG] switch_channel.c:2775
(sofia/external/693xxxxxx at trunk.dialog.pl) Callstate Change ACTIVE ->
HANGUP


--
pozdrawiam - Tomasz Hyziak


From msc at freeswitch.org  Fri Aug 19 19:02:13 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 19 Aug 2011 08:02:13 -0700
Subject: [Freeswitch-users] INFO: Wiki update
In-Reply-To: <13889.1313715039@ccs.covici.com>
References: <CAKzWOxWAojX3A2vfA4+_+yB7niSsxnVbokcvnhTR2J8AVLe5hw@mail.gmail.com>
	<13889.1313715039@ccs.covici.com>
Message-ID: <CAKzWOxVTpPXk1hxG4BuL8gm6kSLCe3roCCPcNr8M9e2S8ANPmg@mail.gmail.com>

FYI,

I turned off the security. The good news is that no bots spammed us last
night. The bad news is that Google probably didn't do any indexing last
night. ;)

We are working on updating the wiki software. Hold tight...

Thanks,
MC

On Thu, Aug 18, 2011 at 5:50 PM, <covici at ccs.covici.com> wrote:

> I got the dialog just by going to wiki.freeswitch.org and the user name
> /password did not work -- but when I escaped out of the dialog, I was
> able to get in to search the wiki.  Very strange.
>
> Michael Collins <msc at freeswitch.org> wrote:
>
> > Hello all,
> >
> > We have been experimenting with some access controls on the wiki. If you
> get
> > a dialog box asking for a password then use the same credentials as you
> > would for pastebin.freeswitch.org. You should only see the dialog box
> when
> > logging in or editing pages. Just searching the wiki does not cause the
> > dialog to show up. We are trying to cut down on the spammers.
> >
> > Let me know if you run into any weirdness.
> >
> > Thanks,
> > MC
> >
> > ----------------------------------------------------
> > Alternatives:
> >
> > ----------------------------------------------------
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>         John Covici
>         covici at ccs.covici.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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110819/21881b43/attachment-0001.html 

From anthony.minessale at gmail.com  Fri Aug 19 19:02:50 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 19 Aug 2011 10:02:50 -0500
Subject: [Freeswitch-users] Original Caller ID/number after attended
	transfer
In-Reply-To: <5003D7D3E06F514E8C682F18D223265C04D3B36D76@AZWSMS03.azwarranty.int>
References: <5003D7D3E06F514E8C682F18D223265C04D3B36D6E@AZWSMS03.azwarranty.int>
	<CAKbxfG-6a26_NqN0UJ46sXXEbr2JHbF76kX5SUmkda7wgRWLZQ@mail.gmail.com>
	<5003D7D3E06F514E8C682F18D223265C04D3B36D76@AZWSMS03.azwarranty.int>
Message-ID: <CAKbxfG_GOZAbJD=zgciLwkFDpmZ7pwbzfdP2e_D6HTCwoP__=A@mail.gmail.com>

use set_profile_var on the inbound leg to set caller_id_name to the new
value.


On Fri, Aug 19, 2011 at 2:56 AM, Weigel, Stefan <
Stefan.Weigel at allianz-warranty.com> wrote:

>  Hi Anthony, list,****
>
> ** **
>
> we?re working with Polycom Soundpoint 560 phones. But meanwhile I could
> solve this problem.****
>
> A ?global_getvar? showed me that ?ignore_display_updates? was ?true?. After
> setting it to ?false? I now get the original caller ID displayed when doing
> a transfer.****
>
> ** **
>
> One point is left that I?m currently working on:****
>
> ** **
>
> on incoming calls I add a leading 0 to the original number. I need to do
> this because if a internal member wants to call outside he/she needs to dial
> with leading 0.****
>
> ** **
>
> [..]****
>
> ** **
>
>         <action application="export" data="caller_id_number=0$1"/>****
>
>         <action application="export"
> data="effective_caller_id_number=0$1"/>****
>
>         <action application="set" data="ignore_display_updates=true"/>****
>
> ** **
>
> [..]****
>
> ** **
>
> The initial incoming call is working, the number has a leading zero. Doing
> an attended transfer to a phone I now have the original number (without
> leading 0). With a blind transfer it?s working.****
>
> ** **
>
> Any suggestions ?****
>
> ** **
>
> ** **
>
> ** **
>
> Thanks in advance and best regards****
>
> ** **
>
> Stefan****
>
> ** **
>
> ** **
>
> *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Anthony
> Minessale
> *Gesendet:* Donnerstag, 18. August 2011 18:45
> *An:* FreeSWITCH Users Help
> *Betreff:* Re: [Freeswitch-users] Original Caller ID/number after attended
> transfer****
>
> ** **
>
> if you get phones that support display updates (polycom, snom, cisco and a
> few others) yes. It already works.****
>
> What phones do you have?****
>
> ** **
>
> On Thu, Aug 18, 2011 at 7:36 AM, Weigel, Stefan <
> Stefan.Weigel at allianz-warranty.com> wrote:****
>
> Hi all,****
>
>  ****
>
> is there a possibility to display the original caller ID & number after
> doing a attended transfer. External call to phone A -> calls phone B (I see
> caller ID & number of phone A) -> doing an attended transfer of external
> call to phone B (still caller ID & number of phone A).****
>
>  ****
>
>  ****
>
> Thanks in advance and best regards,****
>
>  ****
>
> Stefan****
>
>  ****
>
>  ****
>
> *Stefan Weigel*
> System Specialist AITP****
>
> ****
>
> *Allianz Automotive Services GmbH*****
>
> Einsteinring 28
> 85609 Aschheim
> Germany****
>
> Tel.: +49 89 2000 48 975
> Fax: +49 89 2000 48 566
> eMail: Stefan.Weigel at allianz-warranty.com ****
>
>
> http://www.allianz-warranty.com
> Gesch?ftsf?hrung: Andreas R?sing, Horst Ziegler
> Amtsgericht M?nchen, HRB 175682
> F?r Umsatzsteuerzwecke: Ust-ID-Nr.: DE 262 617 720****
>
>  ****
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org****
>
>
>
> ****
>
> ** **
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900****
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110819/5155ef04/attachment.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: image/gif
Size: 4415 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110819/5155ef04/attachment.gif 

From curriegrad2004 at gmail.com  Fri Aug 19 19:26:47 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Fri, 19 Aug 2011 08:26:47 -0700
Subject: [Freeswitch-users] INFO: Wiki update
In-Reply-To: <CAKzWOxVTpPXk1hxG4BuL8gm6kSLCe3roCCPcNr8M9e2S8ANPmg@mail.gmail.com>
References: <CAKzWOxWAojX3A2vfA4+_+yB7niSsxnVbokcvnhTR2J8AVLe5hw@mail.gmail.com>
	<13889.1313715039@ccs.covici.com>
	<CAKzWOxVTpPXk1hxG4BuL8gm6kSLCe3roCCPcNr8M9e2S8ANPmg@mail.gmail.com>
Message-ID: <CAMWiMy_5rCx9v7Uz+gv9DrvAu6xw=d-Cg5ee9Pxa7C1Hpwp=CQ@mail.gmail.com>

On Chrome the wiki security dialog box wouldn't shut up until I hit
the escape button!

On Fri, Aug 19, 2011 at 8:02 AM, Michael Collins <msc at freeswitch.org> wrote:
> FYI,
> I turned off the security. The good news is that no bots spammed us last
> night. The bad news is that Google probably didn't do any indexing last
> night. ;)
> We are working on updating the wiki software. Hold tight...
> Thanks,
> MC
>
> On Thu, Aug 18, 2011 at 5:50 PM, <covici at ccs.covici.com> wrote:
>>
>> I got the dialog just by going to wiki.freeswitch.org and the user name
>> /password did not work -- but when I escaped out of the dialog, I was
>> able to get in to search the wiki. ?Very strange.
>>
>> Michael Collins <msc at freeswitch.org> wrote:
>>
>> > Hello all,
>> >
>> > We have been experimenting with some access controls on the wiki. If you
>> > get
>> > a dialog box asking for a password then use the same credentials as you
>> > would for pastebin.freeswitch.org. You should only see the dialog box
>> > when
>> > logging in or editing pages. Just searching the wiki does not cause the
>> > dialog to show up. We are trying to cut down on the spammers.
>> >
>> > Let me know if you run into any weirdness.
>> >
>> > Thanks,
>> > MC
>> >
>> > ----------------------------------------------------
>> > Alternatives:
>> >
>> > ----------------------------------------------------
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> --
>> Your life is like a penny. ?You're going to lose it. ?The question is:
>> How do
>> you spend it?
>>
>> ? ? ? ? John Covici
>> ? ? ? ? covici at ccs.covici.com
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From x.liu at hw.ac.uk  Fri Aug 19 19:27:47 2011
From: x.liu at hw.ac.uk (xl127)
Date: Fri, 19 Aug 2011 16:27:47 +0100
Subject: [Freeswitch-users] mod_spidermonkey loading error
In-Reply-To: <CAKzWOxV6nM0dnar6kZThiakpv3gCpjzMAQXr6+P2WBuYncC8-Q@mail.gmail.com>
References: <20110817091244.5582.qmail@community37.interfree.it>	<4E4B9352.4010006@hw.ac.uk>	<CAKzWOxVyTnNMeHvxAWviHNM5MfPFW8XQoF_M0X=0ghzLv3Zhpw@mail.gmail.com>	<4E4D34D4.8000401@hw.ac.uk>
	<CAKzWOxV6nM0dnar6kZThiakpv3gCpjzMAQXr6+P2WBuYncC8-Q@mail.gmail.com>
Message-ID: <4E4E80F3.9060605@hw.ac.uk>

I am wondering which FS function will be affected by "--without-libcurl"

My simple dialplan tries to "speak" some text via mod_unmrcp. Exact same 
setups on both CentOS 5 and Fedora 14.
It works on CentOS but on Fedora I didn't hear anything, no sound 
outputs. The debug messages for the speechsythesizer and the mrcp_client
look normal from the console.

Is it because of the libcurl thing, or because of the FS version?

The CentOS FS version is:
FreeSWITCH Version 1.0.head (git-492bc6b 2011-07-23 12-53-04 -0400)

The Fedora FS version is quite new:
FreeSWITCH Version 1.0.head (git-cd31633 2011-08-17 19-34-22 -0500)

Cheers,
Xing



On 18/08/11 17:34, Michael Collins wrote:
>
>
> On Thu, Aug 18, 2011 at 8:50 AM, xl127 <x.liu at hw.ac.uk 
> <mailto:x.liu at hw.ac.uk>> wrote:
>
>     It works on Fedora 14 now, thanks Michael!
>
>     I am wondering if I need to give the flag "--without-libcurl"
>     every time when I add a new module to FS and do "configure"?
>
>
> Only when you need to actually re-run the configure script. Most of 
> the time when you do "git pull && make install" or "make current" 
> you're fine.
>
> -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



-- 
Heriot-Watt University is a Scottish charity
registered under charity number SC000278.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110819/a88b2901/attachment-0001.html 

From msc at freeswitch.org  Fri Aug 19 19:38:06 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 19 Aug 2011 08:38:06 -0700
Subject: [Freeswitch-users] mod_spidermonkey loading error
In-Reply-To: <4E4E80F3.9060605@hw.ac.uk>
References: <20110817091244.5582.qmail@community37.interfree.it>
	<4E4B9352.4010006@hw.ac.uk>
	<CAKzWOxVyTnNMeHvxAWviHNM5MfPFW8XQoF_M0X=0ghzLv3Zhpw@mail.gmail.com>
	<4E4D34D4.8000401@hw.ac.uk>
	<CAKzWOxV6nM0dnar6kZThiakpv3gCpjzMAQXr6+P2WBuYncC8-Q@mail.gmail.com>
	<4E4E80F3.9060605@hw.ac.uk>
Message-ID: <CAKzWOxXwBjpz1oToGMS6F8OniSNMciOx3XQZGTzKNHgN1hzTCg@mail.gmail.com>

I don't know that there's anything affected. I have been using FS on a
CentOS 6 machine and none of the modules built w/o libcurl have had any
adverse affects.

-MC

On Fri, Aug 19, 2011 at 8:27 AM, xl127 <x.liu at hw.ac.uk> wrote:

> **
> I am wondering which FS function will be affected by "--without-libcurl"
>
> My simple dialplan tries to "speak" some text via mod_unmrcp. Exact same
> setups on both CentOS 5 and Fedora 14.
> It works on CentOS but on Fedora I didn't hear anything, no sound outputs.
> The debug messages for the speechsythesizer and the mrcp_client
> look normal from the console.
>
> Is it because of the libcurl thing, or because of the FS version?
>
> The CentOS FS version is:
> FreeSWITCH Version 1.0.head (git-492bc6b 2011-07-23 12-53-04 -0400)
>
> The Fedora FS version is quite new:
> FreeSWITCH Version 1.0.head (git-cd31633 2011-08-17 19-34-22 -0500)
>
> Cheers,
> Xing
>
>
>
>
> On 18/08/11 17:34, Michael Collins wrote:
>
>
>
> On Thu, Aug 18, 2011 at 8:50 AM, xl127 <x.liu at hw.ac.uk> wrote:
>
>>  It works on Fedora 14 now, thanks Michael!
>>
>> I am wondering if I need to give the flag "--without-libcurl" every time
>> when I add a new module to FS and do "configure"?
>>
>
>  Only when you need to actually re-run the configure script. Most of the
> time when you do "git pull && make install" or "make current" you're fine.
>
>  -MC
>
>
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> ------------------------------
>  Heriot-Watt University is a Scottish charity registered under charity
> number SC000278.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110819/a2cc3bb7/attachment.html 

From msc at freeswitch.org  Fri Aug 19 19:39:25 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 19 Aug 2011 08:39:25 -0700
Subject: [Freeswitch-users] Fetching user-variables
In-Reply-To: <CAAMQ8bRTgb_Dxn_L=C4KEeoNM+WTsy6hJKdQp7d7EjdarLqBjA@mail.gmail.com>
References: <CAAMQ8bRTgb_Dxn_L=C4KEeoNM+WTsy6hJKdQp7d7EjdarLqBjA@mail.gmail.com>
Message-ID: <CAKzWOxVrgz6RuOAoGY1jSx_SZG74mRJBhzjGTSfHYNyc2-JgjA@mail.gmail.com>

You probably need to add a "set_user" dp app in there somewhere. Can you
pastebin your relevant dp configs? We'll take a look.

-MC

On Fri, Aug 19, 2011 at 6:11 AM, Christian Benke <benkokakao at gmail.com>wrote:

> Hi!
>
> I'm working on a redirect script and ran into a problem with toll_allow:
> A tries to reach B but B has redirected to C
> B is allowed to make external calls(via the user-specific-variable
> "toll_allow"), but A is not(Could be a external caller or a
> directory-user with restrictions).
> B's redirection will not work as she intended, as the dialplan is
> restricted through toll_allow.
> B is responsible for the external call of A to C, therefore her
> accountcode and her toll_allow should apply - unfortunately the
> dialplan doesn't know that the call has been redirected by B.
> Right now i don't know how to solve this with FreeSWITCH.
>
> Is there a way to fetch user-specific-variables of B, like
> "toll_allow", in a call that was not directly initiated by B?
>
> Best regards
> Christian
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110819/6cead812/attachment.html 

From msc at freeswitch.org  Fri Aug 19 19:43:15 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 19 Aug 2011 08:43:15 -0700
Subject: [Freeswitch-users] enterprise deployment not working
In-Reply-To: <CAC-LwPMAT1zsohk-hCSwpSLetui8-_figFGjDxztB3Z9FFxfZg@mail.gmail.com>
References: <CAC-LwPPSAei6EyeQBtrHoin0DtN73f2zfd6GU5JyQm0Z8_rXPw@mail.gmail.com>
	<CAC-LwPP+oNWNu-5=-Y2KQwGkn0qaaf7Bq59CcXTo_Ura+AZwBA@mail.gmail.com>
	<CAC-LwPMAT1zsohk-hCSwpSLetui8-_figFGjDxztB3Z9FFxfZg@mail.gmail.com>
Message-ID: <CAKzWOxUsDLEkN9qfsgiOf11F5M-3077AT8U_8Rc_fxNJ2-=HyQ@mail.gmail.com>

Perhaps you just need to wrap your scripts and config files in <code></code>
tags. You can also use <nowiki></nowiki> if you have text that is causing
strange wiki features to be turned on.

-MC

On Fri, Aug 19, 2011 at 6:54 AM, vip killa <vipkilla at gmail.com> wrote:

> I managed to get this working and i wanted to contribute how... i edited
> http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS#OpenSIPS
> but the format is all screwed up and i'd like to add an init.d startup
> script
> perhaps someone could format that wiki page better and add an init.d
> script?
>
> On Tue, Aug 16, 2011 at 3:22 PM, vip killa <vipkilla at gmail.com> wrote:
>
>> I did an ngrep trace and opensips is returning:
>>
>> SIP/2.0 503 Service Unavailable
>> CSeq: 54 REGISTER
>>
>> before it even tries the FS server...
>> perhaps this wiki page needs to be updated...
>>
>>
>> On Tue, Aug 16, 2011 at 2:03 PM, vip killa <vipkilla at gmail.com> wrote:
>>
>>> I'm following tutorial at
>>> http://wiki.freeswitch.org/wiki/Enterprise_deployment_<http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS>
>>> OpenSIPS<http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS> and
>>> opensips is not forwarding the registration requests...
>>> Running opensips-1.6.4-2-tls,
>>> Here is what the debugging looks like when i try to register with the
>>> opensip box
>>> 10.20.30.17 is the opensips box.
>>> 10.20.30.18 is the FreeSWITCH box.
>>>
>>>
>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>> DBG:core:parse_msg: SIP Request:
>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>> DBG:core:parse_msg:  method:  <REGISTER>
>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>> DBG:core:parse_msg:  uri:     <sip:10.20.30.17>
>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>> DBG:core:parse_msg:  version: <SIP/2.0>
>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>> DBG:core:parse_headers: flags=2
>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>> DBG:core:get_hdr_field: cseq <CSeq>: <43> <REGISTER>
>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>> DBG:core:parse_via_param: found param type 232, <branch> =
>>> <z9hG4bKda029da0-e604-1910-80bc-002622a67db9>; state=6
>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>> DBG:core:parse_via_param: found param type 235, <rport> = <n/a>; state=17
>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>> DBG:core:parse_via: end of header reached, state=5
>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>> DBG:core:parse_headers: via found, flags=2
>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>> DBG:core:parse_headers: this is the first via
>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>> DBG:core:receive_msg: After parse_msg...
>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>> DBG:core:receive_msg: preparing to run routing scripts...
>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>> DBG:core:parse_headers: flags=100
>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>> DBG:core:parse_to: end of header reached, state=10
>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>> DBG:core:parse_to: display={}, ruri={sip:1000 at 10.20.30.17}
>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>> DBG:core:get_hdr_field: <To> [24]; uri=[sip:1000 at 10.20.30.17]
>>>  ]  16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>> DBG:core:get_hdr_field: to body [<sip:1000 at 10.20.30.17>
>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>> DBG:core:get_hdr_field: content_length=0
>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>> DBG:maxfwd:is_maxfwd_present: value = 70
>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>> DBG:uri:has_totag: no totag
>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>> DBG:core:parse_to_param: tag=70ea9ca0-e604-1910-80bc-002622a67db9
>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>> DBG:core:parse_to: end of header reached, state=29
>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>> DBG:core:parse_to: display={}, ruri={sip:1000 at 10.20.30.17}
>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>> DBG:dispatcher:ds_select_dst: set [1]
>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>> DBG:dispatcher:ds_select_dst: alg hash [1], id [1]
>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>> DBG:core:parse_headers: flags=ffffffffffffffff
>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>> DBG:core:get_hdr_field: found end of header
>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>> DBG:core:check_ip_address: params 72.237.213.162, 72.237.213.162, 0
>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>> DBG:core:destroy_avp_list: destroying list (nil)
>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>> DBG:core:receive_msg: cleaning 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
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110819/fe8a2578/attachment.html 

From cjbujold at accra.ca  Fri Aug 19 20:44:34 2011
From: cjbujold at accra.ca (Charles Bujold)
Date: Fri, 19 Aug 2011 13:44:34 -0300
Subject: [Freeswitch-users] Newbie - external originating leg has no sound
	when calling
Message-ID: <009001cc5e8f$46cc9a50$d465cef0$@accra.ca>

 

Weird situation, we have an external location which if they originate the
call we cannot hear them, but if we originate the call to them everything
works fine.  

 

The configuration we have is Sense as firewall that sends and has all
stipulated Freeswitch ports open and forwards incoming ports to the
Freeswitch server (Ubuntu/Freeswitch/fusionpbx)

 

Internal  to internal call works

Internal call to the external location works  Both parties can talk without
issues.

External location to Internal location:  we do not hear them, but they can
hear us.

External location call to external location :  the originating caller cannot
be heard, the second leg works perfectly. (This suggest that it is not a
firewall issue but a configuration issue.)

External location calls to the voicemail and can record without problems a
recording.

 

The problem seems to be limited to the first leg of the call, in that if it
originates from an external location the first leg cannot be heard.

 

 

Any suggestions on how to fix would be appreciated.  Thanks!

 

Cjb

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110819/e0f92a9e/attachment-0001.html 

From x.liu at hw.ac.uk  Fri Aug 19 21:51:03 2011
From: x.liu at hw.ac.uk (xl127)
Date: Fri, 19 Aug 2011 18:51:03 +0100
Subject: [Freeswitch-users] mod_spidermonkey loading error
In-Reply-To: <CAKzWOxXwBjpz1oToGMS6F8OniSNMciOx3XQZGTzKNHgN1hzTCg@mail.gmail.com>
References: <20110817091244.5582.qmail@community37.interfree.it>	<4E4B9352.4010006@hw.ac.uk>	<CAKzWOxVyTnNMeHvxAWviHNM5MfPFW8XQoF_M0X=0ghzLv3Zhpw@mail.gmail.com>	<4E4D34D4.8000401@hw.ac.uk>	<CAKzWOxV6nM0dnar6kZThiakpv3gCpjzMAQXr6+P2WBuYncC8-Q@mail.gmail.com>	<4E4E80F3.9060605@hw.ac.uk>
	<CAKzWOxXwBjpz1oToGMS6F8OniSNMciOx3XQZGTzKNHgN1hzTCg@mail.gmail.com>
Message-ID: <4E4EA287.6020701@hw.ac.uk>

okay, thanks!

I tried to git the latest version on both CentOS and Fedora. It works 
fine on CentOS but still no audio out for mod_unimrcp on Fedora. All 
firewalls were disabled.
The original pizza demo which uses streamFile from audio files works 
file on Fedora.

So there must be something wrong with FS-UniMRCP on Fedora 14.

Any suggestions about how to find the causes?

Thanks!

Xing


On 19/08/11 16:38, Michael Collins wrote:
> I don't know that there's anything affected. I have been using FS on a 
> CentOS 6 machine and none of the modules built w/o libcurl have had 
> any adverse affects.
>
> -MC
>
> On Fri, Aug 19, 2011 at 8:27 AM, xl127 <x.liu at hw.ac.uk 
> <mailto:x.liu at hw.ac.uk>> wrote:
>
>     I am wondering which FS function will be affected by
>     "--without-libcurl"
>
>     My simple dialplan tries to "speak" some text via mod_unmrcp.
>     Exact same setups on both CentOS 5 and Fedora 14.
>     It works on CentOS but on Fedora I didn't hear anything, no sound
>     outputs. The debug messages for the speechsythesizer and the
>     mrcp_client
>     look normal from the console.
>
>     Is it because of the libcurl thing, or because of the FS version?
>
>     The CentOS FS version is:
>     FreeSWITCH Version 1.0.head (git-492bc6b 2011-07-23 12-53-04 -0400)
>
>     The Fedora FS version is quite new:
>     FreeSWITCH Version 1.0.head (git-cd31633 2011-08-17 19-34-22 -0500)
>
>     Cheers,
>     Xing
>
>
>
>
>     On 18/08/11 17:34, Michael Collins wrote:
>>
>>
>>     On Thu, Aug 18, 2011 at 8:50 AM, xl127 <x.liu at hw.ac.uk
>>     <mailto:x.liu at hw.ac.uk>> wrote:
>>
>>         It works on Fedora 14 now, thanks Michael!
>>
>>         I am wondering if I need to give the flag "--without-libcurl"
>>         every time when I add a new module to FS and do "configure"?
>>
>>
>>     Only when you need to actually re-run the configure script. Most
>>     of the time when you do "git pull && make install" or "make
>>     current" you're fine.
>>
>>     -MC
>>
>>
>>
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>
>     ------------------------------------------------------------------------
>     Heriot-Watt University is a Scottish charity registered under
>     charity number SC000278.
>
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Heriot-Watt University is a Scottish charity
registered under charity number SC000278.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110819/0db865f1/attachment.html 

From jakub.ouhrabka at gmail.com  Fri Aug 19 21:55:18 2011
From: jakub.ouhrabka at gmail.com (Jakub Ouhrabka)
Date: Fri, 19 Aug 2011 19:55:18 +0200
Subject: [Freeswitch-users] Freetdm bridge to SIP DTMF problem
In-Reply-To: <4E4E4653.3090309@gmail.com>
References: <4E4CD94C.3030403@comgate.cz> <4E4CEEB1.3040202@gmail.com>
	<4E4D0B5F.2060004@gmail.com> <4E4E4653.3090309@gmail.com>
Message-ID: <4E4EA386.2060300@gmail.com>

For the archives: <param name="rfc2833-pt" value="101"/> in the SIP 
profile solved my issue. In git master this is the default now.

Another interesting param/var is sip_liberal_dtmf=true which can make 
dtmf working against non complaint sip implementation.

Regards,

Jakub

Dne 19.8.2011 13:17, Jakub Ouhrabka napsal(a):
> Hi,
>
> another self follow-up: It seems that I have problem even when bridging
> SIP channel to SIP channel - DTMF is not sent to the called party. So I
> guess it's not Sangoma/Wanpipe/libpri/FreeTDM related. DTMF is correctly
> decoded but not forwarded (according to tcpdump/wireshark).
>
>  From the log:
>
> 2011-08-19 11:12:41.757636 [DEBUG] switch_rtp.c:3302 RTP RECV DTMF 5:880
> 2011-08-19 11:12:41.757636 [DEBUG] switch_ivr_bridge.c:391 Send signal
> sofia/external/XXXXXX at XXXXX.XX:5060 [BREAK]
>
> It must some stupid error in configuration - is there anything I have to
> set in SIP profile, or in bridge or somewhere else? Or must the called
> party somehow advertise that it is capable of receiving DTMF?
>
> Thanks,
>
> Jakub
>
> Dne 18.8.2011 14:53, Jakub Ouhrabka napsal(a):
>> Hi,
>>
>> I've tried further: both dtmf-type info and rfc2833 are not working
>> (can't see it in packet dump). Only dtmf-type set to "none" together
>> with start_dtmf_generate is working as expected for me - dtmf is
>> generated inband.
>>
>> How to force freeswitch to forward dtmf from freetdm to sofia preferably
>> using rfc2833?
>>
>> Thank you for any hints,
>>
>> Jakub
>>
>> Dne 18.8.2011 12:51, Jakub Ouhrabka napsal(a):
>>> Hi,
>>>
>>> we've changed OpenZAP to FreeTDM sometime ago and from that time we're
>>> experiencing problems with DTMF.
>>>
>>> We're bridging incoming ISDN30 calls from Sangoma A108DE card + libri +
>>> FreeTDM to SIP outgoing calls. Freeswitch is recognizing DTMF correctly
>>> but DTMF is not send to outgoing SIP call - it's not present in RTP
>>> stream trace (we've setup RFC2833 DTMF signalling). When bridging the
>>> call back to ISDN30 DTMF works correctly.
>>>
>>> We're using latest versions of all software as of 1 month ago.
>>>
>>> Below are attached two snippets of logs: first is bridging back to
>>> ISDN30 where DTMF is sent to called party. Second is bridging to SIP
>>> where DTMF is not sent to the called party.
>>>
>>> We've tried it with both <action application="start_dtmf" /> on and off
>>> with no success.
>>>
>>> Any pointers how to investigate the issue?
>>>
>>> Thanks,
>>>
>>> Jakub
>>>
>>> Log snippets:
>>>
>>> ISDN30 to ISDN30 bridge - DTMF ok
>>>
>>> 2011-08-18 12:31:45.950158 [DEBUG] ftmod_wanpipe.c:1545 [s1c11][1:11]
>>> read wanpipe event 3
>>> 2011-08-18 12:31:45.950158 [DEBUG] ftmod_wanpipe.c:1415 [s1c11][1:11]
>>> Queuing wanpipe DTMF: 5
>>> 2011-08-18 12:31:45.950158 [DEBUG] ftdm_io.c:3504 [s1c11][1:11] Queuing
>>> DTMF 5 (debug = 0)
>>> 2011-08-18 12:31:45.950158 [DEBUG] mod_freetdm.c:733 Queuing DTMF [5] in
>>> channel FreeTDM/1:11/XXXXXXXXX
>>> 2011-08-18 12:31:45.970623 [DEBUG] switch_ivr_bridge.c:391 Send signal
>>> FreeTDM/1:5/XXXXXXXXX [BREAK]
>>> 2011-08-18 12:31:45.991084 [DEBUG] ftdm_io.c:3694 [s1c5][1:5] Generating
>>> DTMF [5]
>>>
>>> ISDN30 to SIP bridge - DTMF not forwarded
>>>
>>> 2011-08-18 12:34:05.287480 [DEBUG] ftmod_wanpipe.c:1545 [s1c29][1:29]
>>> read wanpipe event 3
>>> 2011-08-18 12:34:05.607708 [DEBUG] ftmod_wanpipe.c:1545 [s1c29][1:29]
>>> read wanpipe event 3
>>> 2011-08-18 12:34:05.607708 [DEBUG] ftmod_wanpipe.c:1415 [s1c29][1:29]
>>> Queuing wanpipe DTMF: 6
>>> 2011-08-18 12:34:05.607708 [DEBUG] ftdm_io.c:3504 [s1c29][1:29] Queuing
>>> DTMF 6 (debug = 0)
>>> 2011-08-18 12:34:05.607708 [DEBUG] ftmod_libpri.c:1586 -- Caught Event
>>> span 1 18 (KEYPAD_DIGIT)
>>> 2011-08-18 12:34:05.628171 [DEBUG] mod_freetdm.c:733 Queuing DTMF [6] in
>>> channel FreeTDM/1:29/XXXXXXXXXXXXXXX
>>> 2011-08-18 12:34:05.628171 [DEBUG] switch_ivr_bridge.c:391 Send signal
>>> sofia/external/XXXXXXX at XXXXXXX:5060 [BREAK]
>>>


From vipkilla at gmail.com  Fri Aug 19 22:04:55 2011
From: vipkilla at gmail.com (vip killa)
Date: Fri, 19 Aug 2011 14:04:55 -0400
Subject: [Freeswitch-users] enterprise deployment not working
In-Reply-To: <CAKzWOxUsDLEkN9qfsgiOf11F5M-3077AT8U_8Rc_fxNJ2-=HyQ@mail.gmail.com>
References: <CAC-LwPPSAei6EyeQBtrHoin0DtN73f2zfd6GU5JyQm0Z8_rXPw@mail.gmail.com>
	<CAC-LwPP+oNWNu-5=-Y2KQwGkn0qaaf7Bq59CcXTo_Ura+AZwBA@mail.gmail.com>
	<CAC-LwPMAT1zsohk-hCSwpSLetui8-_figFGjDxztB3Z9FFxfZg@mail.gmail.com>
	<CAKzWOxUsDLEkN9qfsgiOf11F5M-3077AT8U_8Rc_fxNJ2-=HyQ@mail.gmail.com>
Message-ID: <CAC-LwPPfpNz4D=1_B21hxPjuC5GP0nG6X9G6ByRjv3Jc7Z9BxQ@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS#OpenSIPS
Done. Hope that's not too much, i tried to make it as complete as possible.


On Fri, Aug 19, 2011 at 11:43 AM, Michael Collins <msc at freeswitch.org>wrote:

> Perhaps you just need to wrap your scripts and config files in
> <code></code> tags. You can also use <nowiki></nowiki> if you have text that
> is causing strange wiki features to be turned on.
>
> -MC
>
> On Fri, Aug 19, 2011 at 6:54 AM, vip killa <vipkilla at gmail.com> wrote:
>
>> I managed to get this working and i wanted to contribute how... i edited
>> http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS#OpenSIPS
>> but the format is all screwed up and i'd like to add an init.d startup
>> script
>> perhaps someone could format that wiki page better and add an init.d
>> script?
>>
>> On Tue, Aug 16, 2011 at 3:22 PM, vip killa <vipkilla at gmail.com> wrote:
>>
>>> I did an ngrep trace and opensips is returning:
>>>
>>> SIP/2.0 503 Service Unavailable
>>> CSeq: 54 REGISTER
>>>
>>> before it even tries the FS server...
>>> perhaps this wiki page needs to be updated...
>>>
>>>
>>> On Tue, Aug 16, 2011 at 2:03 PM, vip killa <vipkilla at gmail.com> wrote:
>>>
>>>> I'm following tutorial at
>>>> http://wiki.freeswitch.org/wiki/Enterprise_deployment_<http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS>
>>>> OpenSIPS<http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS> and
>>>> opensips is not forwarding the registration requests...
>>>> Running opensips-1.6.4-2-tls,
>>>> Here is what the debugging looks like when i try to register with the
>>>> opensip box
>>>> 10.20.30.17 is the opensips box.
>>>> 10.20.30.18 is the FreeSWITCH box.
>>>>
>>>>
>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>> DBG:core:parse_msg: SIP Request:
>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>> DBG:core:parse_msg:  method:  <REGISTER>
>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>> DBG:core:parse_msg:  uri:     <sip:10.20.30.17>
>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>> DBG:core:parse_msg:  version: <SIP/2.0>
>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>> DBG:core:parse_headers: flags=2
>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>> DBG:core:get_hdr_field: cseq <CSeq>: <43> <REGISTER>
>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>> DBG:core:parse_via_param: found param type 232, <branch> =
>>>> <z9hG4bKda029da0-e604-1910-80bc-002622a67db9>; state=6
>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>> DBG:core:parse_via_param: found param type 235, <rport> = <n/a>; state=17
>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>> DBG:core:parse_via: end of header reached, state=5
>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>> DBG:core:parse_headers: via found, flags=2
>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>> DBG:core:parse_headers: this is the first via
>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>> DBG:core:receive_msg: After parse_msg...
>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>> DBG:core:receive_msg: preparing to run routing scripts...
>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>> DBG:core:parse_headers: flags=100
>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>> DBG:core:parse_to: end of header reached, state=10
>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>> DBG:core:parse_to: display={}, ruri={sip:1000 at 10.20.30.17}
>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>> DBG:core:get_hdr_field: <To> [24]; uri=[sip:1000 at 10.20.30.17]
>>>>  ]  16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>> DBG:core:get_hdr_field: to body [<sip:1000 at 10.20.30.17>
>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>> DBG:core:get_hdr_field: content_length=0
>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>> DBG:maxfwd:is_maxfwd_present: value = 70
>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>> DBG:uri:has_totag: no totag
>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>> DBG:core:parse_to_param: tag=70ea9ca0-e604-1910-80bc-002622a67db9
>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>> DBG:core:parse_to: end of header reached, state=29
>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>> DBG:core:parse_to: display={}, ruri={sip:1000 at 10.20.30.17}
>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>> DBG:dispatcher:ds_select_dst: set [1]
>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>> DBG:dispatcher:ds_select_dst: alg hash [1], id [1]
>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>> DBG:core:parse_headers: flags=ffffffffffffffff
>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>> DBG:core:get_hdr_field: found end of header
>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>> DBG:core:check_ip_address: params 72.237.213.162, 72.237.213.162, 0
>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>> DBG:core:destroy_avp_list: destroying list (nil)
>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>> DBG:core:receive_msg: cleaning 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
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110819/75ad08c0/attachment-0001.html 

From msc at freeswitch.org  Fri Aug 19 22:26:56 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 19 Aug 2011 11:26:56 -0700
Subject: [Freeswitch-users] enterprise deployment not working
In-Reply-To: <CAC-LwPPfpNz4D=1_B21hxPjuC5GP0nG6X9G6ByRjv3Jc7Z9BxQ@mail.gmail.com>
References: <CAC-LwPPSAei6EyeQBtrHoin0DtN73f2zfd6GU5JyQm0Z8_rXPw@mail.gmail.com>
	<CAC-LwPP+oNWNu-5=-Y2KQwGkn0qaaf7Bq59CcXTo_Ura+AZwBA@mail.gmail.com>
	<CAC-LwPMAT1zsohk-hCSwpSLetui8-_figFGjDxztB3Z9FFxfZg@mail.gmail.com>
	<CAKzWOxUsDLEkN9qfsgiOf11F5M-3077AT8U_8Rc_fxNJ2-=HyQ@mail.gmail.com>
	<CAC-LwPPfpNz4D=1_B21hxPjuC5GP0nG6X9G6ByRjv3Jc7Z9BxQ@mail.gmail.com>
Message-ID: <CAKzWOxW2scFwq3M7d1J_0y-wZ9+1_qW9X=dU3UGfBavTuS0yTw@mail.gmail.com>

in your example you used non-RFC1918 IP addresses, i.e. public addrs. Was
that by design? Just curious.

-MC

On Fri, Aug 19, 2011 at 11:04 AM, vip killa <vipkilla at gmail.com> wrote:

> http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS#OpenSIPS
> Done. Hope that's not too much, i tried to make it as complete as possible.
>
>
> On Fri, Aug 19, 2011 at 11:43 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Perhaps you just need to wrap your scripts and config files in
>> <code></code> tags. You can also use <nowiki></nowiki> if you have text that
>> is causing strange wiki features to be turned on.
>>
>> -MC
>>
>> On Fri, Aug 19, 2011 at 6:54 AM, vip killa <vipkilla at gmail.com> wrote:
>>
>>> I managed to get this working and i wanted to contribute how... i edited
>>> http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS#OpenSIPS
>>> but the format is all screwed up and i'd like to add an init.d startup
>>> script
>>> perhaps someone could format that wiki page better and add an init.d
>>> script?
>>>
>>> On Tue, Aug 16, 2011 at 3:22 PM, vip killa <vipkilla at gmail.com> wrote:
>>>
>>>> I did an ngrep trace and opensips is returning:
>>>>
>>>> SIP/2.0 503 Service Unavailable
>>>> CSeq: 54 REGISTER
>>>>
>>>> before it even tries the FS server...
>>>> perhaps this wiki page needs to be updated...
>>>>
>>>>
>>>> On Tue, Aug 16, 2011 at 2:03 PM, vip killa <vipkilla at gmail.com> wrote:
>>>>
>>>>> I'm following tutorial at
>>>>> http://wiki.freeswitch.org/wiki/Enterprise_deployment_<http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS>
>>>>> OpenSIPS<http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS> and
>>>>> opensips is not forwarding the registration requests...
>>>>> Running opensips-1.6.4-2-tls,
>>>>> Here is what the debugging looks like when i try to register with the
>>>>> opensip box
>>>>> 10.20.30.17 is the opensips box.
>>>>> 10.20.30.18 is the FreeSWITCH box.
>>>>>
>>>>>
>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>> DBG:core:parse_msg: SIP Request:
>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>> DBG:core:parse_msg:  method:  <REGISTER>
>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>> DBG:core:parse_msg:  uri:     <sip:10.20.30.17>
>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>> DBG:core:parse_msg:  version: <SIP/2.0>
>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>> DBG:core:parse_headers: flags=2
>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>> DBG:core:get_hdr_field: cseq <CSeq>: <43> <REGISTER>
>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>> DBG:core:parse_via_param: found param type 232, <branch> =
>>>>> <z9hG4bKda029da0-e604-1910-80bc-002622a67db9>; state=6
>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>> DBG:core:parse_via_param: found param type 235, <rport> = <n/a>; state=17
>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>> DBG:core:parse_via: end of header reached, state=5
>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>> DBG:core:parse_headers: via found, flags=2
>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>> DBG:core:parse_headers: this is the first via
>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>> DBG:core:receive_msg: After parse_msg...
>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>> DBG:core:receive_msg: preparing to run routing scripts...
>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>> DBG:core:parse_headers: flags=100
>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>> DBG:core:parse_to: end of header reached, state=10
>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>> DBG:core:parse_to: display={}, ruri={sip:1000 at 10.20.30.17}
>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>> DBG:core:get_hdr_field: <To> [24]; uri=[sip:1000 at 10.20.30.17]
>>>>>  ]  16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>> DBG:core:get_hdr_field: to body [<sip:1000 at 10.20.30.17>
>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>> DBG:core:get_hdr_field: content_length=0
>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>> DBG:maxfwd:is_maxfwd_present: value = 70
>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>> DBG:uri:has_totag: no totag
>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>> DBG:core:parse_to_param: tag=70ea9ca0-e604-1910-80bc-002622a67db9
>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>> DBG:core:parse_to: end of header reached, state=29
>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>> DBG:core:parse_to: display={}, ruri={sip:1000 at 10.20.30.17}
>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>> DBG:dispatcher:ds_select_dst: set [1]
>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>> DBG:dispatcher:ds_select_dst: alg hash [1], id [1]
>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>> DBG:core:parse_headers: flags=ffffffffffffffff
>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>> DBG:core:get_hdr_field: found end of header
>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>> DBG:core:check_ip_address: params 72.237.213.162, 72.237.213.162, 0
>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>> DBG:core:destroy_avp_list: destroying list (nil)
>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>> DBG:core:receive_msg: cleaning 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
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110819/17fbc5f8/attachment.html 

From msc at freeswitch.org  Fri Aug 19 22:27:47 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 19 Aug 2011 11:27:47 -0700
Subject: [Freeswitch-users] mod_spidermonkey loading error
In-Reply-To: <4E4EA287.6020701@hw.ac.uk>
References: <20110817091244.5582.qmail@community37.interfree.it>
	<4E4B9352.4010006@hw.ac.uk>
	<CAKzWOxVyTnNMeHvxAWviHNM5MfPFW8XQoF_M0X=0ghzLv3Zhpw@mail.gmail.com>
	<4E4D34D4.8000401@hw.ac.uk>
	<CAKzWOxV6nM0dnar6kZThiakpv3gCpjzMAQXr6+P2WBuYncC8-Q@mail.gmail.com>
	<4E4E80F3.9060605@hw.ac.uk>
	<CAKzWOxXwBjpz1oToGMS6F8OniSNMciOx3XQZGTzKNHgN1hzTCg@mail.gmail.com>
	<4E4EA287.6020701@hw.ac.uk>
Message-ID: <CAKzWOxXmBcCLQXnjAEXmZYk+TVps5VSMOFsmbCtr2doO7v_fVA@mail.gmail.com>

Open a jira if you haven't already done so and hopefully CRienzo can have a
look.

-MC

On Fri, Aug 19, 2011 at 10:51 AM, xl127 <x.liu at hw.ac.uk> wrote:

> **
> okay, thanks!
>
> I tried to git the latest version on both CentOS and Fedora. It works fine
> on CentOS but still no audio out for mod_unimrcp on Fedora. All firewalls
> were disabled.
> The original pizza demo which uses streamFile from audio files works file
> on Fedora.
>
> So there must be something wrong with FS-UniMRCP on Fedora 14.
>
> Any suggestions about how to find the causes?
>
> Thanks!
>
> Xing
>
>
>
> On 19/08/11 16:38, Michael Collins wrote:
>
> I don't know that there's anything affected. I have been using FS on a
> CentOS 6 machine and none of the modules built w/o libcurl have had any
> adverse affects.
>
>  -MC
>
>  On Fri, Aug 19, 2011 at 8:27 AM, xl127 <x.liu at hw.ac.uk> wrote:
>
>>  I am wondering which FS function will be affected by "--without-libcurl"
>>
>> My simple dialplan tries to "speak" some text via mod_unmrcp. Exact same
>> setups on both CentOS 5 and Fedora 14.
>> It works on CentOS but on Fedora I didn't hear anything, no sound outputs.
>> The debug messages for the speechsythesizer and the mrcp_client
>> look normal from the console.
>>
>> Is it because of the libcurl thing, or because of the FS version?
>>
>> The CentOS FS version is:
>> FreeSWITCH Version 1.0.head (git-492bc6b 2011-07-23 12-53-04 -0400)
>>
>> The Fedora FS version is quite new:
>> FreeSWITCH Version 1.0.head (git-cd31633 2011-08-17 19-34-22 -0500)
>>
>> Cheers,
>> Xing
>>
>>
>>
>>
>> On 18/08/11 17:34, Michael Collins wrote:
>>
>>
>>
>> On Thu, Aug 18, 2011 at 8:50 AM, xl127 <x.liu at hw.ac.uk> wrote:
>>
>>>  It works on Fedora 14 now, thanks Michael!
>>>
>>> I am wondering if I need to give the flag "--without-libcurl" every time
>>> when I add a new module to FS and do "configure"?
>>>
>>
>>  Only when you need to actually re-run the configure script. Most of the
>> time when you do "git pull && make install" or "make current" you're fine.
>>
>>  -MC
>>
>>
>>
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>>
>> ------------------------------
>>  Heriot-Watt University is a Scottish charity registered under charity
>> number SC000278.
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> ------------------------------
>  Heriot-Watt University is a Scottish charity registered under charity
> number SC000278.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110819/9d7c70b9/attachment-0001.html 

From msc at freeswitch.org  Fri Aug 19 22:31:57 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 19 Aug 2011 11:31:57 -0700
Subject: [Freeswitch-users] Newbie - external originating leg has no
 sound when calling
In-Reply-To: <009001cc5e8f$46cc9a50$d465cef0$@accra.ca>
References: <009001cc5e8f$46cc9a50$d465cef0$@accra.ca>
Message-ID: <CAKzWOxXaTbFA3oN2eis7zwmshPBfxDS8c001MQN4t-ZM-UDrCA@mail.gmail.com>

This is almost assuredly a NAT traversal issue. Get a pcap of the SIP
traffic and look at it in Wireshark and/or post on pastebin.freeswitch.org.
I'm sure that the gang here can help you figure out what's happening. Also,
include the model of phone you are using. Some work better than others in
this situation.

-MC

On Fri, Aug 19, 2011 at 9:44 AM, Charles Bujold <cjbujold at accra.ca> wrote:

> ** **
>
> Weird situation, we have an external location which if they originate the
> call we cannot hear them, but if we originate the call to them everything
> works fine.  ****
>
> ** **
>
> The configuration we have is Sense as firewall that sends and has all
> stipulated Freeswitch ports open and forwards incoming ports to the
> Freeswitch server (Ubuntu/Freeswitch/fusionpbx)****
>
> ** **
>
> Internal  to internal call works****
>
> Internal call to the external location works  Both parties can talk without
> issues.****
>
> External location to Internal location:  we do not hear them, but they can
> hear us.****
>
> External location call to external location :  the originating caller
> cannot be heard, the second leg works perfectly. (This suggest that it is
> not a firewall issue but a configuration issue.)****
>
> External location calls to the voicemail and can record without problems a
> recording.****
>
> ** **
>
> The problem seems to be limited to the first leg of the call, in that if it
> originates from an external location the first leg cannot be heard.****
>
> ** **
>
> ** **
>
> Any suggestions on how to fix would be appreciated.  Thanks!****
>
> ** **
>
> Cjb****
>
> ** **
>
> ** **
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110819/9319475b/attachment.html 

From benkokakao at gmail.com  Fri Aug 19 22:40:56 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Fri, 19 Aug 2011 20:40:56 +0200
Subject: [Freeswitch-users] Fetching user-variables
In-Reply-To: <CAKzWOxVrgz6RuOAoGY1jSx_SZG74mRJBhzjGTSfHYNyc2-JgjA@mail.gmail.com>
References: <CAAMQ8bRTgb_Dxn_L=C4KEeoNM+WTsy6hJKdQp7d7EjdarLqBjA@mail.gmail.com>
	<CAKzWOxVrgz6RuOAoGY1jSx_SZG74mRJBhzjGTSfHYNyc2-JgjA@mail.gmail.com>
Message-ID: <CAAMQ8bRM5Vav+YTr5K3n-VkavW2mJfpHDY=RKwZFJD=7Gk7s7Q@mail.gmail.com>

On 19 August 2011 17:39, Michael Collins <msc at freeswitch.org> wrote:
> You probably need to add a "set_user" dp app in there somewhere. Can you
> pastebin your relevant dp configs? We'll take a look.

Ah, great - that's what i was looking for! Thanks :-)


From vipkilla at gmail.com  Fri Aug 19 22:47:34 2011
From: vipkilla at gmail.com (vip killa)
Date: Fri, 19 Aug 2011 14:47:34 -0400
Subject: [Freeswitch-users] enterprise deployment not working
In-Reply-To: <CAKzWOxW2scFwq3M7d1J_0y-wZ9+1_qW9X=dU3UGfBavTuS0yTw@mail.gmail.com>
References: <CAC-LwPPSAei6EyeQBtrHoin0DtN73f2zfd6GU5JyQm0Z8_rXPw@mail.gmail.com>
	<CAC-LwPP+oNWNu-5=-Y2KQwGkn0qaaf7Bq59CcXTo_Ura+AZwBA@mail.gmail.com>
	<CAC-LwPMAT1zsohk-hCSwpSLetui8-_figFGjDxztB3Z9FFxfZg@mail.gmail.com>
	<CAKzWOxUsDLEkN9qfsgiOf11F5M-3077AT8U_8Rc_fxNJ2-=HyQ@mail.gmail.com>
	<CAC-LwPPfpNz4D=1_B21hxPjuC5GP0nG6X9G6ByRjv3Jc7Z9BxQ@mail.gmail.com>
	<CAKzWOxW2scFwq3M7d1J_0y-wZ9+1_qW9X=dU3UGfBavTuS0yTw@mail.gmail.com>
Message-ID: <CAC-LwPO8pgURAUM+LYF4A9U79Kpa4ntMn_kL59GLPkHO1=NO9Q@mail.gmail.com>

I didn't create that page, i just added the opensips installation part.
i set the IPs according to whoever initially wrote that page.

On Fri, Aug 19, 2011 at 2:26 PM, Michael Collins <msc at freeswitch.org> wrote:

> in your example you used non-RFC1918 IP addresses, i.e. public addrs. Was
> that by design? Just curious.
>
> -MC
>
>
> On Fri, Aug 19, 2011 at 11:04 AM, vip killa <vipkilla at gmail.com> wrote:
>
>> http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS#OpenSIPS
>> Done. Hope that's not too much, i tried to make it as complete as
>> possible.
>>
>>
>> On Fri, Aug 19, 2011 at 11:43 AM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> Perhaps you just need to wrap your scripts and config files in
>>> <code></code> tags. You can also use <nowiki></nowiki> if you have text that
>>> is causing strange wiki features to be turned on.
>>>
>>> -MC
>>>
>>> On Fri, Aug 19, 2011 at 6:54 AM, vip killa <vipkilla at gmail.com> wrote:
>>>
>>>> I managed to get this working and i wanted to contribute how... i
>>>> edited
>>>> http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS#OpenSIPS
>>>> but the format is all screwed up and i'd like to add an init.d startup
>>>> script
>>>> perhaps someone could format that wiki page better and add an init.d
>>>> script?
>>>>
>>>> On Tue, Aug 16, 2011 at 3:22 PM, vip killa <vipkilla at gmail.com> wrote:
>>>>
>>>>> I did an ngrep trace and opensips is returning:
>>>>>
>>>>> SIP/2.0 503 Service Unavailable
>>>>> CSeq: 54 REGISTER
>>>>>
>>>>> before it even tries the FS server...
>>>>> perhaps this wiki page needs to be updated...
>>>>>
>>>>>
>>>>> On Tue, Aug 16, 2011 at 2:03 PM, vip killa <vipkilla at gmail.com> wrote:
>>>>>
>>>>>> I'm following tutorial at
>>>>>> http://wiki.freeswitch.org/wiki/Enterprise_deployment_<http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS>
>>>>>> OpenSIPS<http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS> and
>>>>>> opensips is not forwarding the registration requests...
>>>>>> Running opensips-1.6.4-2-tls,
>>>>>> Here is what the debugging looks like when i try to register with the
>>>>>> opensip box
>>>>>> 10.20.30.17 is the opensips box.
>>>>>> 10.20.30.18 is the FreeSWITCH box.
>>>>>>
>>>>>>
>>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>>> DBG:core:parse_msg: SIP Request:
>>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>>> DBG:core:parse_msg:  method:  <REGISTER>
>>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>>> DBG:core:parse_msg:  uri:     <sip:10.20.30.17>
>>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>>> DBG:core:parse_msg:  version: <SIP/2.0>
>>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>>> DBG:core:parse_headers: flags=2
>>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>>> DBG:core:get_hdr_field: cseq <CSeq>: <43> <REGISTER>
>>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>>> DBG:core:parse_via_param: found param type 232, <branch> =
>>>>>> <z9hG4bKda029da0-e604-1910-80bc-002622a67db9>; state=6
>>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>>> DBG:core:parse_via_param: found param type 235, <rport> = <n/a>; state=17
>>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>>> DBG:core:parse_via: end of header reached, state=5
>>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>>> DBG:core:parse_headers: via found, flags=2
>>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>>> DBG:core:parse_headers: this is the first via
>>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>>> DBG:core:receive_msg: After parse_msg...
>>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>>> DBG:core:receive_msg: preparing to run routing scripts...
>>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>>> DBG:core:parse_headers: flags=100
>>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>>> DBG:core:parse_to: end of header reached, state=10
>>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>>> DBG:core:parse_to: display={}, ruri={sip:1000 at 10.20.30.17}
>>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>>> DBG:core:get_hdr_field: <To> [24]; uri=[sip:1000 at 10.20.30.17]
>>>>>>  ]  16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>>> DBG:core:get_hdr_field: to body [<sip:1000 at 10.20.30.17>
>>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>>> DBG:core:get_hdr_field: content_length=0
>>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>>> DBG:maxfwd:is_maxfwd_present: value = 70
>>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>>> DBG:uri:has_totag: no totag
>>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>>> DBG:core:parse_to_param: tag=70ea9ca0-e604-1910-80bc-002622a67db9
>>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>>> DBG:core:parse_to: end of header reached, state=29
>>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>>> DBG:core:parse_to: display={}, ruri={sip:1000 at 10.20.30.17}
>>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>>> DBG:dispatcher:ds_select_dst: set [1]
>>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>>> DBG:dispatcher:ds_select_dst: alg hash [1], id [1]
>>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>>> DBG:core:parse_headers: flags=ffffffffffffffff
>>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>>> DBG:core:get_hdr_field: found end of header
>>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>>> DBG:core:check_ip_address: params 72.237.213.162, 72.237.213.162, 0
>>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>>> DBG:core:destroy_avp_list: destroying list (nil)
>>>>>> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]:
>>>>>> DBG:core:receive_msg: cleaning 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
>>>>
>>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110819/0e09cd9c/attachment-0001.html 

From msc at freeswitch.org  Fri Aug 19 23:43:09 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 19 Aug 2011 12:43:09 -0700
Subject: [Freeswitch-users] FreeSWITCH Janitorial Items
Message-ID: <CAKzWOxVmY9WGtBV0tkdkQzh8sL6fpWZ=Sd16Ltq00iecb7G_xA@mail.gmail.com>

Hello all!

We have need of your skills with a few sub-projects. Sorry, nothing
glamorous, just some behind-the-scenes work. First off, I need to know if
anyone out there has experience with UberCart for Drupal. If so, please
contact me off list so we can discuss the items I need help with.

Secondly, I have a housekeeping task. We have some new "standard vocabulary"
files for Callie US English. We need to add them to phrase_en.xml. However,
there are quite a few of them and some of them are the same recording with a
different inflection. For example, the inflection on the words "one" and
"message" are different in these phrases:

"To leave a message, press 1."
"Press 1 to leave a message."

I have a list of all the file names and what's in them but there's a bit of
work that needs to be done to document the inflection for each phrase. We
also need to settle on a naming convention. If you are in a position to
assist with this please contact me off list and I will give you more
information.

Thanks!

-Michael
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110819/75522c99/attachment.html 

From marketing at cluecon.com  Sat Aug 20 00:08:19 2011
From: marketing at cluecon.com (Michael Collins)
Date: Fri, 19 Aug 2011 13:08:19 -0700
Subject: [Freeswitch-users] ClueCon 2011 - Assistance w/ posting videos
Message-ID: <CAKzWOxWV5B-MrK-ss3uMpAMkDOK77cfqzz0hcrzVQb0vekAbrQ@mail.gmail.com>

Hello all!

I am looking for a volunteer to assist me with some Drupal/Web work on
cluecon.com. I would like to be ready to post the videos as they become
available and to do that I need to do some layout work on the cluecon site.
Anyone with any kind of CMS/Drupal/Web dev skills would be a perfect
candidate to help.

If you're in a position to assist please contact me off list.
Thanks!

-- 
Michael S Collins
ClueCon Team
http://www.cluecon.com
877-7-4ACLUE
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110819/b288631b/attachment.html 

From jon111882 at aol.com  Sat Aug 20 00:28:10 2011
From: jon111882 at aol.com (Jon)
Date: Fri, 19 Aug 2011 16:28:10 -0400
Subject: [Freeswitch-users] mod_dingaling delay buildup
In-Reply-To: <CA743B4B.5D31%jon111882@aol.com>
Message-ID: <CA743F3F.5D35%jon111882@aol.com>

I am facing delay problems with mod_dingaling.  The voice delay increases
each minute that I am on an outgoing call.  It starts fine and then will get
up to 5-10 seconds within 3 minutes.  The delay only occurs on the voice of
the called party.  I have googled and read countless threads and entries.  I
am running freeswitch on a VM.  I have run my setup on CentOS 5.6 i386,
CentOS 5.6 x86_64, CentOS 6 x86_64.  I have tried using a stun server as
well as port forwarding from my router.  I have tried using the
rtp-autoflush=true and rtp-timer-name=none settings.  All of my changes have
given me the same results.  I am using the current GIT version.  I have seen
a number of people have a flawless experience with this, so what am I doing
differently?  Any help?

Here are my current jingle configurations:

<extension name="gvoice_out">
   <condition field="destination_number" expression="^1(\d{10})$">
     <action application="set" data="rtp_autoflush=true"/>
     <action application="set" data="hangup_after_bridge=true"/>
     <action application="ring_ready"/>
     <action application="bridge"
data="dingaling/gtalk/+1$1 at voice.google.com"/>
   </condition>
 </extension>

<profile type="client">
    <param name="name" value="gtalk"/>
    <param name="login" value="<email>@gmail.com/talk"/>
    <param name="password" value="***"/>
    <param name="dialplan" value="XML"/>
    <param name="context" value="public"/>
    <param name="message" value="FreeSwitch for everybody"/>
    <param name="rtp-ip" value="auto"/>
    <param name="auto-login" value="true"/>
    <param name="sasl" value="plain"/>
    <param name="server" value="talk.google.com"/>
    <param name="tls" value="true"/>
    <param name="use-rtp-timer" value="true"/>
    <param name="exten" value="1000"/>
    <param name="vad" value="both"/>
    <!--<param name="avatar" value="/path/to/tiny.jpg"/>-->
    <param name="candidate-acl" value="wan.auto"/>
    <param name="local-network-acl" value="localnet.auto"/>
    <param name="rtp-timer-name value="none"/>
    <param name="rtp-autoflush" value="true"/>
  </profile>


-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110819/004453f1/attachment.html 

From justlikeef at gmail.com  Sat Aug 20 00:54:08 2011
From: justlikeef at gmail.com (Rob Hutton)
Date: Fri, 19 Aug 2011 16:54:08 -0400
Subject: [Freeswitch-users] mod_dingaling delay buildup
In-Reply-To: <CA743F3F.5D35%jon111882@aol.com>
References: <CA743F3F.5D35%jon111882@aol.com>
Message-ID: <201108191654.08619.justlikeef@gmail.com>

I think that setting use-rtp-timer=false has worked for some people...

On Friday 19 August 2011 16:28:10 Jon wrote:
> I am facing delay problems with mod_dingaling.  The voice delay increases
> each minute that I am on an outgoing call.  It starts fine and then will get
> up to 5-10 seconds within 3 minutes.  The delay only occurs on the voice of
> the called party.  I have googled and read countless threads and entries.  I
> am running freeswitch on a VM.  I have run my setup on CentOS 5.6 i386,
> CentOS 5.6 x86_64, CentOS 6 x86_64.  I have tried using a stun server as
> well as port forwarding from my router.  I have tried using the
> rtp-autoflush=true and rtp-timer-name=none settings.  All of my changes have
> given me the same results.  I am using the current GIT version.  I have seen
> a number of people have a flawless experience with this, so what am I doing
> differently?  Any help?
> 
> Here are my current jingle configurations:
> 
> <extension name="gvoice_out">
>    <condition field="destination_number" expression="^1(\d{10})$">
>      <action application="set" data="rtp_autoflush=true"/>
>      <action application="set" data="hangup_after_bridge=true"/>
>      <action application="ring_ready"/>
>      <action application="bridge"
> data="dingaling/gtalk/+1$1 at voice.google.com"/>
>    </condition>
>  </extension>
> 
> <profile type="client">
>     <param name="name" value="gtalk"/>
>     <param name="login" value="<email>@gmail.com/talk"/>
>     <param name="password" value="***"/>
>     <param name="dialplan" value="XML"/>
>     <param name="context" value="public"/>
>     <param name="message" value="FreeSwitch for everybody"/>
>     <param name="rtp-ip" value="auto"/>
>     <param name="auto-login" value="true"/>
>     <param name="sasl" value="plain"/>
>     <param name="server" value="talk.google.com"/>
>     <param name="tls" value="true"/>
>     <param name="use-rtp-timer" value="true"/>
>     <param name="exten" value="1000"/>
>     <param name="vad" value="both"/>
>     <!--<param name="avatar" value="/path/to/tiny.jpg"/>-->
>     <param name="candidate-acl" value="wan.auto"/>
>     <param name="local-network-acl" value="localnet.auto"/>
>     <param name="rtp-timer-name value="none"/>
>     <param name="rtp-autoflush" value="true"/>
>   </profile>
> 
> 
> 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110819/39a3fda3/attachment-0001.html 

From johnd at defyne.org  Sat Aug 20 08:39:23 2011
From: johnd at defyne.org (johnd at defyne.org)
Date: Sat, 20 Aug 2011 11:39:23 +0700
Subject: [Freeswitch-users] Returned mail: see transcript for details
Message-ID: <mailman.0.1313815133.29746.freeswitch-users@lists.freeswitch.org>

The message was not delivered due to the following reason:

Your message could not be delivered because the destination server was
not reachable within the allowed queue period. The amount of time
a message is queued before it is returned depends on local configura-
tion parameters.

Most likely there is a network problem that prevented delivery, but
it is also possible that the computer is turned off, or does not
have a mail system running right now.

Your message was not delivered within 6 days:
Host 106.218.121.36 is not responding.

The following recipients could not receive this message:
<freeswitch-users at lists.freeswitch.org>

Please reply to postmaster at lists.freeswitch.org
if you feel this message to be in error.

-------------- next part --------------
A non-text attachment was scrubbed...
Name: TEXT.PIF
Type: application/octet-stream
Size: 28864 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110820/344b5027/attachment.obj 

From jon111882 at aol.com  Sat Aug 20 09:23:12 2011
From: jon111882 at aol.com (Jon)
Date: Sat, 20 Aug 2011 01:23:12 -0400
Subject: [Freeswitch-users] mod_dingaling delay buildup
In-Reply-To: <201108191654.08619.justlikeef@gmail.com>
Message-ID: <CA74BCE8.5D4E%jon111882@aol.com>

Wow!  That was it!  So simple? Thanks for the help.

From:  Rob Hutton <justlikeef at gmail.com>
Date:  Fri, 19 Aug 2011 16:54:08 -0400
To:  <freeswitch-users at lists.freeswitch.org>
Cc:  Jonathan Neuffer <jon111882 at aol.com>
Subject:  Re: [Freeswitch-users] mod_dingaling delay buildup

I think that setting use-rtp-timer=false has worked for some people...



On Friday 19 August 2011 16:28:10 Jon wrote:

> I am facing delay problems with mod_dingaling.  The voice delay increases

> each minute that I am on an outgoing call.  It starts fine and then will get

> up to 5-10 seconds within 3 minutes.  The delay only occurs on the voice of

> the called party.  I have googled and read countless threads and entries.  I

> am running freeswitch on a VM.  I have run my setup on CentOS 5.6 i386,

> CentOS 5.6 x86_64, CentOS 6 x86_64.  I have tried using a stun server as

> well as port forwarding from my router.  I have tried using the

> rtp-autoflush=true and rtp-timer-name=none settings.  All of my changes have

> given me the same results.  I am using the current GIT version.  I have seen

> a number of people have a flawless experience with this, so what am I doing

> differently?  Any help?

> 

> Here are my current jingle configurations:

> 

> <extension name="gvoice_out">

>    <condition field="destination_number" expression="^1(\d{10})$">

>      <action application="set" data="rtp_autoflush=true"/>

>      <action application="set" data="hangup_after_bridge=true"/>

>      <action application="ring_ready"/>

>      <action application="bridge"

> data="dingaling/gtalk/+1$1 at voice.google.com"/>

>    </condition>

>  </extension>

> 

> <profile type="client">

>     <param name="name" value="gtalk"/>

>     <param name="login" value="<email>@gmail.com/talk"/>

>     <param name="password" value="***"/>

>     <param name="dialplan" value="XML"/>

>     <param name="context" value="public"/>

>     <param name="message" value="FreeSwitch for everybody"/>

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

>     <param name="auto-login" value="true"/>

>     <param name="sasl" value="plain"/>

>     <param name="server" value="talk.google.com"/>

>     <param name="tls" value="true"/>

>     <param name="use-rtp-timer" value="true"/>

>     <param name="exten" value="1000"/>

>     <param name="vad" value="both"/>

>     <!--<param name="avatar" value="/path/to/tiny.jpg"/>-->

>     <param name="candidate-acl" value="wan.auto"/>

>     <param name="local-network-acl" value="localnet.auto"/>

>     <param name="rtp-timer-name value="none"/>

>     <param name="rtp-autoflush" value="true"/>

>   </profile>

> 

> 

> 




-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110820/07458ca9/attachment.html 

From vaad.fabi at gmail.com  Sat Aug 20 16:45:06 2011
From: vaad.fabi at gmail.com (vaad.fabi at gmail.com)
Date: Sat, 20 Aug 2011 15:45:06 +0300
Subject: [Freeswitch-users] TLS+SRTP= 415 Bad Security Level
In-Reply-To: <CAKzWOxW2scFwq3M7d1J_0y-wZ9+1_qW9X=dU3UGfBavTuS0yTw@mail.gmail.com>
References: <CAC-LwPPSAei6EyeQBtrHoin0DtN73f2zfd6GU5JyQm0Z8_rXPw@mail.gmail.com>	<CAC-LwPP+oNWNu-5=-Y2KQwGkn0qaaf7Bq59CcXTo_Ura+AZwBA@mail.gmail.com>	<CAC-LwPMAT1zsohk-hCSwpSLetui8-_figFGjDxztB3Z9FFxfZg@mail.gmail.com>	<CAKzWOxUsDLEkN9qfsgiOf11F5M-3077AT8U_8Rc_fxNJ2-=HyQ@mail.gmail.com>	<CAC-LwPPfpNz4D=1_B21hxPjuC5GP0nG6X9G6ByRjv3Jc7Z9BxQ@mail.gmail.com>
	<CAKzWOxW2scFwq3M7d1J_0y-wZ9+1_qW9X=dU3UGfBavTuS0yTw@mail.gmail.com>
Message-ID: <4E4FAC52.9090507@gmail.com>

  Hi all,

Trying to get work freeswitch TLS+SRTP.
All configured as http://wiki.freeswitch.org/wiki/SIP_TLS describes.

Connected 2 eyebeam +tls\srtp, calls from eyebeam(tls+srtp) to 
freeswitch(tls+srtp) and further to sip gw, calls is ok.
But when i try to call between two eyebeams (tls+srtp) connected to FS i 
get "Unsuported Media Type" and console says:

/ua(0x8e2adc0): event i_state 415 Bad Security Level
nua(0x8e2adc0): event i_terminated 415 Bad Security Level/

Codecs prefs in sofia profile and both eyebeams same (g711a,g711u,gsm,g722).

Maybe somebody around? Please put me on the right way :)
thx

-- 
Vadim F.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110820/60e6074e/attachment.html 

From babak.freeswitch at gmail.com  Sat Aug 20 18:40:03 2011
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Sat, 20 Aug 2011 19:10:03 +0430
Subject: [Freeswitch-users] regex help
Message-ID: <CAEVfcaY3kKucJwuURcOAYuDCJ+RseEnJZFpw0KhJ75sPiqSqmQ@mail.gmail.com>

Hi
How can I use playandgetdigits to get 0 or 8 numbers? I mean it should be an
empty string or 8digits long.
I tried this
play_and_get_digits 0 8 3 7000 # file1 file2  number ^$|^d{{8}}$
but it's not passing empty strings
thanx
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110820/ad672d6a/attachment.html 

From bnaylor at sirran.com  Sun Aug 21 02:59:04 2011
From: bnaylor at sirran.com (Ben Naylor)
Date: Sat, 20 Aug 2011 23:59:04 +0100
Subject: [Freeswitch-users] Sangoma Media Gateway
In-Reply-To: <8238755A-2A09-4C9B-BF9C-C57AEFCB9912@gmail.com>
References: <017601cc5cfb$38b5b180$aa211480$@sirran.com>
	<8238755A-2A09-4C9B-BF9C-C57AEFCB9912@gmail.com>
Message-ID: <000001cc5f8c$c39e66a0$4adb33e0$@sirran.com>

Hi Deon

 

Did you encounter any problems when editing the dialplan?

 

I'm not 100% sure, but it would appear that whenever I try and edit
default.xml dialplan, the SMG changes status to 'down' in the webgui.  I
then change it straight back again and it comes back up briefly.

However, I've got to the point now where I can't seem to get the SMG to work
again, the TDM drivers start fine, but the SMG just does not want to start.

I have also check the sangomagw log file in /usr/local/smg/log/ but this
seems to stop logging anything once the SMG goes down.

 

Do you know of any other log files I could check?

 

Also, if I were to re-install the SMG, would re-running the make/install
commands just re-install over the top?  As a newbie to linux I am still not
sure if this is the case! J

 

Thanks for your help.

 

Regards

 

Ben

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Deon
Vermeulen
Sent: 18 August 2011 07:48
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Sangoma Media Gateway

 

Hi Ben

 

I do it as follows.

 

Go to File Editor -> dialplan -> default.xml

 

Look for  <extension name="to-sip">

 

At the very end of the extension there is a part that already shows you how
to route to IP.

 

                                    <!-- uncomment this if you want to dial
to a fixed IP addr -->

                                    <!-- <action application="bridge"
data="sofia/internal/${destnumber}@192.168.1.1
<mailto:sofia/internal/$%7bdestnumber%7d at 192.168.1.1> "/> -->

 

 

So you could have something like this.

 

                                    <action application="set"
data="sip_contact_user_replacement=${destnumber}"/>


                                    <action application="set"
data="hangup_after_bridge=yes"/>

                                    <action application="bridge"
data="sofia/internal/${destnumber}@192.168.1.1
<mailto:sofia/internal/$%7bdestnumber%7d at 192.168.1.1> "/>

                                    <action application="hangup"
data="${originate_disposition}"/>

                        

If you need to setup a DID then you could do something like this.

 

            <extension name="to-switchboard" >

                          <condition field="destination_number"
expression="xxxxxxxx">  ( REPLACE xxxxxx with the number dialled on the
PSTN)

                                    <action application="set"
data="span=${sip_h_X-FreeTDM-SpanName}"/>

                                    <action application="log" data="INFO
Incoming CALL for switchboard on span=${span}"/>

                                    <action application="set"
data="hangup_after_bridge=yes"/>

                                    <action application="bridge"
data="sofia/internal/yyyyyyyyy at 192.168.1.1"/>   (REPLACE yyyyyy with the
destination number of the SIP Phone this call needs to go to)

                                    <action application="hangup"
data="${originate_disposition}"/>

                          </condition>

            </extension>

 

 

If your calls need to be routed through the WAN to branch locations etc and
you don't want to use the default ulaw/alaw then you could force the codec
on the B-Leg of the call to what ever codec you prefer. This will depend on
what codecs is supported on the end devices as well as if your card supports
transcoding.

 

It could look something like this:

 

            <extension name="to-switchboard" >

                           <condition field="destination_number"
expression="xxxxxxxx">             

                                    <action application="set"
data="span=${sip_h_X-FreeTDM-SpanName}"/>

                                    <action application="log" data="INFO
Incoming CALL for switchboard on span=${span}"/>

                                    <action application="set"
data="hangup_after_bridge=yes"/>

                                    <action application="bridge"
data="sofia/internal/yyyyyyyyy at 192.168.1.1"/>

                                    <action application="hangup"
data="{absolute_codec_string=GSM} ${originate_disposition}"/>

                                    <action application="hangup"
data="${originate_disposition}"/>

                           </condition>

            </extension>

 

 

These are just simple examples you can use.

 

Hope this is helpful.

 

 

Regards

Deon

 

 

On Aug 17, 2011, at 5:32 PM, Ben Naylor wrote:





Hello

 

Apologies if this is in the wrong place, I am a complete beginner at
Freeswitch!

 

Has anyone had much joy with using the above to route calls from ISDN to a
SIP provider?  I have setup the SMG which is a stripped down version of
Freeswitch, but am struggling to work out what to configure to get this
working.

So far I have tried to set up an external gateway to my provider, but this
hasn't appeared in the SMG Web-gui as a SIP profile.  My provider doesn't
required auth by the way, they have just given me an IP to connect to.

 

Any help is greatly appreciated

 

Kind regards

 

Ben

 


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

 

  _____  

No virus found in this message.
Checked by AVG - www.avg.com
Version: 10.0.1392 / Virus Database: 1520/3840 - Release Date: 08/17/11

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110820/cafaffcb/attachment-0001.html 

From lakersman2006 at yahoo.com  Sun Aug 21 05:47:11 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Sat, 20 Aug 2011 18:47:11 -0700 (PDT)
Subject: [Freeswitch-users] fs_irvd
Message-ID: <1313891231.59698.YahooMailNeo@web161001.mail.bf1.yahoo.com>

Does anyone know how to make a start up script for the fs_irvd daemon?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110820/94d5e197/attachment.html 

From curriegrad2004 at gmail.com  Sun Aug 21 06:02:59 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sat, 20 Aug 2011 19:02:59 -0700
Subject: [Freeswitch-users] fs_irvd
In-Reply-To: <1313891231.59698.YahooMailNeo@web161001.mail.bf1.yahoo.com>
References: <1313891231.59698.YahooMailNeo@web161001.mail.bf1.yahoo.com>
Message-ID: <CAMWiMy9OJ+k6pajuKsBZRQo0NdfPa9LQpFKsmFxuVE_1aQ-ymQ@mail.gmail.com>

What disro are you using? If you're using CentOS there should be a
skel file in the /etc/init.d folder. Otherwise pick a startup script
that looks simple in the init.d directory and start working from that.

On Sat, Aug 20, 2011 at 6:47 PM, Sam <lakersman2006 at yahoo.com> wrote:
> Does anyone know how to make a start up script for the fs_irvd daemon?
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From moises.silva at gmail.com  Mon Aug 22 02:51:43 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Sun, 21 Aug 2011 18:51:43 -0400
Subject: [Freeswitch-users] freeswitch media_proxy and zrtp
In-Reply-To: <4E4906F6.9070609@petris.info>
References: <4E4906F6.9070609@petris.info>
Message-ID: <CAA4nhyD3Y1476t4CY-GE6hxaz0w3ZZeLkCC+A=8tQ+XbXVUKoA@mail.gmail.com>

On Mon, Aug 15, 2011 at 7:45 AM, Petr Nyklicek <petr at petris.info> wrote:
>
> ... but zrtp stream is still processes by FS
>

Yup, that's the current behaviour. The function read_rtp_packet() is
the one taking care of reading from the actual socket and if SRTP or
ZRTP is enabled, then decode the packet. The Proxy media option is
meant to avoid media processing (as in transcoding) but not to avoid
decryption (although is probably possible at least for ZRTP).

Moises Silva
Senior Software Engineer, Software Development Manager
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
L3R 9R6 Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com


From moises.silva at gmail.com  Mon Aug 22 03:03:48 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Sun, 21 Aug 2011 19:03:48 -0400
Subject: [Freeswitch-users] Sangoma Media Gateway
In-Reply-To: <000001cc5f8c$c39e66a0$4adb33e0$@sirran.com>
References: <017601cc5cfb$38b5b180$aa211480$@sirran.com>
	<8238755A-2A09-4C9B-BF9C-C57AEFCB9912@gmail.com>
	<000001cc5f8c$c39e66a0$4adb33e0$@sirran.com>
Message-ID: <CAA4nhyBg1L5uSWhYFUSvStexCXmAJb3_auz+cz9o69FVn5LasA@mail.gmail.com>

On Sat, Aug 20, 2011 at 6:59 PM, Ben Naylor <bnaylor at sirran.com> wrote:
> I?m not 100% sure, but it would appear that whenever I try and edit
> default.xml dialplan, the SMG changes status to ?down? in the webgui.? I
> then change it straight back again and it comes back up briefly.

The only reason I can think of for such behavior is that you have some
syntax error in the XML and the gateway fails to start. See the logs
/usr/local/smg/log/sangomagw.log to find out why the gateway fails to
start.

> However, I?ve got to the point now where I can?t seem to get the SMG to work
> again, the TDM drivers start fine, but the SMG just does not want to start.
>
> I have also check the sangomagw log file in /usr/local/smg/log/ but this
> seems to stop logging anything once the SMG goes down.
>
> Do you know of any other log files I could check?
>

Depending on the version of SMG  (can't remember the details), you
have /usr/local/smg/init.log or /usr/local/smg/log/init.log

> Also, if I were to re-install the SMG, would re-running the make/install
> commands just re-install over the top?? As a newbie to linux I am still not
> sure if this is the case! J
>

Latest version of smg will backup the auto-generated configuration
(freetdm.conf, wanpipe etc), but will NOT backup your dialplan, that
is up to you. In general I recommend you to backup your whole conf/
directory before upgrading.

Moises Silva
Senior Software Engineer, Software Development Manager
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
L3R 9R6 Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com


From lakersman2006 at yahoo.com  Mon Aug 22 03:56:27 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Sun, 21 Aug 2011 16:56:27 -0700 (PDT)
Subject: [Freeswitch-users] freeswitch php
Message-ID: <1313970987.96106.YahooMailNeo@web161008.mail.bf1.yahoo.com>

Does anyone know how to retrieve channel variables (ie. uuid, etc.) using the php example that was shown in the wiki below?


#!/usr/bin/php -q <?php // set a couple of things so we dont kill the system
ob_implicit_flush(true);
set_time_limit(30); // Open stdin so we can read the data in
$in = fopen("php://stdin", "r"); // Connect
echo "connect\n\n"; // Answer
echo "sendmsg\n";
echo "call-command: execute\n";
echo "execute-app-name: answer\n\n"; // Play a prompt
echo "sendmsg\n";
echo "call-command: execute\n";
echo "execute-app-name: playback\n";
echo "execute-app-arg: /usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav\n\n"; // Wait
sleep(5); // Hangup
echo "sendmsg\n";
echo "call-command: hangup\n\n"; fclose($in); ?>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110821/9b57cbcb/attachment.html 

From mario_fs at mgtech.com  Mon Aug 22 04:28:57 2011
From: mario_fs at mgtech.com (Mario G)
Date: Sun, 21 Aug 2011 17:28:57 -0700
Subject: [Freeswitch-users] Help with dual IP gateways
In-Reply-To: <CAGQG5MYoJf2_bO5tCLKrrq4tTSdd2prbWz7EXY4eGkB19cTmHg@mail.gmail.com>
References: <005501cc5b67$be0ace00$3a206a00$@yahoo.com>
	<A6DCFC10-7528-4364-8D4C-9DC4ACD7AD92@mgtech.com>
	<CAGQG5MYoJf2_bO5tCLKrrq4tTSdd2prbWz7EXY4eGkB19cTmHg@mail.gmail.com>
Message-ID: <A1B61709-C780-4E68-8432-355080CC8C94@mgtech.com>

WIll be forthcoming as soon as I get the time to do it right. Will post here when ready, hopefully Sep-Aug.

On Aug 17, 2011, at 12:29 AM, Abdul Basit wrote:

> indeed.
> 
> On Wed, Aug 17, 2011 at 7:01 AM, Mario G <mario_fs at mgtech.com> wrote:
> I worked a year on this and was planning to put it on the wiki. Does this help: I have 2 DSLs, 1 static, 1 dynamic, FS is on a Mac mini with only 1 nic. The router does dual wan/load balancing. And by golly.... FS can auto switch between DSL lines when one has a problem. Wondering if this info is worth my time putting on the wiki, I did the OS X page and rewrote other stuff and it took weeks. But if there is interest I will do it.
> 
> On Aug 15, 2011, at 9:23 AM, Lars Zeb wrote:
> 
> > Currently my LAN is connected to the internet via DSL. The FreeSWITCH box is
> > on this subnet. To save money, I am moving the data portion of my LAN to a
> > new ISP and I want to segregate the VOIP to another ISP. I am tired of
> > having a bad VOIP connection during lengthy downloads.
> >
> > My VOIP and FreeSWITCH skills are minimal. I have used FreeSWITCH for over a
> > year in a home/business environment. The only reason it is working is with
> > the help of this list.
> >
> > My knowledge of IP is similar. I do not know how to setup a LAN with two
> > gateways with all nodes seeing one another. I do want to be able to call out
> > via FreeSWITCH from a softphone on the data portion of the new LAN.
> >
> > A friend suggested I need a dual ported WAN firewall/router with load
> > balancing to enable all the nodes to be on the same subnet. Can anyone help
> > me with suggestions? Is there a consultant I can hire to help with this?
> >
> > Thanks, Lars
> >
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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,
> 
> Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110821/e4741fe2/attachment.html 

From bnaylor at sirran.com  Mon Aug 22 11:32:30 2011
From: bnaylor at sirran.com (Ben Naylor)
Date: Mon, 22 Aug 2011 08:32:30 +0100
Subject: [Freeswitch-users] Sangoma Media Gateway
In-Reply-To: <CAA4nhyBg1L5uSWhYFUSvStexCXmAJb3_auz+cz9o69FVn5LasA@mail.gmail.com>
References: <017601cc5cfb$38b5b180$aa211480$@sirran.com>	<8238755A-2A09-4C9B-BF9C-C57AEFCB9912@gmail.com>	<000001cc5f8c$c39e66a0$4adb33e0$@sirran.com>
	<CAA4nhyBg1L5uSWhYFUSvStexCXmAJb3_auz+cz9o69FVn5LasA@mail.gmail.com>
Message-ID: <000f01cc609d$a7205df0$f56119d0$@sirran.com>

Thanks Moses

It would appear that there was a slight syntax error in the dialplan that was causing the problem.  This has now been rectified and the SMG is stable again.
I have now got to the point where I can see the call being processed by the dialplan, but it stops at the following point -


2011-08-21 14:18:59.788464 [INFO] ftmod_sangoma_isdn_stack_rcv.c:75 [s1c17][1:18] Received SETUP (suId:1 suInstId:0 spInstId:1)
2011-08-21 14:18:59.788464 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:57 [s1c17][1:18] Processing SETUP (suId:1 suInstId:0 spInstId:1)
2011-08-21 14:18:59.788464 [INFO] ftmod_sangoma_isdn_stack_hndl.c:142 [s1c17][1:18] Incoming call: Called No:[xxxxxxx29999] Calling No:[xxxxxxxxx627]
2011-08-21 14:18:59.788464 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:194 [s1c17][1:18] Changed state from DOWN to RING
2011-08-21 14:18:59.788464 [DEBUG] ftdm_state.c:508 [s1c17][1:18] Executing state processor for RING
2011-08-21 14:18:59.788464 [DEBUG] ftmod_sangoma_isdn.c:614 [s1c17][1:18] Completed state change from DOWN to RING in 0ms
2011-08-21 14:18:59.788464 [DEBUG] ftmod_sangoma_isdn.c:625 [s1c17][1:18] processing state change to RING
2011-08-21 14:18:59.788464 [DEBUG] ftmod_sangoma_isdn.c:649 [s1c17][1:18] Sending incoming call from xxxxxxxxx627 to xxxxxxx29999 to FTDM core
2011-08-21 14:18:59.788464 [DEBUG] mod_freetdm.c:2138 got clear channel sig [START]
2011-08-21 14:18:59.788464 [DEBUG] ftdm_io.c:3077 [s1c17][1:18] Enabled software DTMF detector
2011-08-21 14:18:59.788464 [DEBUG] mod_freetdm.c:378 Set codec PCMA 20ms
2011-08-21 14:18:59.789378 [DEBUG] mod_freetdm.c:1545 Connect inbound channel FreeTDM/1:17/xxxxxxx29999
2011-08-21 14:18:59.789378 [NOTICE] switch_channel.c:811 New Channel FreeTDM/1:17/xxxxxxx29999 [79c560d7-ae19-470d-9d9a-a1ced5690c76]
2011-08-21 14:18:59.789378 [DEBUG] mod_freetdm.c:1596 Call Variable: freetdm_screening_ind=network-provided
2011-08-21 14:18:59.789378 [DEBUG] mod_freetdm.c:1596 Call Variable: freetdm_isdn.prog_ind.loc=public-net-remote-user
2011-08-21 14:18:59.789378 [DEBUG] mod_freetdm.c:1596 Call Variable: freetdm_isdn.prog_ind.descr=origination-is-non-isdn
2011-08-21 14:18:59.789378 [DEBUG] mod_freetdm.c:1596 Call Variable: freetdm_presentation_ind=presentation-allowed
2011-08-21 14:18:59.789378 [DEBUG] mod_freetdm.c:1601 (FreeTDM/1:17/xxxxxxx29999) State Change CS_NEW -> CS_INIT
2011-08-21 14:18:59.789378 [DEBUG] switch_core_session.c:1116 Send signal FreeTDM/1:17/xxxxxxx29999 [BREAK]
2011-08-21 14:18:59.789378 [DEBUG] switch_core_state_machine.c:320 (FreeTDM/1:17/xxxxxxx29999) Running State Change CS_INIT
2011-08-21 14:18:59.789378 [DEBUG] switch_core_state_machine.c:356 (FreeTDM/1:17/xxxxxxx29999) State INIT
2011-08-21 14:18:59.789378 [DEBUG] mod_freetdm.c:406 (FreeTDM/1:17/xxxxxxx29999) State Change CS_INIT -> CS_ROUTING
2011-08-21 14:18:59.789378 [DEBUG] switch_core_session.c:1116 Send signal FreeTDM/1:17/xxxxxxx29999 [BREAK]
2011-08-21 14:18:59.789378 [DEBUG] switch_core_state_machine.c:356 (FreeTDM/1:17/xxxxxxx29999) State INIT going to sleep
2011-08-21 14:18:59.789378 [DEBUG] switch_core_state_machine.c:320 (FreeTDM/1:17/xxxxxxx29999) Running State Change CS_ROUTING
2011-08-21 14:18:59.789378 [DEBUG] switch_channel.c:1660 (FreeTDM/1:17/xxxxxxx29999) Callstate Change DOWN -> RINGING
2011-08-21 14:18:59.791404 [DEBUG] switch_core_state_machine.c:359 (FreeTDM/1:17/xxxxxxx29999) State ROUTING
2011-08-21 14:18:59.791404 [DEBUG] mod_freetdm.c:431 FreeTDM/1:17/xxxxxxx29999 CHANNEL ROUTING
2011-08-21 14:18:59.791404 [DEBUG] mod_freetdm.c:434 [s1c17][1:18] Indicating PROCEED in state RING
2011-08-21 14:18:59.791404 [DEBUG] mod_freetdm.c:434 [s1c17][1:18] Changed state from RING to PROCEED
2011-08-21 14:18:59.791404 [DEBUG] ftdm_state.c:508 [s1c17][1:18] Executing state processor for PROCEED
2011-08-21 14:18:59.791404 [DEBUG] ftmod_sangoma_isdn.c:614 [s1c17][1:18] Completed state change from RING to PROCEED in 0ms
2011-08-21 14:18:59.791404 [DEBUG] ftmod_sangoma_isdn.c:625 [s1c17][1:18] processing state change to PROCEED
2011-08-21 14:18:59.791404 [INFO] ftmod_sangoma_isdn_stack_out.c:167 [s1c17][1:18] Sending PROCEED (suId:1 suInstId:1 spInstId:1 dchan:1 ces:0)
2011-08-21 14:18:59.791404 [DEBUG] switch_core_state_machine.c:77 FreeTDM/1:17/xxxxxxx29999 Standard ROUTING
2011-08-21 14:18:59.791404 [INFO] mod_dialplan_xml.c:331 Processing  <xxxxxxxxx627>->xxxxxxx29999 in context from-pstn
Dialplan: FreeTDM/1:17/xxxxxxx29999 parsing [from-pstn->to-sip] continue=false
Dialplan: FreeTDM/1:17/xxxxxxx29999 Regex (PASS) [to-sip] destination_number(xxxxxxx29999) =~ /^(.{1,})$/ break=never
Dialplan: FreeTDM/1:17/xxxxxxx29999 Action set(destnumber=xxxxxxx29999) 
Dialplan: FreeTDM/1:17/xxxxxxx29999 Regex (PASS) [to-sip] destination_number(xxxxxxx29999) =~ /^(.*)$/ break=on-false
Dialplan: FreeTDM/1:17/xxxxxxx29999 Action set(sip_contact_user_replacement=${destnumber}) 
Dialplan: FreeTDM/1:17/xxxxxxx29999 Action set(hangup_after_bridge=yes) 
Dialplan: FreeTDM/1:17/xxxxxxx29999 Action bridge(sofia/internal/${destnumber}@x.x.x.134) 
Dialplan: FreeTDM/1:17/xxxxxxx29999 Action hangup(${originate_disposition}) 
2011-08-21 14:18:59.793352 [DEBUG] switch_core_state_machine.c:119 (FreeTDM/1:17/xxxxxxx29999) State Change CS_ROUTING -> CS_EXECUTE
2011-08-21 14:18:59.793352 [DEBUG] switch_core_session.c:1116 Send signal FreeTDM/1:17/xxxxxxx29999 [BREAK]
2011-08-21 14:18:59.793352 [DEBUG] switch_core_state_machine.c:359 (FreeTDM/1:17/xxxxxxx29999) State ROUTING going to sleep
2011-08-21 14:18:59.793352 [DEBUG] switch_core_state_machine.c:320 (FreeTDM/1:17/xxxxxxx29999) Running State Change CS_EXECUTE
2011-08-21 14:18:59.793352 [DEBUG] switch_core_state_machine.c:366 (FreeTDM/1:17/xxxxxxx29999) State EXECUTE
2011-08-21 14:18:59.793352 [DEBUG] mod_freetdm.c:451 FreeTDM/1:17/xxxxxxx29999 CHANNEL EXECUTE
2011-08-21 14:18:59.793352 [DEBUG] switch_core_state_machine.c:157 FreeTDM/1:17/xxxxxxx29999 Standard EXECUTE
EXECUTE FreeTDM/1:17/xxxxxxx29999 set(destnumber=xxxxxxx29999)
2011-08-21 14:18:59.793352 [DEBUG] mod_dptools.c:1059 FreeTDM/1:17/xxxxxxx29999 SET [destnumber]=[xxxxxxx29999]
EXECUTE FreeTDM/1:17/xxxxxxx29999 set(sip_contact_user_replacement=xxxxxxx29999)
2011-08-21 14:18:59.794441 [DEBUG] mod_dptools.c:1059 FreeTDM/1:17/xxxxxxx29999 SET [sip_contact_user_replacement]=[xxxxxxx29999]
EXECUTE FreeTDM/1:17/xxxxxxx29999 set(hangup_after_bridge=yes)
2011-08-21 14:18:59.794441 [DEBUG] mod_dptools.c:1059 FreeTDM/1:17/xxxxxxx29999 SET [hangup_after_bridge]=[yes]
EXECUTE FreeTDM/1:17/xxxxxxx29999 bridge(sofia/internal/xxxxxxx29999 at x.x.x.134)
2011-08-21 14:18:59.795371 [NOTICE] switch_channel.c:811 New Channel sofia/internal/xxxxxxx29999 at x.x.x.134 [b0588ca1-6910-4d19-9747-99ad8c61634b]
2011-08-21 14:18:59.795371 [DEBUG] mod_sofia.c:4237 (sofia/internal/xxxxxxx29999 at x.x.x.134) State Change CS_NEW -> CS_INIT
2011-08-21 14:18:59.795371 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/xxxxxxx29999 at x.x.x.134 [BREAK]
2011-08-21 14:18:59.796356 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/xxxxxxx29999 at x.x.x.134) Running State Change CS_INIT
2011-08-21 14:18:59.796356 [DEBUG] switch_core_state_machine.c:356 (sofia/internal/xxxxxxx29999 at x.x.x.134) State INIT
2011-08-21 14:18:59.796356 [DEBUG] mod_sofia.c:84 sofia/internal/xxxxxxx29999 at x.x.x.134 SOFIA INIT
2011-08-21 14:18:59.796356 [DEBUG] mod_sofia.c:124 (sofia/internal/xxxxxxx29999 at x.x.x.134) State Change CS_INIT -> CS_ROUTING
2011-08-21 14:18:59.796356 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/xxxxxxx29999 at x.x.x.134 [BREAK]
2011-08-21 14:18:59.796356 [DEBUG] switch_core_state_machine.c:356 (sofia/internal/xxxxxxx29999 at x.x.x.134) State INIT going to sleep
2011-08-21 14:18:59.796356 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/xxxxxxx29999 at x.x.x.134) Running State Change CS_ROUTING
2011-08-21 14:18:59.796356 [DEBUG] switch_channel.c:1660 (sofia/internal/xxxxxxx29999 at x.x.x.134) Callstate Change DOWN -> RINGING
2011-08-21 14:18:59.797451 [DEBUG] switch_core_state_machine.c:359 (sofia/internal/xxxxxxx29999 at x.x.x.134) State ROUTING
2011-08-21 14:18:59.797451 [DEBUG] mod_sofia.c:147 sofia/internal/xxxxxxx29999 at x.x.x.134 SOFIA ROUTING
2011-08-21 14:18:59.797451 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/xxxxxxx29999 at x.x.x.134) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2011-08-21 14:18:59.797451 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/xxxxxxx29999 at x.x.x.134 [BREAK]
2011-08-21 14:18:59.797451 [DEBUG] switch_core_state_machine.c:359 (sofia/internal/xxxxxxx29999 at x.x.x.134) State ROUTING going to sleep
2011-08-21 14:18:59.797451 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/xxxxxxx29999 at x.x.x.134) Running State Change CS_CONSUME_MEDIA
2011-08-21 14:18:59.797451 [DEBUG] switch_core_state_machine.c:378 (sofia/internal/xxxxxxx29999 at x.x.x.134) State CONSUME_MEDIA
2011-08-21 14:18:59.797451 [DEBUG] switch_core_state_machine.c:378 (sofia/internal/xxxxxxx29999 at x.x.x.134) State CONSUME_MEDIA going to sleep
2011-08-21 14:18:59.797451 [DEBUG] sofia.c:4659 Channel sofia/internal/xxxxxxx29999 at x.x.x.134 entering state [calling][0]


It stays like this until I hang up the call, Freeswitch then tears down the call.  To me it looks like my SIP provider is not accepting the call in that format?  I have tried changing the way the number is presented to Freeswitch by altering the routing on the telco switch, but still no joy.
Is there anything else that could be causing an issue looking at the log above?

Thanks in advance for your help.

Kind regards

Ben

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Moises Silva
Sent: 22 August 2011 00:04
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Sangoma Media Gateway

On Sat, Aug 20, 2011 at 6:59 PM, Ben Naylor <bnaylor at sirran.com> wrote:
> I?m not 100% sure, but it would appear that whenever I try and edit 
> default.xml dialplan, the SMG changes status to ?down? in the webgui.  
> I then change it straight back again and it comes back up briefly.

The only reason I can think of for such behavior is that you have some syntax error in the XML and the gateway fails to start. See the logs /usr/local/smg/log/sangomagw.log to find out why the gateway fails to start.

> However, I?ve got to the point now where I can?t seem to get the SMG 
> to work again, the TDM drivers start fine, but the SMG just does not want to start.
>
> I have also check the sangomagw log file in /usr/local/smg/log/ but 
> this seems to stop logging anything once the SMG goes down.
>
> Do you know of any other log files I could check?
>

Depending on the version of SMG  (can't remember the details), you have /usr/local/smg/init.log or /usr/local/smg/log/init.log

> Also, if I were to re-install the SMG, would re-running the 
> make/install commands just re-install over the top?  As a newbie to 
> linux I am still not sure if this is the case! J
>

Latest version of smg will backup the auto-generated configuration (freetdm.conf, wanpipe etc), but will NOT backup your dialplan, that is up to you. In general I recommend you to backup your whole conf/ directory before upgrading.

Moises Silva
Senior Software Engineer, Software Development Manager Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. moy at sangoma.com


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

-----
No virus found in this message.
Checked by AVG - www.avg.com
Version: 10.0.1392 / Virus Database: 1520/3848 - Release Date: 08/21/11



From rentmycoder at gmail.com  Mon Aug 22 11:46:33 2011
From: rentmycoder at gmail.com (rentmycoder rentmycoder)
Date: Mon, 22 Aug 2011 09:46:33 +0200
Subject: [Freeswitch-users] intercept sethanguphook failure
Message-ID: <CAEW218J34LUCEgCLk59O0CfDZb54wQXbMkSfMpnZHF=81-FV5w@mail.gmail.com>

Hi guys,

I'm trying to make a callgroup intercept and need to make some tasks
after the intercepted call hang's up...
I've tried both intercept and uuid_bridge method, both works, but
neither way the hanguphook does not get's called after intercept...
settings: hangupafterbridge=true and continueonfail=true...
Testing on win32 using latest GIT.

lua script:
...
session:setHangupHook("extensions_hanguphook");
...
api:executeString("uuid_bridge " .. tostring(intercept_source) .. " "
.. tostring(session.uuid));
or this way:
session:execute("intercept", intercept_source);

I haven't looked into the source yet, any idea???

I don't now it's a bug it it's my mistake...

Thanks,
John


From Stefan.Weigel at allianz-warranty.com  Mon Aug 22 12:36:44 2011
From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan)
Date: Mon, 22 Aug 2011 10:36:44 +0200
Subject: [Freeswitch-users] Original Caller ID/number after
	attended	transfer
In-Reply-To: <CAKbxfG_GOZAbJD=zgciLwkFDpmZ7pwbzfdP2e_D6HTCwoP__=A@mail.gmail.com>
References: <5003D7D3E06F514E8C682F18D223265C04D3B36D6E@AZWSMS03.azwarranty.int>
	<CAKbxfG-6a26_NqN0UJ46sXXEbr2JHbF76kX5SUmkda7wgRWLZQ@mail.gmail.com>
	<5003D7D3E06F514E8C682F18D223265C04D3B36D76@AZWSMS03.azwarranty.int>
	<CAKbxfG_GOZAbJD=zgciLwkFDpmZ7pwbzfdP2e_D6HTCwoP__=A@mail.gmail.com>
Message-ID: <5003D7D3E06F514E8C682F18D223265C04D3B36D77@AZWSMS03.azwarranty.int>

Hi Anthony, list,



thanks for the tip, it's now working!





Thanks and best regards



Stefan



Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Anthony Minessale
Gesendet: Freitag, 19. August 2011 17:03
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] Original Caller ID/number after attended transfer



use set_profile_var on the inbound leg to set caller_id_name to the new value.



On Fri, Aug 19, 2011 at 2:56 AM, Weigel, Stefan <Stefan.Weigel at allianz-warranty.com<mailto:Stefan.Weigel at allianz-warranty.com>> wrote:

Hi Anthony, list,



we're working with Polycom Soundpoint 560 phones. But meanwhile I could solve this problem.

A 'global_getvar' showed me that 'ignore_display_updates' was 'true'. After setting it to 'false' I now get the original caller ID displayed when doing a transfer.



One point is left that I'm currently working on:



on incoming calls I add a leading 0 to the original number. I need to do this because if a internal member wants to call outside he/she needs to dial with leading 0.



[..]



        <action application="export" data="caller_id_number=0$1"/>

        <action application="export" data="effective_caller_id_number=0$1"/>

        <action application="set" data="ignore_display_updates=true"/>



[..]



The initial incoming call is working, the number has a leading zero. Doing an attended transfer to a phone I now have the original number (without leading 0). With a blind transfer it's working.



Any suggestions ?







Thanks in advance and best regards



Stefan





Von: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.f<mailto:freeswitch-users-bounces at lists.f>reeswitch.org<http://reeswitch.org>] Im Auftrag von Anthony Minessale
Gesendet: Donnerstag, 18. August 2011 18:45
An: FreeSWITCH Users Help
Betreff: Re: [Freeswitch-users] Original Caller ID/number after attended transfer



if you get phones that support display updates (polycom, snom, cisco and a few others) yes. It already works.

What phones do you have?



On Thu, Aug 18, 2011 at 7:36 AM, Weigel, Stefan <Stefan.Weigel at allianz-warranty.com<mailto:Stefan.Weigel at allianz-warranty.com>> wrote:

Hi all,



is there a possibility to display the original caller ID & number after doing a attended transfer. External call to phone A -> calls phone B (I see caller ID & number of phone A) -> doing an attended transfer of external call to phone B (still caller ID & number of phone A).





Thanks in advance and best regards,



Stefan





Stefan Weigel
System Specialist AITP



Allianz Automotive Services GmbH

Einsteinring 28
85609 Aschheim
Germany

Tel.: +49 89 2000 48 975<tel:%2B49%2089%202000%2048%20975>
Fax: +49 89 2000 48 566<tel:%2B49%2089%202000%2048%20566>
eMail: Stefan.Weigel at allianz-warranty.com<mailto:Stefan.Weigel at allianz-warranty.com>


http://www.allianz-warranty.com
Gesch?ftsf?hrung: Andreas R?sing, Horst Ziegler
Amtsgericht M?nchen, HRB 175682
F?r Umsatzsteuerzwecke: Ust-ID-Nr.: DE 262 617 720





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







--
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900<tel:%2B19193869900>



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







--
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:+19193869900

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110822/797efd73/attachment-0001.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.gif
Type: image/gif
Size: 4415 bytes
Desc: image001.gif
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110822/797efd73/attachment-0001.gif 

From mkopacki at gmail.com  Mon Aug 22 12:58:19 2011
From: mkopacki at gmail.com (Michal Kopacki)
Date: Mon, 22 Aug 2011 10:58:19 +0200
Subject: [Freeswitch-users] AMR transcoding
Message-ID: <4E521A2B.1050808@gmail.com>

     Hello,

   I have to create configuration for transcoding AMR <-> G.729/G711. My 
question is how to do this on freeswitch. I'm searching for some 
description but it appears it's only for asterisk. I appreciate any help.

-- 
Regards,
Michal


From sascha.daniels at amooma.de  Mon Aug 22 15:37:43 2011
From: sascha.daniels at amooma.de (Sascha Daniels)
Date: Mon, 22 Aug 2011 13:37:43 +0200
Subject: [Freeswitch-users] vm-alternate-greet-id is ignored
In-Reply-To: <4E257C0C.7040103@amooma.de>
References: <4E257C0C.7040103@amooma.de>
Message-ID: <4E523F87.9010109@amooma.de>

Hi together,

I found the solution.

You can't use param in a condition.

<action application="set"
data="voicemail_alternate_greet_id=${sip_to_user}"/>

Does the job.


Reards Sascha

Am 19.07.2011 14:43, schrieb Sascha Daniels:
> Hi together,
>
> I need to set a different Number in voicemail greetings.
>
> Just for testing purpose I tried to set it without a variable.
>
> xml.condition( :field => 'destination_number', :expression =>
> '^-vbox-(.+)$' ) {
>                     xml.action( :application => 'answer', :data =>
> 'voicemail_authorized=true' )
>                     xml.param( :name => 'vm-alternate-greet-id', :value
> => '4444'  )
>                     xml.action( :application => 'voicemail', :data =>
> 'default ${domain_name} $1' )
>                 }
>
> Unfortunately the real sip account is played.
>
> Did I get the documentation wrong?
>
> I am using "FreeSWITCH version: 1.0.head (git-2e651c8 2011-07-03
> 22-35-44 -0500)"
>
> Regards
>
>                 Sascha
>
>


-- 
AMOOMA GmbH - Bachstr. 124 - 56566 Neuwied  -->  http://www.amooma.de
Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister Montabaur B14998

B?cher: http://das-asterisk-buch.de - http://ruby-auf-schienen.de



From x.liu at hw.ac.uk  Mon Aug 22 16:16:12 2011
From: x.liu at hw.ac.uk (xl127)
Date: Mon, 22 Aug 2011 13:16:12 +0100
Subject: [Freeswitch-users] "make current" does not work with Fedora 14
In-Reply-To: <CAEVfcaY3kKucJwuURcOAYuDCJ+RseEnJZFpw0KhJ75sPiqSqmQ@mail.gmail.com>
References: <CAEVfcaY3kKucJwuURcOAYuDCJ+RseEnJZFpw0KhJ75sPiqSqmQ@mail.gmail.com>
Message-ID: <4E52488C.4010306@hw.ac.uk>

Hi, Before I submit an issue to Jira, I tried "make current" on Fedora 
14 and I got some errors.
I first tried "make current", noticed some errors then tried "make 
current > makeCurrentOutput.txt".

I pasted the contents of the file makeCurrentOutput.txt to the pastebin 
via user "xing"

The console outputs are as follows:

[root at localhost freeswitch]# make current > makeCurrentOutput.txt
grep: ../../../..//src/include/switch_version.h: No such file or directory
grep: ../../../..//src/include/switch_version.h: No such file or directory
grep: ../../../..//src/include/switch_version.h: No such file or directory
grep: ../../../..//src/include/switch_version.h: No such file or directory
grep: ../../../..//src/include/switch_version.h: No such file or directory
grep: ../../../..//src/include/switch_version.h: No such file or directory
/usr/lib/libnss3.so: undefined reference to `PR_FindSymbol'
/usr/lib/libnss3.so: undefined reference to `PR_RWLock_Rlock'
/usr/lib/libssl3.so: undefined reference to `PR_OpenAnonFileMap'
/usr/lib/libssl3.so: undefined reference to `PR_UnloadLibrary'
/usr/lib/libnss3.so: undefined reference to `PL_InitArenaPool'
/usr/lib/libnss3.so: undefined reference to `PR_NewRWLock'
/usr/lib/libnss3.so: undefined reference to `PR_RWLock_Wlock'
/usr/lib/libnss3.so: undefined reference to `PR_LoadLibrary'
/usr/lib/libldap_r-2.4.so.2: undefined reference to `PR_GetEnv'
/usr/lib/libssl3.so: undefined reference to `PR_LoadLibraryWithFlags'
/usr/lib/libnssutil3.so: undefined reference to `PL_ClearArenaPool'
/usr/lib/libnss3.so: undefined reference to `PR_DestroyRWLock'
/usr/lib/libnss3.so: undefined reference to `PR_NewTCPSocket'
/usr/lib/libldap_r-2.4.so.2: undefined reference to `PR_GetLibraryName'
/usr/lib/libssl3.so: undefined reference to `PR_ExportFileMapAsString'
/usr/lib/libssl3.so: undefined reference to `PR_GetLibraryFilePathname'
/usr/lib/libssl3.so: undefined reference to `PR_FindFunctionSymbol'
/usr/lib/libsmime3.so: undefined reference to `PL_NewHashTable'
/usr/lib/libldap_r-2.4.so.2: undefined reference to `PR_ErrorToString'
/usr/lib/libnss3.so: undefined reference to `PR_RWLock_Unlock'
/usr/lib/libssl3.so: undefined reference to `PR_ImportFileMapFromString'
/usr/lib/libldap_r-2.4.so.2: undefined reference to 
`PR_GetDirectorySeparator'
collect2: ld returned 1 exit status
make[3]: *** [freeswitch] Error 1
libtool: link: warning: `-version-info/-version-number' is ignored for 
convenience libraries
make[2]: *** [all-recursive] Error 1
make[1]: *** [all] Error 2
make: *** [current] Error 2


Now I have to do a fresh FS installation again.

Xing





-- 
Heriot-Watt University is a Scottish charity
registered under charity number SC000278.



From moises.silva at gmail.com  Mon Aug 22 16:58:36 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Mon, 22 Aug 2011 08:58:36 -0400
Subject: [Freeswitch-users] Sangoma Media Gateway
In-Reply-To: <000f01cc609d$a7205df0$f56119d0$@sirran.com>
References: <017601cc5cfb$38b5b180$aa211480$@sirran.com>
	<8238755A-2A09-4C9B-BF9C-C57AEFCB9912@gmail.com>
	<000001cc5f8c$c39e66a0$4adb33e0$@sirran.com>
	<CAA4nhyBg1L5uSWhYFUSvStexCXmAJb3_auz+cz9o69FVn5LasA@mail.gmail.com>
	<000f01cc609d$a7205df0$f56119d0$@sirran.com>
Message-ID: <CAA4nhyA_NM35uXt6aFgo5oDV2qxC=Nb8pOEJhNGZYwX50Vy2TA@mail.gmail.com>

On Mon, Aug 22, 2011 at 3:32 AM, Ben Naylor <bnaylor at sirran.com> wrote:
> 2011-08-21 14:18:59.797451 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/xxxxxxx29999 at x.x.x.134) Running State Change CS_CONSUME_MEDIA
> 2011-08-21 14:18:59.797451 [DEBUG] switch_core_state_machine.c:378 (sofia/internal/xxxxxxx29999 at x.x.x.134) State CONSUME_MEDIA
> 2011-08-21 14:18:59.797451 [DEBUG] switch_core_state_machine.c:378 (sofia/internal/xxxxxxx29999 at x.x.x.134) State CONSUME_MEDIA going to sleep
> 2011-08-21 14:18:59.797451 [DEBUG] sofia.c:4659 Channel sofia/internal/xxxxxxx29999 at x.x.x.134 entering state [calling][0]
>
>
> It stays like this until I hang up the call, Freeswitch then tears down the call. ?To me it looks like my SIP provider is not accepting the call in that format? ?I have tried changing the way the number is presented to Freeswitch by altering the routing on the telco switch, but still no joy.
> Is there anything else that could be causing an issue looking at the log above?

It seems to me you did not include the most important part of the log,
which is the unexpected hangup. All the logging you posted points to a
successful call (at least to the point where is trying to reach the
SIP peer).


Moises Silva
Senior Software Engineer, Software Development Manager
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
L3R 9R6 Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com


From moises.silva at gmail.com  Mon Aug 22 17:01:42 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Mon, 22 Aug 2011 09:01:42 -0400
Subject: [Freeswitch-users] AMR transcoding
In-Reply-To: <4E521A2B.1050808@gmail.com>
References: <4E521A2B.1050808@gmail.com>
Message-ID: <CAA4nhyBFkosothGquPiAS5Po8OvZ9GTx5MP3FRnN999+w4TqPg@mail.gmail.com>

On Mon, Aug 22, 2011 at 4:58 AM, Michal Kopacki <mkopacki at gmail.com> wrote:
> ? ? Hello,
>
> ? I have to create configuration for transcoding AMR <-> G.729/G711. My
> question is how to do this on freeswitch. I'm searching for some
> description but it appears it's only for asterisk. I appreciate any help.
>

You have to pay licenses to use AMR (AFAIK). Then you can use the
Sangoma transcoding card (D100, D150, D500 etc) to do the transcoding.

Moises Silva
Senior Software Engineer, Software Development Manager
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
L3R 9R6 Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com


From mkopacki at gmail.com  Mon Aug 22 17:36:16 2011
From: mkopacki at gmail.com (Michal Kopacki)
Date: Mon, 22 Aug 2011 15:36:16 +0200
Subject: [Freeswitch-users] AMR transcoding
In-Reply-To: <CAA4nhyBFkosothGquPiAS5Po8OvZ9GTx5MP3FRnN999+w4TqPg@mail.gmail.com>
References: <4E521A2B.1050808@gmail.com>
	<CAA4nhyBFkosothGquPiAS5Po8OvZ9GTx5MP3FRnN999+w4TqPg@mail.gmail.com>
Message-ID: <4E525B50.6080404@gmail.com>

     Yes. I'm aware of licensing and Sangoma cards (and I belive I don't 
need license if use D500) and I'm testing this too but sadly I need also 
fully software solution to implement on some dedicated vps without 
physical access.

-- 
Regards,
Michal

On 2011-08-22 15:01, Moises Silva wrote:
> On Mon, Aug 22, 2011 at 4:58 AM, Michal Kopacki<mkopacki at gmail.com>  wrote:
>>      Hello,
>>
>>    I have to create configuration for transcoding AMR<->  G.729/G711. My
>> question is how to do this on freeswitch. I'm searching for some
>> description but it appears it's only for asterisk. I appreciate any help.
>>
> You have to pay licenses to use AMR (AFAIK). Then you can use the
> Sangoma transcoding card (D100, D150, D500 etc) to do the transcoding.
>
> Moises Silva
> Senior Software Engineer, Software Development Manager
> Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
> L3R 9R6 Canada
> t. 1 905 474 1990 x128 | e. moy at sangoma.com
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From johns1433 at gmail.com  Mon Aug 22 18:01:11 2011
From: johns1433 at gmail.com (John Smith)
Date: Mon, 22 Aug 2011 16:01:11 +0200
Subject: [Freeswitch-users] mod_rtmp & Flex client
Message-ID: <CA+9AvwTBxfSc58NDOwcFPrMyAefv2RD0WSOaj3wiaUouvZQ-VQ@mail.gmail.com>

 Hi All,

I have installed a recent version of FS (git-cd31633 2011-08-17) in order to
use mod_rtmp.
The rtmp connexion to the server works fine but I don?t manage to add a sip
call (?+New Call? button).
When I fill the sip address and click ?call? I got an error from the Flash
Player: flash.net.NetConnection is not able to call the callback function
onMakeCall (this a javascript function defined in freeswitch.html).

I made several tests in order to find the source of the problem and it seems
to come from the add_call function called within onMakeCall. Without the
call to add_call onMakeCall works fine.
I tried several browsers, several version of the Flash Player (above 10) and
the problem occurred in all cases.

I?m now running short of ideas.
Has someone some suggestions of what I could try to investigate and solve
this issue?

Regards,

John
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110822/d0dc3aee/attachment.html 

From mrene_lists at avgs.ca  Mon Aug 22 18:14:49 2011
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Mon, 22 Aug 2011 16:14:49 +0200
Subject: [Freeswitch-users] mod_rtmp & Flex client
In-Reply-To: <CA+9AvwTBxfSc58NDOwcFPrMyAefv2RD0WSOaj3wiaUouvZQ-VQ@mail.gmail.com>
References: <CA+9AvwTBxfSc58NDOwcFPrMyAefv2RD0WSOaj3wiaUouvZQ-VQ@mail.gmail.com>
Message-ID: <F82AC524-B285-4798-B8E1-67FB5820EC0A@avgs.ca>

Hi,

It seems that the jquery.tmpl version in-tree is using a different function than the one I had when developing the client (they basically were using render() and now they call it tmpl()).

Fixed in 3d3e5c6.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 2011-08-22, at 4:01 PM, John Smith wrote:

>  Hi All,
> 
> I have installed a recent version of FS (git-cd31633 2011-08-17) in order to use mod_rtmp.
> The rtmp connexion to the server works fine but I don?t manage to add a sip call (?+New Call? button). 
> When I fill the sip address and click ?call? I got an error from the Flash Player: flash.net.NetConnection is not able to call the callback function onMakeCall (this a javascript function defined in freeswitch.html).
> 
> I made several tests in order to find the source of the problem and it seems to come from the add_call function called within onMakeCall. Without the call to add_call onMakeCall works fine.
> I tried several browsers, several version of the Flash Player (above 10) and the problem occurred in all cases.
> 
> I?m now running short of ideas. 
> Has someone some suggestions of what I could try to investigate and solve this issue?
> 
> Regards,
> 
> John
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110822/2930596d/attachment-0001.html 

From norstar at bigmir.net  Mon Aug 22 12:46:03 2011
From: norstar at bigmir.net (sucsession)
Date: Mon, 22 Aug 2011 11:46:03 +0300
Subject: [Freeswitch-users] Incoming DID
Message-ID: <172326630.20110822114603@bigmir.net>

Hello Freeswitch-users,

I have problem with incoming DID routing.
Internal  phones  normally  registered  and can make outgoing call and
call to each other.
But incoming DID call routed to voice mail or "goodbye" prompt.
Calls to other services (IVR, conference) are routed normally.
I'm connected to Nortel CS1000 via sip trunk.
Call log below:

2011-08-22 04:44:54.970599 [WARNING] sofia_reg.c:1241 SIP auth challenge (REGISTER) on sofia profile 'internal' for [8100 at 10.160.0.3] from ip 10.160.1.212
2011-08-22 04:45:00.718687 [NOTICE] switch_channel.c:816 New Channel sofia/external/7777 [39a3c5ec-834a-4c80-9b2e-b95e1b0f6fbd]
2011-08-22 04:45:00.719679 [DEBUG] sofia.c:4761 Channel sofia/external/7777 entering state [received][100]
2011-08-22 04:45:00.719679 [DEBUG] sofia.c:4772 Remote SDP:
v=0

o=- 663 1 IN IP4 10.1.1.22

s=-

t=0 0

m=audio 5200 RTP/AVP 8 0 18 101 111

c=IN IP4 10.160.0.205

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=rtpmap:111 X-nt-inforeq/8000

a=ptime:20


2011-08-22 04:45:00.719679 [DEBUG] sofia_glue.c:4656 Audio Codec Compare [PCMA:8:8000:20:64000]/[G722:9:8000:20:64000]
2011-08-22 04:45:00.719679 [DEBUG] sofia_glue.c:4656 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000]
2011-08-22 04:45:00.719679 [DEBUG] sofia_glue.c:4656 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000]
2011-08-22 04:45:00.719679 [DEBUG] sofia_glue.c:2788 Set Codec sofia/external/7777 PCMA/8000 20 ms 160 samples 64000 bits
2011-08-22 04:45:00.719679 [DEBUG] sofia_glue.c:4770 Set 2833 dtmf send/recv payload to 101
2011-08-22 04:45:00.719679 [DEBUG] sofia.c:4943 (sofia/external/7777) State Change CS_NEW -> CS_INIT
2011-08-22 04:45:00.719679 [DEBUG] switch_core_session.c:1114 Send signal sofia/external/7777 [BREAK]
2011-08-22 04:45:00.719679 [DEBUG] switch_core_state_machine.c:325 (sofia/external/7777) Running State Change CS_INIT
2011-08-22 04:45:00.719679 [DEBUG] switch_core_state_machine.c:361 (sofia/external/7777) State INIT
2011-08-22 04:45:00.719679 [DEBUG] mod_sofia.c:84 sofia/external/7777 SOFIA INIT
2011-08-22 04:45:00.719679 [DEBUG] mod_sofia.c:124 (sofia/external/7777) State Change CS_INIT -> CS_ROUTING
2011-08-22 04:45:00.719679 [DEBUG] switch_core_session.c:1114 Send signal sofia/external/7777 [BREAK]
2011-08-22 04:45:00.719679 [DEBUG] switch_core_state_machine.c:361 (sofia/external/7777) State INIT going to sleep
2011-08-22 04:45:00.719679 [DEBUG] switch_core_state_machine.c:325 (sofia/external/7777) Running State Change CS_ROUTING
2011-08-22 04:45:00.719679 [DEBUG] switch_channel.c:1667 (sofia/external/7777) Callstate Change DOWN -> RINGING
2011-08-22 04:45:00.719679 [DEBUG] switch_core_state_machine.c:364 (sofia/external/7777) State ROUTING
2011-08-22 04:45:00.719679 [DEBUG] mod_sofia.c:147 sofia/external/7777 SOFIA ROUTING
2011-08-22 04:45:00.719679 [DEBUG] switch_core_state_machine.c:77 sofia/external/7777 Standard ROUTING
2011-08-22 04:45:00.719679 [INFO] mod_dialplan_xml.c:331 Processing Vasya <7777;phone-context=cdp.udp>->8100;phone-context=cdp.udp in context public
Dialplan: sofia/external/7777 parsing [public->unloop] continue=false
Dialplan: sofia/external/7777 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/7777 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/external/7777 parsing [public->outside_call] continue=true
Dialplan: sofia/external/7777 Absolute Condition [outside_call]
Dialplan: sofia/external/7777 Action set(outside_call=true) 
Dialplan: sofia/external/7777 Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) 
Dialplan: sofia/external/7777 parsing [public->call_debug] continue=true
Dialplan: sofia/external/7777 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/external/7777 parsing [public->public_extensions] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [public_extensions] destination_number(8100;phone-context=cdp.udp) =~ /^(10[01][0-9])$/ break=on-false
Dialplan: sofia/external/7777 parsing [public->8000] continue=true
Dialplan: sofia/external/7777 Regex (PASS) [8000] context(public) =~ /public/ break=on-false
Dialplan: sofia/external/7777 Regex (FAIL) [8000] destination_number(8100;phone-context=cdp.udp) =~ /8000/ break=on-false
Dialplan: sofia/external/7777 parsing [public->8001] continue=false
Dialplan: sofia/external/7777 Regex (PASS) [8001] context(public) =~ /public/ break=on-false
Dialplan: sofia/external/7777 Regex (FAIL) [8001] destination_number(8100;phone-context=cdp.udp) =~ /8001/ break=on-false
Dialplan: sofia/external/7777 parsing [public->8100] continue=false
Dialplan: sofia/external/7777 Regex (PASS) [8100] context(public) =~ /public/ break=on-false
Dialplan: sofia/external/7777 Regex (PASS) [8100] destination_number(8100;phone-context=cdp.udp) =~ /8100/ break=on-false
Dialplan: sofia/external/7777 Action transfer(8100 XML Default) 
2011-08-22 04:45:00.720681 [DEBUG] switch_core_state_machine.c:119 (sofia/external/7777) State Change CS_ROUTING -> CS_EXECUTE
2011-08-22 04:45:00.720681 [DEBUG] switch_core_session.c:1114 Send signal sofia/external/7777 [BREAK]
2011-08-22 04:45:00.720681 [DEBUG] switch_core_state_machine.c:364 (sofia/external/7777) State ROUTING going to sleep
2011-08-22 04:45:00.720681 [DEBUG] switch_core_state_machine.c:325 (sofia/external/7777) Running State Change CS_EXECUTE
2011-08-22 04:45:00.720681 [DEBUG] switch_core_state_machine.c:371 (sofia/external/7777) State EXECUTE
2011-08-22 04:45:00.721680 [DEBUG] mod_sofia.c:240 sofia/external/7777 SOFIA EXECUTE
2011-08-22 04:45:00.721680 [DEBUG] switch_core_state_machine.c:157 sofia/external/7777 Standard EXECUTE
EXECUTE sofia/external/7777 set(outside_call=true)
2011-08-22 04:45:00.721680 [DEBUG] mod_dptools.c:1060 sofia/external/7777 SET [outside_call]=[true]
EXECUTE sofia/external/7777 set(RFC2822_DATE=Mon, 22 Aug 2011 04:45:00 +0300)
2011-08-22 04:45:00.721680 [DEBUG] mod_dptools.c:1060 sofia/external/7777 SET [RFC2822_DATE]=[Mon, 22 Aug 2011 04:45:00 +0300]
EXECUTE sofia/external/7777 transfer(8100 XML Default)
2011-08-22 04:45:00.721680 [DEBUG] switch_ivr.c:1597 (sofia/external/7777) State Change CS_EXECUTE -> CS_ROUTING
2011-08-22 04:45:00.721680 [DEBUG] switch_core_session.c:1114 Send signal sofia/external/7777 [BREAK]
2011-08-22 04:45:00.721680 [DEBUG] switch_core_session.c:707 Send signal sofia/external/7777 [BREAK]
2011-08-22 04:45:00.721680 [NOTICE] switch_ivr.c:1603 Transfer sofia/external/7777 to XML[8100 at Default]
2011-08-22 04:45:00.721680 [DEBUG] switch_core_state_machine.c:371 (sofia/external/7777) State EXECUTE going to sleep
2011-08-22 04:45:00.721680 [DEBUG] switch_core_state_machine.c:325 (sofia/external/7777) Running State Change CS_ROUTING
2011-08-22 04:45:00.722681 [DEBUG] switch_core_state_machine.c:364 (sofia/external/7777) State ROUTING
2011-08-22 04:45:00.722681 [DEBUG] mod_sofia.c:147 sofia/external/7777 SOFIA ROUTING
2011-08-22 04:45:00.722681 [DEBUG] switch_core_state_machine.c:77 sofia/external/7777 Standard ROUTING
2011-08-22 04:45:00.722681 [INFO] mod_dialplan_xml.c:331 Processing Golubenkoff A. V. <7777;phone-context=cdp.udp>->8100 in context Default
Dialplan: sofia/external/7777 parsing [Default->unloop] continue=false
Dialplan: sofia/external/7777 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/7777 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->tod_example] continue=true
Dialplan: sofia/external/7777 Date/Time Match (FAIL) [tod_example] break=on-false
Dialplan: sofia/external/7777 parsing [Default->holiday_example] continue=true
Dialplan: sofia/external/7777 Date/Time Match (FAIL) [holiday_example] break=on-false
Dialplan: sofia/external/7777 parsing [Default->global-intercept] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [global-intercept] destination_number(8100) =~ /^\*886$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->group-intercept] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [group-intercept] destination_number(8100) =~ /^\*8$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->intercept-ext] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [intercept-ext] destination_number(8100) =~ /^\*\*(\d+)$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->redial] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [redial] destination_number(8100) =~ /^(redial|\*870)$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->global] continue=true
Dialplan: sofia/external/7777 Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/external/7777 Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never
Dialplan: sofia/external/7777 Absolute Condition [global]
Dialplan: sofia/external/7777 Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) 
Dialplan: sofia/external/7777 Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) 
Dialplan: sofia/external/7777 Action hash(insert/${domain_name}-last_dial/global/${uuid}) 
Dialplan: sofia/external/7777 Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) 
Dialplan: sofia/external/7777 parsing [Default->snom-demo-2] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [snom-demo-2] destination_number(8100) =~ /^\*9001$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->snom-demo-1] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [snom-demo-1] destination_number(8100) =~ /^\*9000$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->eavesdrop] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [eavesdrop] destination_number(8100) =~ /^\*88(\d{2,7})$|^\*0(.*)$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->eavesdrop] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [eavesdrop] destination_number(8100) =~ /^\*779$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->call_privacy] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [call_privacy] destination_number(8100) =~ /^\*67(\d+)$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->call_return] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [call_return] destination_number(8100) =~ /^\*69$|^869$|^lcr$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->del-group] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [del-group] destination_number(8100) =~ /^\*\*80(\d{2})$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->add-group] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [add-group] destination_number(8100) =~ /^\*\*81(\d{2})$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->call-group-simo] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [call-group-simo] destination_number(8100) =~ /^\*\*82(\d{2})$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->call-group-order] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [call-group-order] destination_number(8100) =~ /^\*83(\d{2})$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->extension-intercom] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [extension-intercom] destination_number(8100) =~ /^\*8(\d{2,7})$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->Local_Extension_Skinny] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [Local_Extension_Skinny] destination_number(8100) =~ /^(11[01][0-9])$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->send_to_voicemail] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [send_to_voicemail] destination_number(8100) =~ /^\*99(\d{2,7})$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->Conference] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [Conference] destination_number(8100) =~ /^8888$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->sc1000.10d] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [sc1000.10d] destination_number(8100) =~ /^(\d{10})$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->sc1000.7d] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [sc1000.7d] destination_number(8100) =~ /^(\d{7})$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->sc1000.d4] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [sc1000.d4] destination_number(8100) =~ /^(7\d{3})$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->Conference_Equation] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [Conference_Equation] destination_number(8100) =~ /^8800$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->101] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [101] destination_number(8100) =~ /^101$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->pizza_demo] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [pizza_demo] destination_number(8100) =~ /^(pizza|74992)$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->Talking Clock Time] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [Talking Clock Time] destination_number(8100) =~ /9170/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->Talking Clock Date] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [Talking Clock Date] destination_number(8100) =~ /9171/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->Talking Clock Date and Time] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [Talking Clock Date and Time] destination_number(8100) =~ /9172/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->Recordings] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [Recordings] destination_number(8100) =~ /^\*732$|^\*732673$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->group_dial_sales] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [group_dial_sales] destination_number(8100) =~ /^\*2000$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->group_dial_support] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [group_dial_support] destination_number(8100) =~ /^\*2001$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->group_dial_billing] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [group_dial_billing] destination_number(8100) =~ /^\*2002$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->operator] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [operator] destination_number(8100) =~ /^operator$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->vmain] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [vmain] destination_number(8100) =~ /^vmain$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->vmain1] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [vmain1] destination_number(8100) =~ /^vmain1$|^\*97$|^\*4000$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->vmain2] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [vmain2] destination_number(8100) =~ /^vmain2$|^\*98$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->sip_uri] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [sip_uri] destination_number(8100) =~ /^sip:(.*)$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->nb_conferences] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [nb_conferences] destination_number(8100) =~ /^\*(30\d{2})$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->wb_conferences] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [wb_conferences] destination_number(8100) =~ /^\*(31\d{2})$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->uwb_conferences] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [uwb_conferences] destination_number(8100) =~ /^\*(32\d{2})$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->cdquality_conferences] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [cdquality_conferences] destination_number(8100) =~ /^\*(33\d{2})$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->freeswitch_public_conf_via_sip] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [freeswitch_public_conf_via_sip] destination_number(8100) =~ /^\*9(888|8888|1616|3232)$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->mad_boss_intercom] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [mad_boss_intercom] destination_number(8100) =~ /^\*0911$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->mad_boss_intercom] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [mad_boss_intercom] destination_number(8100) =~ /^\*0912$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->mad_boss] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [mad_boss] destination_number(8100) =~ /^\*0913$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->ivr_demo] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [ivr_demo] destination_number(8100) =~ /^\*5000$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->dynamic_conference] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [dynamic_conference] destination_number(8100) =~ /^\*5001$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->rtp_multicast_page] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [rtp_multicast_page] destination_number(8100) =~ /^pagegroup$|^\*7243$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->park] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [park] destination_number(8100) =~ /^\*5900$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->unpark] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [unpark] destination_number(8100) =~ /^\*5901$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->valet_park_in] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [valet_park_in] destination_number(8100) =~ /^\*(6000)$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->valet_park_out] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [valet_park_out] destination_number(8100) =~ /^\*(60\d\d)$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->park] continue=false
Dialplan: sofia/external/7777 Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/external/7777 Regex (FAIL) [park] destination_number(8100) =~ /park\+(\d+)/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->unpark] continue=false
Dialplan: sofia/external/7777 Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/external/7777 Regex (FAIL) [unpark] destination_number(8100) =~ /^parking$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->park] continue=false
Dialplan: sofia/external/7777 Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/external/7777 Regex (FAIL) [park] destination_number(8100) =~ /callpark/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->unpark] continue=false
Dialplan: sofia/external/7777 Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/external/7777 Regex (FAIL) [unpark] destination_number(8100) =~ /pickup/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->wait] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [wait] destination_number(8100) =~ /^wait$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->fax_receive] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [fax_receive] destination_number(8100) =~ /^\*9178$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->fax_transmit] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [fax_transmit] destination_number(8100) =~ /^\*9179$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->ringback_180] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [ringback_180] destination_number(8100) =~ /^\*9180$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->ringback_183_uk_ring] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [ringback_183_uk_ring] destination_number(8100) =~ /^\*9181$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->ringback_183_music_ring] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [ringback_183_music_ring] destination_number(8100) =~ /^\*9182$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->ringback_post_answer_uk_ring] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [ringback_post_answer_uk_ring] destination_number(8100) =~ /^\*9183$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->ringback_post_answer_music] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [ringback_post_answer_music] destination_number(8100) =~ /^\*9184$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->ClueCon] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [ClueCon] destination_number(8100) =~ /^\*9191$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->show_info] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [show_info] destination_number(8100) =~ /^\*9192$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->video_record] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [video_record] destination_number(8100) =~ /^\*9193$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->video_playback] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [video_playback] destination_number(8100) =~ /^\*9194$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->delay_echo] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [delay_echo] destination_number(8100) =~ /^\*9195$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->echo] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [echo] destination_number(8100) =~ /^\*9196$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->milliwatt] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [milliwatt] destination_number(8100) =~ /^\*9197$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->tone_stream] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [tone_stream] destination_number(8100) =~ /^\*9198$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->zrtp_enrollement] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [zrtp_enrollement] destination_number(8100) =~ /^\*9787$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->hold_music] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [hold_music] destination_number(8100) =~ /^\*9664$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->Local_Extension] continue=false
Dialplan: sofia/external/7777 Regex (PASS) [Local_Extension] destination_number(8100) =~ /(^\d{2,7}$)/ break=on-false
Dialplan: sofia/external/7777 Action set(dialed_extension=8100) 
Dialplan: sofia/external/7777 Action export(dialed_extension=8100) 
Dialplan: sofia/external/7777 Action bind_meta_app(1 b s execute_extension::dx XML features) 
Dialplan: sofia/external/7777 Action bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/archive/${strftime(%Y)}/${strftime(%b)}/${strftime(%d)}/${uuid}.wav) 
Dialplan: sofia/external/7777 Action bind_meta_app(3 b s execute_extension::cf XML features) 
Dialplan: sofia/external/7777 Action set(ringback=${us-ring}) 
Dialplan: sofia/external/7777 Action set(transfer_ringback=local_stream://moh) 
Dialplan: sofia/external/7777 Action set(call_timeout=30) 
Dialplan: sofia/external/7777 Action set(hangup_after_bridge=true) 
Dialplan: sofia/external/7777 Action set(continue_on_fail=true) 
Dialplan: sofia/external/7777 Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}) 
Dialplan: sofia/external/7777 Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) 
Dialplan: sofia/external/7777 Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}) 
Dialplan: sofia/external/7777 Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) 
Dialplan: sofia/external/7777 Action bridge(user/${dialed_extension}@${domain_name}) 
Dialplan: sofia/external/7777 Action answer() 
Dialplan: sofia/external/7777 Action sleep(1000) 
Dialplan: sofia/external/7777 Action voicemail(default ${domain_name} ${dialed_extension}) 
2011-08-22 04:45:00.724680 [DEBUG] switch_core_state_machine.c:119 (sofia/external/7777) State Change CS_ROUTING -> CS_EXECUTE
2011-08-22 04:45:00.724680 [DEBUG] switch_core_session.c:1114 Send signal sofia/external/7777 [BREAK]
2011-08-22 04:45:00.724680 [DEBUG] switch_core_state_machine.c:364 (sofia/external/7777) State ROUTING going to sleep
2011-08-22 04:45:00.724680 [DEBUG] switch_core_state_machine.c:325 (sofia/external/7777) Running State Change CS_EXECUTE
2011-08-22 04:45:00.724680 [DEBUG] switch_core_state_machine.c:371 (sofia/external/7777) State EXECUTE
2011-08-22 04:45:00.724680 [DEBUG] mod_sofia.c:240 sofia/external/7777 SOFIA EXECUTE
2011-08-22 04:45:00.724680 [DEBUG] switch_core_state_machine.c:157 sofia/external/7777 Standard EXECUTE
EXECUTE sofia/external/7777 hash(insert/10.160.0.3-spymap/7777;phone-context=cdp.udp/39a3c5ec-834a-4c80-9b2e-b95e1b0f6fbd)
EXECUTE sofia/external/7777 hash(insert/10.160.0.3-last_dial/7777;phone-context=cdp.udp/8100)
EXECUTE sofia/external/7777 hash(insert/10.160.0.3-last_dial/global/39a3c5ec-834a-4c80-9b2e-b95e1b0f6fbd)
EXECUTE sofia/external/7777 set(RFC2822_DATE=Mon, 22 Aug 2011 04:45:00 +0300)
2011-08-22 04:45:00.725680 [DEBUG] mod_dptools.c:1060 sofia/external/7777 SET [RFC2822_DATE]=[Mon, 22 Aug 2011 04:45:00 +0300]
EXECUTE sofia/external/7777 set(dialed_extension=8100)
2011-08-22 04:45:00.725680 [DEBUG] mod_dptools.c:1060 sofia/external/7777 SET [dialed_extension]=[8100]
EXECUTE sofia/external/7777 export(dialed_extension=8100)
2011-08-22 04:45:00.725680 [DEBUG] switch_channel.c:965 EXPORT (export_vars) [dialed_extension]=[8100]
EXECUTE sofia/external/7777 bind_meta_app(1 b s execute_extension::dx XML features)
2011-08-22 04:45:00.726681 [INFO] switch_ivr_async.c:3014 Bound B-Leg: *1 execute_extension::dx XML features
EXECUTE sofia/external/7777 bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/archive/2011/Aug/22/39a3c5ec-834a-4c80-9b2e-b95e1b0f6fbd.wav)
2011-08-22 04:45:00.726681 [INFO] switch_ivr_async.c:3014 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/archive/2011/Aug/22/39a3c5ec-834a-4c80-9b2e-b95e1b0f6fbd.wav
EXECUTE sofia/external/7777 bind_meta_app(3 b s execute_extension::cf XML features)
2011-08-22 04:45:00.726681 [INFO] switch_ivr_async.c:3014 Bound B-Leg: *3 execute_extension::cf XML features
EXECUTE sofia/external/7777 set(ringback=%(2000, 4000, 440.0, 480.0))
2011-08-22 04:45:00.726681 [DEBUG] mod_dptools.c:1060 sofia/external/7777 SET [ringback]=[%(2000, 4000, 440.0, 480.0)]
EXECUTE sofia/external/7777 set(transfer_ringback=local_stream://moh)
2011-08-22 04:45:00.727682 [DEBUG] mod_dptools.c:1060 sofia/external/7777 SET [transfer_ringback]=[local_stream://moh]
EXECUTE sofia/external/7777 set(call_timeout=30)
2011-08-22 04:45:00.727682 [DEBUG] mod_dptools.c:1060 sofia/external/7777 SET [call_timeout]=[30]
EXECUTE sofia/external/7777 set(hangup_after_bridge=true)
2011-08-22 04:45:00.727682 [DEBUG] mod_dptools.c:1060 sofia/external/7777 SET [hangup_after_bridge]=[true]
EXECUTE sofia/external/7777 set(continue_on_fail=true)
2011-08-22 04:45:00.727682 [DEBUG] mod_dptools.c:1060 sofia/external/7777 SET [continue_on_fail]=[true]
EXECUTE sofia/external/7777 hash(insert/10.160.0.3-call_return/8100/7777;phone-context=cdp.udp)
EXECUTE sofia/external/7777 hash(insert/10.160.0.3-last_dial_ext/8100/39a3c5ec-834a-4c80-9b2e-b95e1b0f6fbd)
EXECUTE sofia/external/7777 set(called_party_callgroup=)
2011-08-22 04:45:00.728678 [DEBUG] mod_dptools.c:1060 sofia/external/7777 SET [called_party_callgroup]=[UNDEF]
EXECUTE sofia/external/7777 hash(insert/10.160.0.3-last_dial//39a3c5ec-834a-4c80-9b2e-b95e1b0f6fbd)
EXECUTE sofia/external/7777 bridge(user/8100 at 10.160.0.3)
2011-08-22 04:45:00.729678 [DEBUG] switch_channel.c:922 sofia/external/7777 EXPORTING[export_vars] [dialed_extension]=[8100] to event
2011-08-22 04:45:00.729678 [DEBUG] switch_ivr_originate.c:1873 Parsing global variables
2011-08-22 04:45:00.729678 [DEBUG] switch_channel.c:922 sofia/external/7777 EXPORTING[export_vars] [dialed_extension]=[8100] to event
2011-08-22 04:45:00.729678 [DEBUG] switch_ivr_originate.c:1873 Parsing global variables
2011-08-22 04:45:00.729678 [DEBUG] switch_event.c:1170 Parsing variable [presence_id]=[8100 at 10.160.0.3]
2011-08-22 04:45:00.730678 [NOTICE] switch_channel.c:816 New Channel sofia/internal/sip:8100 at 10.160.1.212:5060 [cb0e41a2-c78f-492d-bf62-0253dc747736]
2011-08-22 04:45:00.730678 [DEBUG] mod_sofia.c:4305 (sofia/internal/sip:8100 at 10.160.1.212:5060) State Change CS_NEW -> CS_INIT
2011-08-22 04:45:00.730678 [DEBUG] switch_core_session.c:1114 Send signal sofia/internal/sip:8100 at 10.160.1.212:5060 [BREAK]
2011-08-22 04:45:00.730678 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/sip:8100 at 10.160.1.212:5060) Running State Change CS_INIT
2011-08-22 04:45:00.730678 [DEBUG] switch_core_state_machine.c:361 (sofia/internal/sip:8100 at 10.160.1.212:5060) State INIT
2011-08-22 04:45:00.730678 [DEBUG] mod_sofia.c:84 sofia/internal/sip:8100 at 10.160.1.212:5060 SOFIA INIT
2011-08-22 04:45:00.731677 [DEBUG] mod_sofia.c:124 (sofia/internal/sip:8100 at 10.160.1.212:5060) State Change CS_INIT -> CS_ROUTING
2011-08-22 04:45:00.731677 [DEBUG] switch_core_session.c:1114 Send signal sofia/internal/sip:8100 at 10.160.1.212:5060 [BREAK]
2011-08-22 04:45:00.731677 [DEBUG] switch_core_state_machine.c:361 (sofia/internal/sip:8100 at 10.160.1.212:5060) State INIT going to sleep
2011-08-22 04:45:00.731677 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/sip:8100 at 10.160.1.212:5060) Running State Change CS_ROUTING
2011-08-22 04:45:00.731677 [DEBUG] switch_channel.c:1667 (sofia/internal/sip:8100 at 10.160.1.212:5060) Callstate Change DOWN -> RINGING
2011-08-22 04:45:00.731677 [DEBUG] sofia.c:4761 Channel sofia/internal/sip:8100 at 10.160.1.212:5060 entering state [calling][0]
2011-08-22 04:45:00.731677 [DEBUG] switch_core_state_machine.c:364 (sofia/internal/sip:8100 at 10.160.1.212:5060) State ROUTING
2011-08-22 04:45:00.731677 [DEBUG] mod_sofia.c:147 sofia/internal/sip:8100 at 10.160.1.212:5060 SOFIA ROUTING
2011-08-22 04:45:00.731677 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/sip:8100 at 10.160.1.212:5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2011-08-22 04:45:00.731677 [DEBUG] switch_core_session.c:1114 Send signal sofia/internal/sip:8100 at 10.160.1.212:5060 [BREAK]
2011-08-22 04:45:00.731677 [DEBUG] switch_core_state_machine.c:364 (sofia/internal/sip:8100 at 10.160.1.212:5060) State ROUTING going to sleep
2011-08-22 04:45:00.731677 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/sip:8100 at 10.160.1.212:5060) Running State Change CS_CONSUME_MEDIA
2011-08-22 04:45:00.731677 [DEBUG] switch_core_state_machine.c:383 (sofia/internal/sip:8100 at 10.160.1.212:5060) State CONSUME_MEDIA
2011-08-22 04:45:00.731677 [DEBUG] switch_core_state_machine.c:383 (sofia/internal/sip:8100 at 10.160.1.212:5060) State CONSUME_MEDIA going to sleep
2011-08-22 04:45:00.741675 [DEBUG] sofia.c:4761 Channel sofia/internal/sip:8100 at 10.160.1.212:5060 entering state [terminated][415]
2011-08-22 04:45:00.741675 [DEBUG] switch_channel.c:2562 (sofia/internal/sip:8100 at 10.160.1.212:5060) Callstate Change RINGING -> HANGUP
2011-08-22 04:45:00.741675 [NOTICE] sofia.c:5407 Hangup sofia/internal/sip:8100 at 10.160.1.212:5060 [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
2011-08-22 04:45:00.741675 [DEBUG] switch_channel.c:2578 Send signal sofia/internal/sip:8100 at 10.160.1.212:5060 [KILL]
2011-08-22 04:45:00.741675 [DEBUG] switch_core_session.c:1114 Send signal sofia/internal/sip:8100 at 10.160.1.212:5060 [BREAK]
2011-08-22 04:45:00.741675 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/sip:8100 at 10.160.1.212:5060) Running State Change CS_HANGUP
2011-08-22 04:45:00.741675 [DEBUG] switch_ivr_originate.c:3299 Originate Resulted in Error Cause: 79 [SERVICE_NOT_IMPLEMENTED]
2011-08-22 04:45:00.741675 [ERR] switch_ivr_originate.c:2447 Cannot create outgoing channel of type [user] cause: [SERVICE_NOT_IMPLEMENTED]
2011-08-22 04:45:00.741675 [DEBUG] switch_ivr_originate.c:3299 Originate Resulted in Error Cause: 79 [SERVICE_NOT_IMPLEMENTED]
2011-08-22 04:45:00.741675 [INFO] mod_dptools.c:2647 Originate Failed.  Cause: SERVICE_NOT_IMPLEMENTED
EXECUTE sofia/external/7777 answer()
2011-08-22 04:45:00.742676 [DEBUG] sofia_glue.c:3022 AUDIO RTP [sofia/external/7777] 10.160.0.3 port 23546 -> 10.160.0.205 port 5200 codec: 8 ms: 20
2011-08-22 04:45:00.742676 [DEBUG] switch_rtp.c:1623 Starting timer [soft] 160 bytes per 20ms
2011-08-22 04:45:00.742676 [DEBUG] switch_core_state_machine.c:565 (sofia/internal/sip:8100 at 10.160.1.212:5060) State HANGUP
2011-08-22 04:45:00.742676 [DEBUG] mod_sofia.c:451 sofia/internal/sip:8100 at 10.160.1.212:5060 Overriding SIP cause 501 with 415 from the other leg
2011-08-22 04:45:00.742676 [DEBUG] mod_sofia.c:457 Channel sofia/internal/sip:8100 at 10.160.1.212:5060 hanging up, cause: SERVICE_NOT_IMPLEMENTED
2011-08-22 04:45:00.743676 [DEBUG] switch_core_state_machine.c:46 sofia/internal/sip:8100 at 10.160.1.212:5060 Standard HANGUP, cause: SERVICE_NOT_IMPLEMENTED
2011-08-22 04:45:00.743676 [DEBUG] switch_core_state_machine.c:565 (sofia/internal/sip:8100 at 10.160.1.212:5060) State HANGUP going to sleep
2011-08-22 04:45:00.743676 [DEBUG] switch_core_state_machine.c:356 (sofia/internal/sip:8100 at 10.160.1.212:5060) State Change CS_HANGUP -> CS_REPORTING
2011-08-22 04:45:00.743676 [DEBUG] switch_core_session.c:1114 Send signal sofia/internal/sip:8100 at 10.160.1.212:5060 [BREAK]
2011-08-22 04:45:00.743676 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/sip:8100 at 10.160.1.212:5060) Running State Change CS_REPORTING
2011-08-22 04:45:00.743676 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/sip:8100 at 10.160.1.212:5060) State REPORTING
2011-08-22 04:45:00.743676 [DEBUG] switch_core_state_machine.c:53 sofia/internal/sip:8100 at 10.160.1.212:5060 Standard REPORTING, cause: SERVICE_NOT_IMPLEMENTED
2011-08-22 04:45:00.743676 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/sip:8100 at 10.160.1.212:5060) State REPORTING going to sleep
2011-08-22 04:45:00.743676 [DEBUG] switch_core_state_machine.c:350 (sofia/internal/sip:8100 at 10.160.1.212:5060) State Change CS_REPORTING -> CS_DESTROY
2011-08-22 04:45:00.743676 [DEBUG] switch_core_session.c:1114 Send signal sofia/internal/sip:8100 at 10.160.1.212:5060 [BREAK]
2011-08-22 04:45:00.743676 [DEBUG] switch_core_session.c:1286 Session 530 (sofia/internal/sip:8100 at 10.160.1.212:5060) Locked, Waiting on external entities
2011-08-22 04:45:00.743676 [NOTICE] switch_core_session.c:1304 Session 530 (sofia/internal/sip:8100 at 10.160.1.212:5060) Ended
2011-08-22 04:45:00.743676 [NOTICE] switch_core_session.c:1306 Close Channel sofia/internal/sip:8100 at 10.160.1.212:5060 [CS_DESTROY]
2011-08-22 04:45:00.743676 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/sip:8100 at 10.160.1.212:5060) Callstate Change HANGUP -> DOWN
2011-08-22 04:45:00.743676 [DEBUG] switch_core_state_machine.c:457 (sofia/internal/sip:8100 at 10.160.1.212:5060) Running State Change CS_DESTROY
2011-08-22 04:45:00.743676 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/sip:8100 at 10.160.1.212:5060) State DESTROY
2011-08-22 04:45:00.743676 [DEBUG] mod_sofia.c:362 sofia/internal/sip:8100 at 10.160.1.212:5060 SOFIA DESTROY
2011-08-22 04:45:00.743676 [DEBUG] switch_core_state_machine.c:60 sofia/internal/sip:8100 at 10.160.1.212:5060 Standard DESTROY
2011-08-22 04:45:00.743676 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/sip:8100 at 10.160.1.212:5060) State DESTROY going to sleep
2011-08-22 04:45:00.743676 [DEBUG] sofia_glue.c:3284 Set 2833 dtmf send payload to 101
2011-08-22 04:45:00.744676 [DEBUG] sofia_glue.c:3289 Set 2833 dtmf receive payload to 101
2011-08-22 04:45:00.744676 [DEBUG] mod_sofia.c:681 Local SDP sofia/external/7777:
v=0
o=FreeSWITCH 1313953954 1313953955 IN IP4 10.160.0.3
s=FreeSWITCH
c=IN IP4 10.160.0.3
t=0 0
m=audio 23546 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2011-08-22 04:45:00.744676 [DEBUG] switch_core_session.c:707 Send signal sofia/external/7777 [BREAK]
2011-08-22 04:45:00.744676 [DEBUG] switch_channel.c:2829 (sofia/external/7777) Callstate Change RINGING -> ACTIVE
2011-08-22 04:45:00.744676 [NOTICE] mod_dptools.c:930 Channel [sofia/external/7777] has been answered
2011-08-22 04:45:00.744676 [DEBUG] sofia.c:4761 Channel sofia/external/7777 entering state [completed][200]
EXECUTE sofia/external/7777 sleep(1000)
2011-08-22 04:45:00.746677 [DEBUG] sofia.c:4761 Channel sofia/external/7777 entering state [ready][200]
2011-08-22 04:45:00.945642 [DEBUG] switch_rtp.c:3082 Correct ip/port confirmed.
EXECUTE sofia/external/7777 voicemail(default 10.160.0.3 8100)
2011-08-22 04:45:01.866492 [DEBUG] switch_ivr_play_say.c:67 No language specified - Using [ru]
2011-08-22 04:45:01.877489 [DEBUG] switch_ivr_play_say.c:244 Handle play-file:[voicemail/vm-person.wav] (ru:ru)
2011-08-22 04:45:01.877489 [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
2011-08-22 04:45:02.220434 [DEBUG] switch_channel.c:2562 (sofia/external/7777) Callstate Change ACTIVE -> HANGUP
2011-08-22 04:45:02.220434 [NOTICE] sofia.c:538 Hangup sofia/external/7777 [CS_EXECUTE] [NORMAL_CLEARING]
2011-08-22 04:45:02.220434 [DEBUG] switch_channel.c:2578 Send signal sofia/external/7777 [KILL]
2011-08-22 04:45:02.221602 [DEBUG] switch_core_session.c:1114 Send signal sofia/external/7777 [BREAK]
2011-08-22 04:45:02.226433 [DEBUG] switch_ivr_play_say.c:1649 done playing file
2011-08-22 04:45:02.326421 [DEBUG] switch_ivr_play_say.c:244 Handle say:[8100] (ru:ru)
2011-08-22 04:45:02.428408 [DEBUG] switch_core_session.c:2057 sofia/external/7777 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
2011-08-22 04:45:02.428408 [DEBUG] switch_core_state_machine.c:371 (sofia/external/7777) State EXECUTE going to sleep
2011-08-22 04:45:02.428408 [DEBUG] switch_core_state_machine.c:325 (sofia/external/7777) Running State Change CS_HANGUP
2011-08-22 04:45:02.429464 [DEBUG] switch_core_state_machine.c:565 (sofia/external/7777) State HANGUP
2011-08-22 04:45:02.429464 [DEBUG] mod_sofia.c:451 sofia/external/7777 Overriding SIP cause 480 with 200 from the other leg
2011-08-22 04:45:02.429464 [DEBUG] mod_sofia.c:457 Channel sofia/external/7777 hanging up, cause: NORMAL_CLEARING
2011-08-22 04:45:02.429464 [DEBUG] switch_core_state_machine.c:46 sofia/external/7777 Standard HANGUP, cause: NORMAL_CLEARING
2011-08-22 04:45:02.429464 [DEBUG] switch_core_state_machine.c:565 (sofia/external/7777) State HANGUP going to sleep
2011-08-22 04:45:02.429464 [DEBUG] switch_core_state_machine.c:356 (sofia/external/7777) State Change CS_HANGUP -> CS_REPORTING
2011-08-22 04:45:02.429464 [DEBUG] switch_core_session.c:1114 Send signal sofia/external/7777 [BREAK]
2011-08-22 04:45:02.429464 [DEBUG] switch_core_state_machine.c:325 (sofia/external/7777) Running State Change CS_REPORTING
2011-08-22 04:45:02.429464 [DEBUG] switch_core_state_machine.c:625 (sofia/external/7777) State REPORTING
2011-08-22 04:45:02.473396 [DEBUG] switch_core_state_machine.c:53 sofia/external/7777 Standard REPORTING, cause: NORMAL_CLEARING
2011-08-22 04:45:02.473396 [DEBUG] switch_core_state_machine.c:625 (sofia/external/7777) State REPORTING going to sleep
2011-08-22 04:45:02.473396 [DEBUG] switch_core_state_machine.c:350 (sofia/external/7777) State Change CS_REPORTING -> CS_DESTROY
2011-08-22 04:45:02.473396 [DEBUG] switch_core_session.c:1114 Send signal sofia/external/7777 [BREAK]
2011-08-22 04:45:02.473396 [DEBUG] switch_core_session.c:1286 Session 529 (sofia/external/7777) Locked, Waiting on external entities
2011-08-22 04:45:02.473396 [NOTICE] switch_core_session.c:1304 Session 529 (sofia/external/7777) Ended
2011-08-22 04:45:02.473396 [NOTICE] switch_core_session.c:1306 Close Channel sofia/external/7777 [CS_DESTROY]
2011-08-22 04:45:02.473396 [DEBUG] switch_core_state_machine.c:454 (sofia/external/7777) Callstate Change HANGUP -> DOWN


-- 
Best regards,
 sucsession                          mailto:norstar at bigmir.net



From royce3 at gmail.com  Sun Aug 21 10:52:37 2011
From: royce3 at gmail.com (Royce Mitchell III)
Date: Sun, 21 Aug 2011 01:52:37 -0500
Subject: [Freeswitch-users] originate through dialplan?
Message-ID: <CAHwzD9kT4=a4fyp6dyYJtWb0KOPhwkXYSQtHna8BRW_TMZ7ngA@mail.gmail.com>

Hi,

I need to be able to originate a call on behalf of an agent through
the dialplan, and I cannot seem to get it to work.

I've tried "uuid_transfer <5021's uuid> 98885551212 xml default", but
it appears to destroy "transfer_on_bridge" and leaves my agent session
in a weird state when the call ends.

I've tried "originate user/5021 &bridge(98885551212) xml default", but
it rings 5021 instead of bridging the existing 5021 session to the
outbound call.

I can't seem to get anything like this to work "originate 98885551212
&bridge(user/5021)" which is what I think I really need.

The only solution I've been able to think of would involve using
transfer and then trying to catch the call termination then transfer
the agent back into the callcenter standby mode, but this feels like
it would be error-prone.

Would it be too difficult to implement a new ivr function to bridge an
existing uuid to a transfer? Maybe something like "uuid_bridgeto uuid
exten dialplan context"? Is this an outlandish request or is there
some other way of accomplishing this?

Thanks in advance

-- 
There's a fine line between genius and insanity. I like to use it for
dental floss.


From bryan at bryanlemon.com  Mon Aug 22 19:20:38 2011
From: bryan at bryanlemon.com (Bryan Lemon)
Date: Mon, 22 Aug 2011 11:20:38 -0400
Subject: [Freeswitch-users] Question about ext-rtp-ip and ext-sip-ip
In-Reply-To: <CAEjbKNT3oQJZ34FZFuq7JRxRV7zo7GaHdaybBRYYrSCKSTY_CQ@mail.gmail.com>
References: <CAEjbKNQ3tgWrVrr5uiTVKzv53avDLp3cRoavLuZsOiJfUaW_fg@mail.gmail.com>
	<65727391-DF08-4074-BB7F-BDB766DF7942@freeswitch.org>
	<CAEjbKNSGi2prQqoeLSGbKjtYbqOsD7-hggPeEUcVEWPgtCp2UQ@mail.gmail.com>
	<CAEjbKNTt8Fd=Q6TnGz407j5N4zfe4X8FSLpf2ttmpUydT9YJpA@mail.gmail.com>
	<7C7183C2-3601-47D4-B8DE-D9E292B592D3@freeswitch.org>
	<CAEjbKNT3oQJZ34FZFuq7JRxRV7zo7GaHdaybBRYYrSCKSTY_CQ@mail.gmail.com>
Message-ID: <CAEjbKNTmqOUa1VyLTh74JfwVVzjxO9Tui+OLGgJPNxxo3wV1Cw@mail.gmail.com>

Any other input/ideas on this one?

Thank you,
Bryan Lemon
(302) 648-2747



On Wed, Aug 17, 2011 at 23:08, Bryan Lemon <bryan at bryanlemon.com> wrote:

> I tried it, and it still does not work.
>
> Relevant details:
> freeswitch at MEDIAPC> sofia status profile external
>
>
> =================================================================================================
> Name             external
> Domain Name       N/A
> Auto-NAT         true
> DBName           sofia_reg_external
> Pres Hosts
> Dialplan         XML
> Context           public
> Challenge Realm   auto_to
> RTP-IP           10.0.10.144
> Ext-RTP-IP       204.111.*.*
> SIP-IP           10.0.10.144
> Ext-SIP-IP       204.111.*.*
> URL               sip:mod_sofia at 10.0.10.144:5080
> BIND-URL         sip:mod_sofia at 10.0.10.144:5080
> HOLD-MUSIC       local_stream://moh
>
>
> external.xml
> <profile name="external">
>   <gateways>
>     <X-PRE-PROCESS cmd="include" data="external/*.xml"/>
>   </gateways>
>
>   <aliases>
>     <!--
>
>
>     <alias name="outbound"/>
>
>
>     <alias name="nat"/>
>
>
>     -->
>   </aliases>
>
>   <domains>
>     <domain name="all" alias="false" parse="true"/>
>   </domains>
>
>   <settings>
>     <param name="debug" value="0"/>
>     <param name="sip-trace" value="no"/>
>     <param name="rfc2833-pt" value="101"/>
>     <param name="sip-port" value="$${external_sip_port}"/>
>     <param name="dialplan" value="XML"/>
>     <param name="context" value="public"/>
>     <param name="dtmf-duration" value="2000"/>
>     <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
>     <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
>     <param name="hold-music" value="$${hold_music}"/>
>     <param name="rtp-timer-name" value="soft"/>
>     <param name="local-network-acl" value="localnet.auto"/>
>     <param name="manage-presence" value="false"/>
>     <!--<param name="aggressive-nat-detection" value="true"/>-->
>     <param name="inbound-codec-negotiation" value="generous"/>
>     <param name="nonce-ttl" value="60"/>
>     <param name="auth-calls" value="false"/>
>     <param name="rtp-ip" value="$${local_ip_v4}"/>
>     <param name="sip-ip" value="$${local_ip_v4}"/>
>     <param name="ext-rtp-ip" value="autonat:204.111.187.118"/>
>     <param name="ext-sip-ip" value="autonat:204.111.187.118"/>
>      <param name="rtp-timeout-sec" value="300"/>
>     <param name="rtp-hold-timeout-sec" value="1800"/>
>     <param name="tls" value="$${external_ssl_enable}"/>
>     <param name="tls-bind-params" value="transport=tls"/>
>     <param name="tls-sip-port" value="$${external_tls_port}"/>
>     <param name="tls-cert-dir" value="$${external_ssl_dir}"/>
>     <param name="tls-version" value="$${sip_tls_version}"/>
>
>   </settings>
> </profile>
>
> Thank you,
> Bryan Lemon
> (302) 648-2747
>
>
>
> On Wed, Aug 17, 2011 at 12:55, Brian West <brian at freeswitch.org> wrote:
>
>> You need to change these to be "autonat:x.x.x.x"  and specify the IP to
>> use.
>>
>> /b
>>
>> On Aug 17, 2011, at 10:10 AM, Bryan Lemon wrote:
>>
>> >    <param name="ext-rtp-ip" value="auto-nat"/>
>> >    <param name="ext-sip-ip" value="auto-nat"/>
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110822/b9ff70b8/attachment.html 

From johns1433 at gmail.com  Mon Aug 22 19:22:31 2011
From: johns1433 at gmail.com (John Smith)
Date: Mon, 22 Aug 2011 17:22:31 +0200
Subject: [Freeswitch-users] mod_rtmp & Flex client
Message-ID: <CA+9AvwQK2_DOut4wg4+uHCJ=dwYDT+WAx946pZdd5Jw2d6AR9w@mail.gmail.com>

Thanks a lot for your answer. It is working now with the latest flex client
in git.
John

2011/8/22 Mathieu Rene <mrene_lists at avgs.ca>

> Hi,
>
> It seems that the jquery.tmpl version in-tree is using a different function
> than the one I had when developing the client (they basically were using
> render() and now they call it tmpl()).
>
> Fixed in 3d3e5c6.
>
>   Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
>   On 2011-08-22, at 4:01 PM, John Smith wrote:
>
>   Hi All,
>
> I have installed a recent version of FS (git-cd31633 2011-08-17) in order
> to use mod_rtmp.
> The rtmp connexion to the server works fine but I don?t manage to add a sip
> call (?+New Call? button).
> When I fill the sip address and click ?call? I got an error from the Flash
> Player: flash.net.NetConnection is not able to call the callback function
> onMakeCall (this a javascript function defined in freeswitch.html).
>
> I made several tests in order to find the source of the problem and it
> seems to come from the add_call function called within onMakeCall. Without
> the call to add_call onMakeCall works fine.
> I tried several browsers, several version of the Flash Player (above 10)
> and the problem occurred in all cases.
>
> I?m now running short of ideas.
> Has someone some suggestions of what I could try to investigate and solve
> this issue?
>
> Regards,
>
> John
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110822/9bcbf8e8/attachment-0001.html 

From curriegrad2004 at gmail.com  Mon Aug 22 20:46:51 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Mon, 22 Aug 2011 09:46:51 -0700
Subject: [Freeswitch-users] "make current" does not work with Fedora 14
In-Reply-To: <4E52488C.4010306@hw.ac.uk>
References: <CAEVfcaY3kKucJwuURcOAYuDCJ+RseEnJZFpw0KhJ75sPiqSqmQ@mail.gmail.com>
	<4E52488C.4010306@hw.ac.uk>
Message-ID: <CAMWiMy9vvQpMoCNY3HWaT7Km8nj0x2piCAD1oFWcMOnGjXZ-=g@mail.gmail.com>

okay, try a git reset --hard, rerun ./bootstrap.sh and rerun the
configure and rebuild it from that point.

On Mon, Aug 22, 2011 at 5:16 AM, xl127 <x.liu at hw.ac.uk> wrote:
> Hi, Before I submit an issue to Jira, I tried "make current" on Fedora
> 14 and I got some errors.
> I first tried "make current", noticed some errors then tried "make
> current > makeCurrentOutput.txt".
>
> I pasted the contents of the file makeCurrentOutput.txt to the pastebin
> via user "xing"
>
> The console outputs are as follows:
>
> [root at localhost freeswitch]# make current > makeCurrentOutput.txt
> grep: ../../../..//src/include/switch_version.h: No such file or directory
> grep: ../../../..//src/include/switch_version.h: No such file or directory
> grep: ../../../..//src/include/switch_version.h: No such file or directory
> grep: ../../../..//src/include/switch_version.h: No such file or directory
> grep: ../../../..//src/include/switch_version.h: No such file or directory
> grep: ../../../..//src/include/switch_version.h: No such file or directory
> /usr/lib/libnss3.so: undefined reference to `PR_FindSymbol'
> /usr/lib/libnss3.so: undefined reference to `PR_RWLock_Rlock'
> /usr/lib/libssl3.so: undefined reference to `PR_OpenAnonFileMap'
> /usr/lib/libssl3.so: undefined reference to `PR_UnloadLibrary'
> /usr/lib/libnss3.so: undefined reference to `PL_InitArenaPool'
> /usr/lib/libnss3.so: undefined reference to `PR_NewRWLock'
> /usr/lib/libnss3.so: undefined reference to `PR_RWLock_Wlock'
> /usr/lib/libnss3.so: undefined reference to `PR_LoadLibrary'
> /usr/lib/libldap_r-2.4.so.2: undefined reference to `PR_GetEnv'
> /usr/lib/libssl3.so: undefined reference to `PR_LoadLibraryWithFlags'
> /usr/lib/libnssutil3.so: undefined reference to `PL_ClearArenaPool'
> /usr/lib/libnss3.so: undefined reference to `PR_DestroyRWLock'
> /usr/lib/libnss3.so: undefined reference to `PR_NewTCPSocket'
> /usr/lib/libldap_r-2.4.so.2: undefined reference to `PR_GetLibraryName'
> /usr/lib/libssl3.so: undefined reference to `PR_ExportFileMapAsString'
> /usr/lib/libssl3.so: undefined reference to `PR_GetLibraryFilePathname'
> /usr/lib/libssl3.so: undefined reference to `PR_FindFunctionSymbol'
> /usr/lib/libsmime3.so: undefined reference to `PL_NewHashTable'
> /usr/lib/libldap_r-2.4.so.2: undefined reference to `PR_ErrorToString'
> /usr/lib/libnss3.so: undefined reference to `PR_RWLock_Unlock'
> /usr/lib/libssl3.so: undefined reference to `PR_ImportFileMapFromString'
> /usr/lib/libldap_r-2.4.so.2: undefined reference to
> `PR_GetDirectorySeparator'
> collect2: ld returned 1 exit status
> make[3]: *** [freeswitch] Error 1
> libtool: link: warning: `-version-info/-version-number' is ignored for
> convenience libraries
> make[2]: *** [all-recursive] Error 1
> make[1]: *** [all] Error 2
> make: *** [current] Error 2
>
>
> Now I have to do a fresh FS installation again.
>
> Xing
>
>
>
>
>
> --
> Heriot-Watt University is a Scottish charity
> registered under charity number SC000278.
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From x.liu at hw.ac.uk  Mon Aug 22 21:03:32 2011
From: x.liu at hw.ac.uk (xl127)
Date: Mon, 22 Aug 2011 18:03:32 +0100
Subject: [Freeswitch-users] "make current" does not work with Fedora 14
In-Reply-To: <CAMWiMy9vvQpMoCNY3HWaT7Km8nj0x2piCAD1oFWcMOnGjXZ-=g@mail.gmail.com>
References: <CAEVfcaY3kKucJwuURcOAYuDCJ+RseEnJZFpw0KhJ75sPiqSqmQ@mail.gmail.com>	<4E52488C.4010306@hw.ac.uk>
	<CAMWiMy9vvQpMoCNY3HWaT7Km8nj0x2piCAD1oFWcMOnGjXZ-=g@mail.gmail.com>
Message-ID: <4E528BE4.8040400@hw.ac.uk>

I did a fresh installation after the failure of "make current" via

git clone git://git.freeswitch.org/freeswitch.git, ./bootstrap.sh, ./configure --without-libcurl, make&&  make install.



Thanks anyway!

Xing


On 22/08/11 17:46, curriegrad2004 wrote:
> okay, try a git reset --hard, rerun ./bootstrap.sh and rerun the
> configure and rebuild it from that point.
>
> On Mon, Aug 22, 2011 at 5:16 AM, xl127<x.liu at hw.ac.uk>  wrote:
>> Hi, Before I submit an issue to Jira, I tried "make current" on Fedora
>> 14 and I got some errors.
>> I first tried "make current", noticed some errors then tried "make
>> current>  makeCurrentOutput.txt".
>>
>> I pasted the contents of the file makeCurrentOutput.txt to the pastebin
>> via user "xing"
>>
>> The console outputs are as follows:
>>
>> [root at localhost freeswitch]# make current>  makeCurrentOutput.txt
>> grep: ../../../..//src/include/switch_version.h: No such file or directory
>> grep: ../../../..//src/include/switch_version.h: No such file or directory
>> grep: ../../../..//src/include/switch_version.h: No such file or directory
>> grep: ../../../..//src/include/switch_version.h: No such file or directory
>> grep: ../../../..//src/include/switch_version.h: No such file or directory
>> grep: ../../../..//src/include/switch_version.h: No such file or directory
>> /usr/lib/libnss3.so: undefined reference to `PR_FindSymbol'
>> /usr/lib/libnss3.so: undefined reference to `PR_RWLock_Rlock'
>> /usr/lib/libssl3.so: undefined reference to `PR_OpenAnonFileMap'
>> /usr/lib/libssl3.so: undefined reference to `PR_UnloadLibrary'
>> /usr/lib/libnss3.so: undefined reference to `PL_InitArenaPool'
>> /usr/lib/libnss3.so: undefined reference to `PR_NewRWLock'
>> /usr/lib/libnss3.so: undefined reference to `PR_RWLock_Wlock'
>> /usr/lib/libnss3.so: undefined reference to `PR_LoadLibrary'
>> /usr/lib/libldap_r-2.4.so.2: undefined reference to `PR_GetEnv'
>> /usr/lib/libssl3.so: undefined reference to `PR_LoadLibraryWithFlags'
>> /usr/lib/libnssutil3.so: undefined reference to `PL_ClearArenaPool'
>> /usr/lib/libnss3.so: undefined reference to `PR_DestroyRWLock'
>> /usr/lib/libnss3.so: undefined reference to `PR_NewTCPSocket'
>> /usr/lib/libldap_r-2.4.so.2: undefined reference to `PR_GetLibraryName'
>> /usr/lib/libssl3.so: undefined reference to `PR_ExportFileMapAsString'
>> /usr/lib/libssl3.so: undefined reference to `PR_GetLibraryFilePathname'
>> /usr/lib/libssl3.so: undefined reference to `PR_FindFunctionSymbol'
>> /usr/lib/libsmime3.so: undefined reference to `PL_NewHashTable'
>> /usr/lib/libldap_r-2.4.so.2: undefined reference to `PR_ErrorToString'
>> /usr/lib/libnss3.so: undefined reference to `PR_RWLock_Unlock'
>> /usr/lib/libssl3.so: undefined reference to `PR_ImportFileMapFromString'
>> /usr/lib/libldap_r-2.4.so.2: undefined reference to
>> `PR_GetDirectorySeparator'
>> collect2: ld returned 1 exit status
>> make[3]: *** [freeswitch] Error 1
>> libtool: link: warning: `-version-info/-version-number' is ignored for
>> convenience libraries
>> make[2]: *** [all-recursive] Error 1
>> make[1]: *** [all] Error 2
>> make: *** [current] Error 2
>>
>>
>> Now I have to do a fresh FS installation again.
>>
>> Xing
>>
>>
>>
>>
>>
>> --
>> Heriot-Watt University is a Scottish charity
>> registered under charity number SC000278.
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Heriot-Watt University is a Scottish charity
registered under charity number SC000278.



From brian at freeswitch.org  Mon Aug 22 21:27:34 2011
From: brian at freeswitch.org (Brian West)
Date: Mon, 22 Aug 2011 12:27:34 -0500
Subject: [Freeswitch-users] Question about ext-rtp-ip and ext-sip-ip
In-Reply-To: <CAEjbKNTmqOUa1VyLTh74JfwVVzjxO9Tui+OLGgJPNxxo3wV1Cw@mail.gmail.com>
References: <CAEjbKNQ3tgWrVrr5uiTVKzv53avDLp3cRoavLuZsOiJfUaW_fg@mail.gmail.com>
	<65727391-DF08-4074-BB7F-BDB766DF7942@freeswitch.org>
	<CAEjbKNSGi2prQqoeLSGbKjtYbqOsD7-hggPeEUcVEWPgtCp2UQ@mail.gmail.com>
	<CAEjbKNTt8Fd=Q6TnGz407j5N4zfe4X8FSLpf2ttmpUydT9YJpA@mail.gmail.com>
	<7C7183C2-3601-47D4-B8DE-D9E292B592D3@freeswitch.org>
	<CAEjbKNT3oQJZ34FZFuq7JRxRV7zo7GaHdaybBRYYrSCKSTY_CQ@mail.gmail.com>
	<CAEjbKNTmqOUa1VyLTh74JfwVVzjxO9Tui+OLGgJPNxxo3wV1Cw@mail.gmail.com>
Message-ID: <F3681824-2198-41B1-A825-5D9401EACD08@freeswitch.org>

What kind of nat / firewall are you traversing?

/b

On Aug 22, 2011, at 10:20 AM, Bryan Lemon wrote:

> Any other input/ideas on this one?
> 
> Thank you,
> Bryan Lemon
> (302) 648-2747
> 
> 



From bryan at bryanlemon.com  Mon Aug 22 21:48:33 2011
From: bryan at bryanlemon.com (Bryan Lemon)
Date: Mon, 22 Aug 2011 13:48:33 -0400
Subject: [Freeswitch-users] Question about ext-rtp-ip and ext-sip-ip
In-Reply-To: <F3681824-2198-41B1-A825-5D9401EACD08@freeswitch.org>
References: <CAEjbKNQ3tgWrVrr5uiTVKzv53avDLp3cRoavLuZsOiJfUaW_fg@mail.gmail.com>
	<65727391-DF08-4074-BB7F-BDB766DF7942@freeswitch.org>
	<CAEjbKNSGi2prQqoeLSGbKjtYbqOsD7-hggPeEUcVEWPgtCp2UQ@mail.gmail.com>
	<CAEjbKNTt8Fd=Q6TnGz407j5N4zfe4X8FSLpf2ttmpUydT9YJpA@mail.gmail.com>
	<7C7183C2-3601-47D4-B8DE-D9E292B592D3@freeswitch.org>
	<CAEjbKNT3oQJZ34FZFuq7JRxRV7zo7GaHdaybBRYYrSCKSTY_CQ@mail.gmail.com>
	<CAEjbKNTmqOUa1VyLTh74JfwVVzjxO9Tui+OLGgJPNxxo3wV1Cw@mail.gmail.com>
	<F3681824-2198-41B1-A825-5D9401EACD08@freeswitch.org>
Message-ID: <CAEjbKNQLzTB+d57BwpdrBdsVtabtZjKsf6yvEGZUiZX3wc3DrQ@mail.gmail.com>

Just a standard dd-wrt router.

Thank you,
Bryan Lemon
(302) 648-2747



On Mon, Aug 22, 2011 at 13:27, Brian West <brian at freeswitch.org> wrote:

> What kind of nat / firewall are you traversing?
>
> /b
>
> On Aug 22, 2011, at 10:20 AM, Bryan Lemon wrote:
>
> > Any other input/ideas on this one?
> >
> > Thank you,
> > Bryan Lemon
> > (302) 648-2747
> >
> >
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110822/72ecc440/attachment.html 

From lakersman2006 at yahoo.com  Mon Aug 22 22:36:55 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Mon, 22 Aug 2011 11:36:55 -0700 (PDT)
Subject: [Freeswitch-users] Freeswitch say digits too quick
Message-ID: <1314038215.46613.YahooMailNeo@web161002.mail.bf1.yahoo.com>

Is there a way to slow down the speed when using the say app to repeat digits like 12345 (one two three four five), right now it is a bit too fast.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110822/cae3dec8/attachment.html 

From anthony.minessale at gmail.com  Mon Aug 22 23:00:27 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 22 Aug 2011 14:00:27 -0500
Subject: [Freeswitch-users] Freeswitch say digits too quick
In-Reply-To: <1314038215.46613.YahooMailNeo@web161002.mail.bf1.yahoo.com>
References: <1314038215.46613.YahooMailNeo@web161002.mail.bf1.yahoo.com>
Message-ID: <CAKbxfG_eCtJrx7coQ1=qgEc7Kax+vz-ZekCJeeH9KXGoBTQ_UQ@mail.gmail.com>

if you are on latest, all you could do is modify mod_dptools.c line
3628, the 250 on that line is the ms  to delay between each new file
played by the file string .


On Mon, Aug 22, 2011 at 1:36 PM, Sam <lakersman2006 at yahoo.com> wrote:
> Is there a way to slow down the speed when using the say app to repeat
> digits like 12345 (one two three four five), right now it is a bit too fast.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From jack at livecall.com  Tue Aug 23 00:10:24 2011
From: jack at livecall.com (Jack)
Date: Mon, 22 Aug 2011 13:10:24 -0700
Subject: [Freeswitch-users] How to authencticate with mod_rtmp?
In-Reply-To: <CADDV4gzH8gW1cP_K3vMgOtouACRs4PNQSRKC9aVd-m0uwxTjQw@mail.gmail.com>
References: <CADDV4gzH8gW1cP_K3vMgOtouACRs4PNQSRKC9aVd-m0uwxTjQw@mail.gmail.com>
Message-ID: <4E52B7B0.4020204@livecall.com>

Hi Dmitry,
  Did you ever figure the authentication out?   I am experiencing the 
same thing now with the latest git.
Thanks,
  Jack

On 7/19/2011 4:53 AM, Dmitry Kravchenko wrote:
> Hi!
>
> It is said in book that user registry is single and centralized for 
> all components (p. 64).
>
> But while trying to login via FLEX telephone to default number of 1000 
> with password of 1234, I get "Authentication failed" message.
>
> In freeswitch CLI I see message
>
> 2011-07-19 15:46:51.473828 [WARNING] rtmp.c:227 Authentication failed 
> for 1000 at 192.168.10.196 <mailto:1000 at 192.168.10.196>
>
> where 192.168.10.196 is true IP of my freeswitch machine.
>
> Connection with X-Lite goes OK with message
>
> 2011-07-19 15:50:19.473829 [WARNING] sofia_reg.c:1337 SIP auth 
> challenge (REGISTER) on sofia profile 'internal' for 
> [1000 at 192.168.10.196 <mailto:1000 at 192.168.10.196>] from ip 192.168.10.56
>
> If I put wrong password into X-Lite then I get
>
> 2011-07-19 15:50:19.493774 [WARNING] sofia_reg.c:1295 SIP auth failure 
> (REGISTER) on sofia profile 'internal' for [1000 at 192.168.10.196 
> <mailto:1000 at 192.168.10.196>] from ip 192.168.10.56
>
> So my conclusion is that SIP phone is treates as internal profile by 
> default. Is it possible to set RTMP profiles as internal too?
>
> Or may be I should better create exteral accounts to authenticate with 
> RTMP?
>
> Thanks.
>
>
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110822/374ad043/attachment-0001.html 

From msc at freeswitch.org  Tue Aug 23 00:53:41 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 22 Aug 2011 13:53:41 -0700
Subject: [Freeswitch-users] intercept sethanguphook failure
In-Reply-To: <CAEW218J34LUCEgCLk59O0CfDZb54wQXbMkSfMpnZHF=81-FV5w@mail.gmail.com>
References: <CAEW218J34LUCEgCLk59O0CfDZb54wQXbMkSfMpnZHF=81-FV5w@mail.gmail.com>
Message-ID: <CAKzWOxXdR8=_L7Oo82NAKNQ6hJCnwXYwLEf0LCH0=pU5q1XA9A@mail.gmail.com>

I think you'll need to provide more information. Use
pastebin.freeswitch.orgto post your relevant dialplan. Also, pastebin
a console debug log of a call
from start to finish. Perhaps there's a clue as to why it is not executing
your hangup hook.

-MC

On Mon, Aug 22, 2011 at 12:46 AM, rentmycoder rentmycoder <
rentmycoder at gmail.com> wrote:

> Hi guys,
>
> I'm trying to make a callgroup intercept and need to make some tasks
> after the intercepted call hang's up...
> I've tried both intercept and uuid_bridge method, both works, but
> neither way the hanguphook does not get's called after intercept...
> settings: hangupafterbridge=true and continueonfail=true...
> Testing on win32 using latest GIT.
>
> lua script:
> ...
> session:setHangupHook("extensions_hanguphook");
> ...
> api:executeString("uuid_bridge " .. tostring(intercept_source) .. " "
> .. tostring(session.uuid));
> or this way:
> session:execute("intercept", intercept_source);
>
> I haven't looked into the source yet, any idea???
>
> I don't now it's a bug it it's my mistake...
>
> Thanks,
> John
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110822/740e169f/attachment.html 

From msc at freeswitch.org  Tue Aug 23 04:17:52 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 22 Aug 2011 17:17:52 -0700
Subject: [Freeswitch-users] originate through dialplan?
In-Reply-To: <CAHwzD9kT4=a4fyp6dyYJtWb0KOPhwkXYSQtHna8BRW_TMZ7ngA@mail.gmail.com>
References: <CAHwzD9kT4=a4fyp6dyYJtWb0KOPhwkXYSQtHna8BRW_TMZ7ngA@mail.gmail.com>
Message-ID: <CAKzWOxW8VXUo04LikzVhaKw6WZ3DNiBuF=qTFbhAXmiA4135OA@mail.gmail.com>

Royce,

Did I see you on IRC today? I think maybe you got this answered. If not, may
I ask what the ultimate goal is here? Are you attempting to bridge an
outbound leg to an existing call leg who is an agent of some sort? What kind
of agent is it? And what do you want to have happen with that agent when the
bridge is done?

-MC

On Sat, Aug 20, 2011 at 11:52 PM, Royce Mitchell III <royce3 at gmail.com>wrote:

> Hi,
>
> I need to be able to originate a call on behalf of an agent through
> the dialplan, and I cannot seem to get it to work.
>
> I've tried "uuid_transfer <5021's uuid> 98885551212 xml default", but
> it appears to destroy "transfer_on_bridge" and leaves my agent session
> in a weird state when the call ends.
>
> I've tried "originate user/5021 &bridge(98885551212) xml default", but
> it rings 5021 instead of bridging the existing 5021 session to the
> outbound call.
>
> I can't seem to get anything like this to work "originate 98885551212
> &bridge(user/5021)" which is what I think I really need.
>
> The only solution I've been able to think of would involve using
> transfer and then trying to catch the call termination then transfer
> the agent back into the callcenter standby mode, but this feels like
> it would be error-prone.
>
> Would it be too difficult to implement a new ivr function to bridge an
> existing uuid to a transfer? Maybe something like "uuid_bridgeto uuid
> exten dialplan context"? Is this an outlandish request or is there
> some other way of accomplishing this?
>
> Thanks in advance
>
> --
> There's a fine line between genius and insanity. I like to use it for
> dental floss.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110822/81e9a169/attachment.html 

From msc at freeswitch.org  Tue Aug 23 04:21:11 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 22 Aug 2011 17:21:11 -0700
Subject: [Freeswitch-users] regex help
In-Reply-To: <CAEVfcaY3kKucJwuURcOAYuDCJ+RseEnJZFpw0KhJ75sPiqSqmQ@mail.gmail.com>
References: <CAEVfcaY3kKucJwuURcOAYuDCJ+RseEnJZFpw0KhJ75sPiqSqmQ@mail.gmail.com>
Message-ID: <CAKzWOxVUL=FPA5QEmL4F=2tncum19RfB=sLaDk=DyYjJt0YZAQ@mail.gmail.com>

Did you figure this one out yet? How do you do an empty string? Just have
the caller press # without dialing any digits?

-MC

On Sat, Aug 20, 2011 at 7:40 AM, babak yakhchali <babak.freeswitch at gmail.com
> wrote:

> Hi
> How can I use playandgetdigits to get 0 or 8 numbers? I mean it should be
> an empty string or 8digits long.
> I tried this
> play_and_get_digits 0 8 3 7000 # file1 file2  number ^$|^d{{8}}$
> but it's not passing empty strings
> thanx
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110822/d9662a2b/attachment.html 

From msc at freeswitch.org  Tue Aug 23 04:27:33 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 22 Aug 2011 17:27:33 -0700
Subject: [Freeswitch-users] freeswitch php
In-Reply-To: <1313970987.96106.YahooMailNeo@web161008.mail.bf1.yahoo.com>
References: <1313970987.96106.YahooMailNeo@web161008.mail.bf1.yahoo.com>
Message-ID: <CAKzWOxWTGEYrAwHz20QO0JAYSk7oc9DyLOkVXiJ1=u8v4AAkpA@mail.gmail.com>

This probably isn't the most efficient way of using the event socket. In
fact, I'm not sure if this is anything more than a proof of concept. You are
MUCH better off using the ESL lib with PHP and taking advantage of the
abstraction you get.

-MC

On Sun, Aug 21, 2011 at 4:56 PM, Sam <lakersman2006 at yahoo.com> wrote:

> Does anyone know how to retrieve channel variables (ie. uuid, etc.) using
> the php example that was shown in the wiki below?
>
> #!/usr/bin/php -q
>
> <?php
>
> // set a couple of things so we dont kill the system
> ob_implicit_flush(true);
> set_time_limit(30);
>
> // Open stdin so we can read the data in
> $in = fopen("php://stdin", "r");
>
> // Connect
> echo "connect\n\n";
>
> // Answer
> echo "sendmsg\n";
> echo "call-command: execute\n";
> echo "execute-app-name: answer\n\n";
>
> // Play a prompt
> echo "sendmsg\n";
> echo "call-command: execute\n";
> echo "execute-app-name: playback\n";
> echo "execute-app-arg: /usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav\n\n";
>
> // Wait
> sleep(5);
>
> // Hangup
> echo "sendmsg\n";
> echo "call-command: hangup\n\n";
>
> fclose($in);
>
> ?>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110822/70083bb6/attachment.html 

From royce3 at gmail.com  Tue Aug 23 04:43:32 2011
From: royce3 at gmail.com (Royce Mitchell III)
Date: Mon, 22 Aug 2011 19:43:32 -0500
Subject: [Freeswitch-users] originate through dialplan?
In-Reply-To: <CAKzWOxW8VXUo04LikzVhaKw6WZ3DNiBuF=qTFbhAXmiA4135OA@mail.gmail.com>
References: <CAHwzD9kT4=a4fyp6dyYJtWb0KOPhwkXYSQtHna8BRW_TMZ7ngA@mail.gmail.com>
	<CAKzWOxW8VXUo04LikzVhaKw6WZ3DNiBuF=qTFbhAXmiA4135OA@mail.gmail.com>
Message-ID: <CAHwzD9na+VfQwNc6wVSVrYUp3pnfad_xU-sbRcC6jj4e7j-XPg@mail.gmail.com>

I didn't get an answer on IRC, but I did eventually figure it out.

What I didn't understand is that uuid_transfer does create a bridge
and it does honor the transfer_after_bridge setting. What I discovered
was that I was having a NAT issue and that was what was actually
causing the call to end up in limbo.

The way I discovered this was by doing a "uuid_getvar <uuid>
transfer_after_bridge". I noticed that even after the transfer, the
setting was still in-tact.

I then decided to transfer to a local extension instead of an outside
destination and everything behaved perfectly.

Thanks for the reply :)

On Mon, Aug 22, 2011 at 7:17 PM, Michael Collins <msc at freeswitch.org> wrote:
> Royce,
> Did I see you on IRC today? I think maybe you got this answered. If not, may
> I ask what the ultimate goal is here? Are you attempting to bridge an
> outbound leg to an existing call leg who is an agent of some sort? What kind
> of agent is it? And what do you want to have happen with that agent when the
> bridge is done?
> -MC
> On Sat, Aug 20, 2011 at 11:52 PM, Royce Mitchell III <royce3 at gmail.com>
> wrote:
>>
>> Hi,
>>
>> I need to be able to originate a call on behalf of an agent through
>> the dialplan, and I cannot seem to get it to work.
>>
>> I've tried "uuid_transfer <5021's uuid> 98885551212 xml default", but
>> it appears to destroy "transfer_on_bridge" and leaves my agent session
>> in a weird state when the call ends.
>>
>> I've tried "originate user/5021 &bridge(98885551212) xml default", but
>> it rings 5021 instead of bridging the existing 5021 session to the
>> outbound call.
>>
>> I can't seem to get anything like this to work "originate 98885551212
>> &bridge(user/5021)" which is what I think I really need.
>>
>> The only solution I've been able to think of would involve using
>> transfer and then trying to catch the call termination then transfer
>> the agent back into the callcenter standby mode, but this feels like
>> it would be error-prone.
>>
>> Would it be too difficult to implement a new ivr function to bridge an
>> existing uuid to a transfer? Maybe something like "uuid_bridgeto uuid
>> exten dialplan context"? Is this an outlandish request or is there
>> some other way of accomplishing this?
>>
>> Thanks in advance
>>
>> --
>> There's a fine line between genius and insanity. I like to use it for
>> dental floss.
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
There's a fine line between genius and insanity. I like to use it for
dental floss.


From lakersman2006 at yahoo.com  Tue Aug 23 04:54:43 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Mon, 22 Aug 2011 17:54:43 -0700 (PDT)
Subject: [Freeswitch-users] freeswitch php
In-Reply-To: <CAKzWOxWTGEYrAwHz20QO0JAYSk7oc9DyLOkVXiJ1=u8v4AAkpA@mail.gmail.com>
References: <1313970987.96106.YahooMailNeo@web161008.mail.bf1.yahoo.com>
	<CAKzWOxWTGEYrAwHz20QO0JAYSk7oc9DyLOkVXiJ1=u8v4AAkpA@mail.gmail.com>
Message-ID: <1314060883.62351.YahooMailNeo@web161008.mail.bf1.yahoo.com>

OK so if I were to use the ESL lib with PHP, can it still be used with the fs_ivrd daemon? I have not seen any ESL examples that is used with the fs_ivrd deamon, so I am not exactly to use the ESL lib.



________________________________
From: Michael Collins <msc at freeswitch.org>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Monday, August 22, 2011 5:27 PM
Subject: Re: [Freeswitch-users] freeswitch php


This probably isn't the most efficient way of using the event socket. In fact, I'm not sure if this is anything more than a proof of concept. You are MUCH better off using the ESL lib with PHP and taking advantage of the abstraction you get.

-MC


On Sun, Aug 21, 2011 at 4:56 PM, Sam <lakersman2006 at yahoo.com> wrote:

Does anyone know how to retrieve channel variables (ie. uuid, etc.) using the php example that was shown in the wiki below?
>
>
>
>#!/usr/bin/php -q <?php // set a couple of things so we dont kill the system
ob_implicit_flush(true);
set_time_limit(30); // Open stdin so we can read the data in
$in = fopen("php://stdin", "r"); // Connect
echo "connect\n\n"; // Answer
echo "sendmsg\n";
echo "call-command: execute\n";
echo "execute-app-name: answer\n\n"; // Play a prompt
echo "sendmsg\n";
echo "call-command: execute\n";
echo "execute-app-name: playback\n";
echo "execute-app-arg: /usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav\n\n"; // Wait
sleep(5); // Hangup
echo "sendmsg\n";
echo "call-command: hangup\n\n"; fclose($in); ?>
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>


FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110822/d059fd55/attachment-0001.html 

From lakersman2006 at yahoo.com  Tue Aug 23 05:08:46 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Mon, 22 Aug 2011 18:08:46 -0700 (PDT)
Subject: [Freeswitch-users] Freeswitch say digits too quick
Message-ID: <1314061726.88443.YahooMailMobile@web161007.mail.bf1.yahoo.com>

If I wanted to record my own digit sound files is there a way to set the directory for the say app?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110822/b43a75a5/attachment.html 

From moises.silva at gmail.com  Tue Aug 23 06:27:33 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Mon, 22 Aug 2011 22:27:33 -0400
Subject: [Freeswitch-users] AMR transcoding
In-Reply-To: <4E525B50.6080404@gmail.com>
References: <4E521A2B.1050808@gmail.com>
	<CAA4nhyBFkosothGquPiAS5Po8OvZ9GTx5MP3FRnN999+w4TqPg@mail.gmail.com>
	<4E525B50.6080404@gmail.com>
Message-ID: <CAA4nhyA4F9+GztJHh8Y7WJOAUzN+hdX+==FxtOKGZagvF5c_1w@mail.gmail.com>

On Mon, Aug 22, 2011 at 9:36 AM, Michal Kopacki <mkopacki at gmail.com> wrote:
> ? ? Yes. I'm aware of licensing and Sangoma cards (and I belive I don't
> need license if use D500) and I'm testing this too but sadly I need also
> fully software solution to implement on some dedicated vps without
> physical access.

Sadly, for the specific case of AMR, you still need a license even
with D500. It is my understanding that the patent holders of AMR did
not allow including the licensing from the manufacturer, they rather
want to deal with the end user.

Note that this is not the case for other codecs such as G.729

Moises Silva
Senior Software Engineer, Software Development Manager
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
L3R 9R6 Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com


From mkopacki at gmail.com  Tue Aug 23 10:11:19 2011
From: mkopacki at gmail.com (Michal Kopacki)
Date: Tue, 23 Aug 2011 08:11:19 +0200
Subject: [Freeswitch-users] AMR transcoding
In-Reply-To: <CAA4nhyA4F9+GztJHh8Y7WJOAUzN+hdX+==FxtOKGZagvF5c_1w@mail.gmail.com>
References: <4E521A2B.1050808@gmail.com>	<CAA4nhyBFkosothGquPiAS5Po8OvZ9GTx5MP3FRnN999+w4TqPg@mail.gmail.com>	<4E525B50.6080404@gmail.com>
	<CAA4nhyA4F9+GztJHh8Y7WJOAUzN+hdX+==FxtOKGZagvF5c_1w@mail.gmail.com>
Message-ID: <4E534487.40805@gmail.com>

     I checked this and it seems you may be right - thank you for 
pointed me to that matter.  But still I'm not going forward with 
software solution to transcoding amr <-> 729/711 ...

-- 
Regards,
Michal

On 2011-08-23 04:27, Moises Silva wrote:
> On Mon, Aug 22, 2011 at 9:36 AM, Michal Kopacki<mkopacki at gmail.com>  wrote:
>>      Yes. I'm aware of licensing and Sangoma cards (and I belive I don't
>> need license if use D500) and I'm testing this too but sadly I need also
>> fully software solution to implement on some dedicated vps without
>> physical access.
> Sadly, for the specific case of AMR, you still need a license even
> with D500. It is my understanding that the patent holders of AMR did
> not allow including the licensing from the manufacturer, they rather
> want to deal with the end user.
>
> Note that this is not the case for other codecs such as G.729
>
> Moises Silva
> Senior Software Engineer, Software Development Manager
> Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
> L3R 9R6 Canada
> t. 1 905 474 1990 x128 | e. moy at sangoma.com
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mitja.thomas1 at ewetel.de  Tue Aug 23 12:02:27 2011
From: mitja.thomas1 at ewetel.de (Mitja Thomas)
Date: Tue, 23 Aug 2011 10:02:27 +0200
Subject: [Freeswitch-users] Ignore INFO DTMF
Message-ID: <4E535E93.8090602@ewetel.de>

Hi,

we are experiencing problems with DTMF between Snom Sip Phones and FS. 
The DMTF Type is set to SIP INFO (sip info only on Snom Phones). We 
could nail it down to a missing "telephone-event" parameter in sip sdp:
INVITE sip:***6 at spklw.x;user:phone SIP/2.0
Via: SIP/2.0/UDP ip:5060;branch=z9hG4bK-15eyqwai33r1;rport
From: "Test" <sip:8000 at spklw.x>;tag=qqnbrunjmq
To: <sip:***6 at spklw.x;user:phone>
Call-ID: 3c33e6c729b3-t9yol2613q6v
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:8000 at ip:5060;line=9lwqdnrx>;reg-id=1
X-Serialnumber: 000413262C33
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom370/8.4.32
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, 
PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 244

v=0
o=root 334284706 334284706 IN IP4 phoneip
s=call
c=IN IP4 phoneip
t=0 0
m=audio 12610 RTP/AVP 8 9
a=direction:both
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a:ptime:20
a=rtcp-xr:stat-summary=loss,dup,jitt
a=sendrecv

That resulted in sofia_glue.c disabling DTMF. A slight change in 
sofia_glue.c (Line 4839 in commit 
71964f61ac4817ecfc516edaf4b643f41236868d) did the trick for us:
if (tech_pvt->dtmf_type == DTMF_INFO) { //EWETEL
} else {                                //EWETEL
                 switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), 
SWITCH_LOG_DEBUG, "Disable 2833 dtmf\n");
                 switch_channel_set_variable(tech_pvt->channel, 
"dtmf_type", "none");
                 tech_pvt->dtmf_type = DTMF_NONE;
                 te = tech_pvt->recv_te = 0;
} //EWETEL

Now the question is:
Is this a too restrictive behaviour in FS or a shortcoming from the Snom 
phones (which would mean we have to submit a bug, wait, wait, wait and 
then hopefully someday it gets fixed :) )?

Regards,
Mitja

-- 
Mitja Thomas
Vertrieb Gesch?ftskunden

Branchenl?sungen / Service Entwicklung

Telefon: 	+49 (0) 441 -  8000-4916
E-Mail: 	mitja.thomas at ewe.de


EWE TEL GmbH
Cloppenburger Stra?e 310
26133 Oldenburg
E-Mail: info at ewe.de
Internet: www.ewe.de

Handelsregister Amtsgericht Oldenburg HRB 3723
Vorsitzender des Aufsichtsrates: Dr. Werner Brinker
Gesch?ftsf?hrung: Konrad Meier (Vorsitzender), Dirk Brameier, Ulf Heggenberger, Norbert Westfal

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110823/e0a73b68/attachment.html 

From asilva at wirelessmundi.com  Tue Aug 23 12:10:13 2011
From: asilva at wirelessmundi.com (Antonio)
Date: Tue, 23 Aug 2011 10:10:13 +0200
Subject: [Freeswitch-users] Problem receiving fax
In-Reply-To: <1313496873.30552.82.camel@marces.madrid.commsmundi.com>
References: <1313493347.30552.80.camel@marces.madrid.commsmundi.com>
	<CAAMQ8bROOOci2ocWZkcn_qxh-vc8kY1Tg+3deUJMs3Lvf0PMpw@mail.gmail.com>
	<1313496873.30552.82.camel@marces.madrid.commsmundi.com>
Message-ID: <1314087013.29574.59.camel@marces.madrid.commsmundi.com>

I tried with the last version, and the same occurred.

After restarting the server i can receive faxes and them it stops
receiving it. 

I can't find out what is the cause... Can anyone help me how to find out
where could be the problem.


I'm think replacing the hardware, just to be sure that is not an
hardware problem.



Thanks,

Ant?nio






On Tue, 2011-08-16 at 14:14 +0200, Antonio wrote: 
> I'm using libpri-1.4.11 and freeswitch head. I'm going to try with the
> latest libpri-1.4.12.
> 
> And post the results.
> 
> Thanks,
> Ant?nio
> 
> 
> On Tue, 2011-08-16 at 13:46 +0200, Christian Benke wrote: 
> > On 16 August 2011 13:15, Antonio <asilva at wirelessmundi.com> wrote:
> > > I'm having problems receiving fax in a pri E1 line.
> > > The log can be found at http://pastebin.freeswitch.org/17047
> > 
> > Hi!
> > 
> > I had the same issue a few days ago("FLOW T.30 Bad HDLC CRC received").
> > Recompiling&Reinstalling libpri&FreeSWITCH helped.
> > 
> > hthu2
> > Christian
> > 
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 

-- 

Un cordial saludo / Best regards, 

 _________________________

Ant?nio Silva

E-mail:asilva at wirelessmundi.com



From michal.bielicki at seventhsignal.de  Tue Aug 23 13:09:07 2011
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Tue, 23 Aug 2011 11:09:07 +0200
Subject: [Freeswitch-users] strdup error on solaris
In-Reply-To: <DA68975C70B74A1399C5B8BBE649A6F3@gmail.com>
References: <7E3EF9F827D54F628C130D0AC650BAD0@gmail.com>
	<CAKbxfG8BUmhTfr0GEoRrb=g1LcaeRLasmf+Mxt24xgPaG6Eiag@mail.gmail.com>
	<DA68975C70B74A1399C5B8BBE649A6F3@gmail.com>
Message-ID: <E4A3D22A-447F-4BBB-8317-FE1A2631FDAD@seventhsignal.de>

we define __EXTENSIONS__ if you use suncc. gcc was never tested on solaris.

Am 18.08.2011 um 23:41 schrieb Seven Du:

> On Thursday, August 18, 2011 at 10:56 PM, Anthony Minessale wrote:
>> 
>> hmm, did solaris stop putting strdup in string.h ?
>> 
>> 
> looks it's still there. Do you think I should report a jira?
> 
> 
> 
> #if defined(__EXTENSIONS__) || \
>         (!defined(_STRICT_STDC) && !defined(__XOPEN_OR_POSIX)) || \
>         defined(_XPG4_2)
> extern char *strdup(const char *);
> #endif
> 
> 
> #if defined(__EXTENSIONS__) || !defined(__XOPEN_OR_POSIX) || defined(_XPG4_2)
> extern char *strdup();
> #endif
> 
> 
> http://pastebin.freeswitch.org/17088
>  
>> On Thu, Aug 18, 2011 at 9:52 AM, Seven Du <dujinfang at gmail.com> wrote:
>>> Hi,
>>> I'm new to solaris
>>> SunOS solaris 5.11 snv_134 i86pc i386 i86pc
>>> I didn't follow the wiki about installing on solaris but installed building
>>> tool chain with pkg install SUNWgcc etc.
>>> bootstrap and configure was ok, however, I got error on gmake. Any highlight
>>> on this?  Thanks.
>>> libs/stfu/stfu.c: In function `stfu_n_debug':
>>> libs/stfu/stfu.c:224: warning: implicit declaration of function `strdup'
>>> libs/stfu/stfu.c:224: warning: assignment makes pointer from integer without
>>> a cast
>>> libs/stfu/stfu.c:227: warning: assignment makes pointer from integer without
>>> a cast
>>> libs/stfu/stfu.c: In function `stfu_n_init':
>>> libs/stfu/stfu.c:302: warning: assignment makes pointer from integer without
>>> a cast
>>> gmake[1]: *** [libfreeswitch_la-stfu.lo] #### 1
>>> gmake: *** [all] #### 2
>>> link at solaris:~/seven/freeswitch# vi libs/stfu/stfu.c:224
>>> link at solaris:~/seven/freeswitch# vi libs/stfu/stfu.c
>>> 
>>> link at solaris:~/seven/freeswitch# gcc -v
>>> Reading specs from /usr/sfw/lib/gcc/i386-pc-solaris2.11/3.4.3/specs
>>> Configured with: /builds2/sfwnv-gate/usr/src/cmd/gcc/gcc-3.4.3/configure
>>> --prefix=/usr/sfw --with-as=/usr/sfw/bin/gas --with-gnu-as
>>> --with-ld=/usr/ccs/bin/ld --without-gnu-ld --enable-languages=c,c++,f77,objc
>>> --enable-shared
>>> Thread model: posix
>>> gcc version 3.4.3 (csl-sol210-3_4-20050802)
>>> --
>>> Seven Du
>>> About: http://about.me/dujinfang
>>> Blog: http://www.dujinfang.com
>>> Proj:  http://www.freeswitch.org.cn
>>> Sent with Sparrow
>>> 
>>> 
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> 
>> -- 
>> Anthony Minessale II
>> 
>> 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
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>> 
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Michal Bielicki
Gesch?ftsf?hrer / CEO

Seventh Signal Ltd. & Co. KG
Weigandufer 45, B?ro 115, D-12059 Berlin
Voice: +49 30 60988730

Amtsgericht Charlottenburg HRA 44413 B
Ust.-ID: DE266981999
Gesch?ftsf?hrer: Michal Bielicki
Pers?nlich Haftende Gesellschafterin:
Seventh Signal Ltd, 69 Great Hampton St. Birmingham, 
B18 6EW, GB, Company Nr.: 06889439
WWW.: http://www.seventhsignal.de



----

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110823/208a4522/attachment-0001.html 

From mrene_lists at avgs.ca  Tue Aug 23 13:30:48 2011
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Tue, 23 Aug 2011 11:30:48 +0200
Subject: [Freeswitch-users] How to authencticate with mod_rtmp?
In-Reply-To: <4E52B7B0.4020204@livecall.com>
References: <CADDV4gzH8gW1cP_K3vMgOtouACRs4PNQSRKC9aVd-m0uwxTjQw@mail.gmail.com>
	<4E52B7B0.4020204@livecall.com>
Message-ID: <BCE15AF6-4991-4D07-983B-6687C5D93CCD@avgs.ca>

Hi,

Accounts aren't tied to profiled in any ways. Did you try usIng the full user at domain for the username? Like 1000 at 192.168.10.196 ? 

Let me know if that works.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 2011-08-22, at 10:10 PM, Jack wrote:

> Hi Dmitry,
>  Did you ever figure the authentication out?   I am experiencing the same thing now with the latest git. 
> Thanks,
>  Jack
> 
> On 7/19/2011 4:53 AM, Dmitry Kravchenko wrote:
>> 
>> Hi!
>> 
>> It is said in book that user registry is single and centralized for all components (p. 64).
>> 
>> But while trying to login via FLEX telephone to default number of 1000 with password of 1234, I get "Authentication failed" message.
>> 
>> In freeswitch CLI I see message 
>> 
>> 2011-07-19 15:46:51.473828 [WARNING] rtmp.c:227 Authentication failed for 1000 at 192.168.10.196
>> 
>> where 192.168.10.196 is true IP of my freeswitch machine.
>> 
>> Connection with X-Lite goes OK with message
>> 
>> 2011-07-19 15:50:19.473829 [WARNING] sofia_reg.c:1337 SIP auth challenge (REGISTER) on sofia profile 'internal' for [1000 at 192.168.10.196] from ip 192.168.10.56
>> 
>> If I put wrong password into X-Lite then I get
>> 
>> 2011-07-19 15:50:19.493774 [WARNING] sofia_reg.c:1295 SIP auth failure (REGISTER) on sofia profile 'internal' for [1000 at 192.168.10.196] from ip 192.168.10.56
>> 
>> So my conclusion is that SIP phone is treates as internal profile by default. Is it possible to set RTMP profiles as internal too? 
>> 
>> Or may be I should better create exteral accounts to authenticate with RTMP? 
>> 
>> Thanks.
>> 
>> 
>> 
>> 
>> 
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110823/7ef72459/attachment.html 

From avi at avimarcus.net  Tue Aug 23 13:41:09 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 23 Aug 2011 12:41:09 +0300
Subject: [Freeswitch-users] Ignore INFO DTMF
In-Reply-To: <4E535E93.8090602@ewetel.de>
References: <4E535E93.8090602@ewetel.de>
Message-ID: <CALNZuw=mmPiqmFme0PG3HXkbuQ0kmA6ceiMXWXpdwN4LhLN9+A@mail.gmail.com>

It's negotiates INFO but sends rfc2833?
The liberal-dtmf option will accept rfc2833 always. See here:
http://wiki.freeswitch.org/wiki/DTMF#DTMF_Options


-Avi



On Tue, Aug 23, 2011 at 11:02 AM, Mitja Thomas <mitja.thomas1 at ewetel.de>wrote:

>  Hi,
>
> we are experiencing problems with DTMF between Snom Sip Phones and FS. The
> DMTF Type is set to SIP INFO (sip info only on Snom Phones). We could nail
> it down to a missing "telephone-event" parameter in sip sdp:
> INVITE sip:***6 at spklw.x;user:phone SIP/2.0
> Via: SIP/2.0/UDP ip:5060;branch=z9hG4bK-15eyqwai33r1;rport
> From: "Test" <sip:8000 at spklw.x> <sip:8000 at spklw.x>;tag=qqnbrunjmq
> To: <sip:***6 at spklw.x;user:phone>
> Call-ID: 3c33e6c729b3-t9yol2613q6v
> CSeq: 1 INVITE
> Max-Forwards: 70
> Contact: <sip:8000 at ip:5060;line=9lwqdnrx><sip:8000 at 85.16.245.234:5060;line=9lwqdnrx>
> ;reg-id=1
> X-Serialnumber: 000413262C33
> P-Key-Flags: resolution="31x13", keys="4"
> User-Agent: snom370/8.4.32
> Accept: application/sdp
> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
> MESSAGE, INFO, UPDATE
> Allow-Events: talk, hold, refer, call-info
> Supported: timer, 100rel, replaces, from-change
> Session-Expires: 3600;refresher=uas
> Min-SE: 90
> Content-Type: application/sdp
> Content-Length: 244
>
> v=0
> o=root 334284706 334284706 IN IP4 phoneip
> s=call
> c=IN IP4 phoneip
> t=0 0
> m=audio 12610 RTP/AVP 8 9
> a=direction:both
> a=rtpmap:8 PCMA/8000
> a=rtpmap:9 G722/8000
> a:ptime:20
> a=rtcp-xr:stat-summary=loss,dup,jitt
> a=sendrecv
>
> That resulted in sofia_glue.c disabling DTMF. A slight change in
> sofia_glue.c (Line 4839 in commit 71964f61ac4817ecfc516edaf4b643f41236868d)
> did the trick for us:
> if (tech_pvt->dtmf_type == DTMF_INFO) { //EWETEL
> } else {                                //EWETEL
>                 switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session),
> SWITCH_LOG_DEBUG, "Disable 2833 dtmf\n");
>                 switch_channel_set_variable(tech_pvt->channel, "dtmf_type",
> "none");
>                 tech_pvt->dtmf_type = DTMF_NONE;
>                 te = tech_pvt->recv_te = 0;
> } //EWETEL
>
> Now the question is:
> Is this a too restrictive behaviour in FS or a shortcoming from the Snom
> phones (which would mean we have to submit a bug, wait, wait, wait and then
> hopefully someday it gets fixed :) )?
>
> Regards,
> Mitja
>
> --
> Mitja Thomas
> Vertrieb Gesch?ftskunden
>
> Branchenl?sungen / Service Entwicklung
>
> Telefon: 	+49 (0) 441 -  8000-4916
> E-Mail: 	mitja.thomas at ewe.de
>
>
> EWE TEL GmbH
> Cloppenburger Stra?e 310
> 26133 Oldenburg
> E-Mail: info at ewe.de
> Internet: www.ewe.de
>
> Handelsregister Amtsgericht Oldenburg HRB 3723
> Vorsitzender des Aufsichtsrates: Dr. Werner Brinker
> Gesch?ftsf?hrung: Konrad Meier (Vorsitzender), Dirk Brameier, Ulf Heggenberger, Norbert Westfal
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110823/d78e944d/attachment.html 

From avi at avimarcus.net  Tue Aug 23 16:36:46 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 23 Aug 2011 15:36:46 +0300
Subject: [Freeswitch-users] nf_ct_sip dropped packets?
Message-ID: <CALNZuw=RNqTePsEQwVo-OZaVHXBqsfG=noZ_vsikHsj03-W_+g@mail.gmail.com>

Hi, I keep seeing that packets are being dropped every few days from a
yealink phone. The phone has made calls, and I don't see dropped packets in
my logs during most of the successful call. FS is on a public IP, the phone
is behind nat.

I don't know if this is actually a problem, but I'm concerned about this
error message because I don't know what it is. Is iptables rate-limiting
something..?

Situataion, kern.log error, iptables/ufw list:
http://pastebin.freeswitch.org/17156

Thanks!
-Avi
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110823/fa0c373b/attachment.html 

From steveu at coppice.org  Tue Aug 23 17:20:47 2011
From: steveu at coppice.org (Steve Underwood)
Date: Tue, 23 Aug 2011 21:20:47 +0800
Subject: [Freeswitch-users] AMR transcoding
In-Reply-To: <CAA4nhyA4F9+GztJHh8Y7WJOAUzN+hdX+==FxtOKGZagvF5c_1w@mail.gmail.com>
References: <4E521A2B.1050808@gmail.com>	<CAA4nhyBFkosothGquPiAS5Po8OvZ9GTx5MP3FRnN999+w4TqPg@mail.gmail.com>	<4E525B50.6080404@gmail.com>
	<CAA4nhyA4F9+GztJHh8Y7WJOAUzN+hdX+==FxtOKGZagvF5c_1w@mail.gmail.com>
Message-ID: <4E53A92F.2080504@coppice.org>

Hi Moises,

On 08/23/2011 10:27 AM, Moises Silva wrote:
> On Mon, Aug 22, 2011 at 9:36 AM, Michal Kopacki<mkopacki at gmail.com>  wrote:
>>      Yes. I'm aware of licensing and Sangoma cards (and I belive I don't
>> need license if use D500) and I'm testing this too but sadly I need also
>> fully software solution to implement on some dedicated vps without
>> physical access.
> Sadly, for the specific case of AMR, you still need a license even
> with D500. It is my understanding that the patent holders of AMR did
> not allow including the licensing from the manufacturer, they rather
> want to deal with the end user.
Can you explain how that works? There appears to be no way for an end 
user to licence AMR.

Steve



From manjiri05_deshpande at yahoo.co.in  Tue Aug 23 17:24:49 2011
From: manjiri05_deshpande at yahoo.co.in (Manjiri Deshpande)
Date: Tue, 23 Aug 2011 18:54:49 +0530 (IST)
Subject: [Freeswitch-users] How to get devices registered with skinny
	protocol
Message-ID: <1314105889.93376.YahooMailNeo@web95910.mail.in.yahoo.com>

Hi,
?
In sofia we have following command to get users currently registered with a profile.
?
?sofia xmlstatus/status profile <profilename> 
?
In Skinny there is no way to find which devices are registered with skinny protocol.I tried following command
?
skinny status profile <profilename>
?
But above command output does not show devices registered.
I have a requirement to get currently registred devices in xml format(like sofia xmlstatus command).
?
?
Thanks,
Manjiri
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110823/a62735ac/attachment-0001.html 

From cmrienzo at gmail.com  Tue Aug 23 18:09:59 2011
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Tue, 23 Aug 2011 10:09:59 -0400
Subject: [Freeswitch-users] mod_spidermonkey loading error
In-Reply-To: <CAKzWOxXmBcCLQXnjAEXmZYk+TVps5VSMOFsmbCtr2doO7v_fVA@mail.gmail.com>
References: <20110817091244.5582.qmail@community37.interfree.it>
	<4E4B9352.4010006@hw.ac.uk>
	<CAKzWOxVyTnNMeHvxAWviHNM5MfPFW8XQoF_M0X=0ghzLv3Zhpw@mail.gmail.com>
	<4E4D34D4.8000401@hw.ac.uk>
	<CAKzWOxV6nM0dnar6kZThiakpv3gCpjzMAQXr6+P2WBuYncC8-Q@mail.gmail.com>
	<4E4E80F3.9060605@hw.ac.uk>
	<CAKzWOxXwBjpz1oToGMS6F8OniSNMciOx3XQZGTzKNHgN1hzTCg@mail.gmail.com>
	<4E4EA287.6020701@hw.ac.uk>
	<CAKzWOxXmBcCLQXnjAEXmZYk+TVps5VSMOFsmbCtr2doO7v_fVA@mail.gmail.com>
Message-ID: <CAD-qRkhkynR92yWaMUXo=egwCUN1wgS46ei7ZzwbkDypEf1nFQ@mail.gmail.com>

Changing the MRCP params rtp-ip and rtp-ext-ip from "auto" to the IP address
resolved the issue.


On Fri, Aug 19, 2011 at 2:27 PM, Michael Collins <msc at freeswitch.org> wrote:

> Open a jira if you haven't already done so and hopefully CRienzo can have a
> look.
>
> -MC
>
>
> On Fri, Aug 19, 2011 at 10:51 AM, xl127 <x.liu at hw.ac.uk> wrote:
>
>> **
>> okay, thanks!
>>
>> I tried to git the latest version on both CentOS and Fedora. It works fine
>> on CentOS but still no audio out for mod_unimrcp on Fedora. All firewalls
>> were disabled.
>> The original pizza demo which uses streamFile from audio files works file
>> on Fedora.
>>
>> So there must be something wrong with FS-UniMRCP on Fedora 14.
>>
>> Any suggestions about how to find the causes?
>>
>> Thanks!
>>
>> Xing
>>
>>
>>
>> On 19/08/11 16:38, Michael Collins wrote:
>>
>> I don't know that there's anything affected. I have been using FS on a
>> CentOS 6 machine and none of the modules built w/o libcurl have had any
>> adverse affects.
>>
>>  -MC
>>
>>  On Fri, Aug 19, 2011 at 8:27 AM, xl127 <x.liu at hw.ac.uk> wrote:
>>
>>>  I am wondering which FS function will be affected by "--without-libcurl"
>>>
>>> My simple dialplan tries to "speak" some text via mod_unmrcp. Exact same
>>> setups on both CentOS 5 and Fedora 14.
>>> It works on CentOS but on Fedora I didn't hear anything, no sound
>>> outputs. The debug messages for the speechsythesizer and the mrcp_client
>>> look normal from the console.
>>>
>>> Is it because of the libcurl thing, or because of the FS version?
>>>
>>> The CentOS FS version is:
>>> FreeSWITCH Version 1.0.head (git-492bc6b 2011-07-23 12-53-04 -0400)
>>>
>>> The Fedora FS version is quite new:
>>> FreeSWITCH Version 1.0.head (git-cd31633 2011-08-17 19-34-22 -0500)
>>>
>>> Cheers,
>>> Xing
>>>
>>>
>>>
>>>
>>> On 18/08/11 17:34, Michael Collins wrote:
>>>
>>>
>>>
>>> On Thu, Aug 18, 2011 at 8:50 AM, xl127 <x.liu at hw.ac.uk> wrote:
>>>
>>>>  It works on Fedora 14 now, thanks Michael!
>>>>
>>>> I am wondering if I need to give the flag "--without-libcurl" every time
>>>> when I add a new module to FS and do "configure"?
>>>>
>>>
>>>  Only when you need to actually re-run the configure script. Most of the
>>> time when you do "git pull && make install" or "make current" you're fine.
>>>
>>>  -MC
>>>
>>>
>>>
>>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>
>>>
>>>
>>> ------------------------------
>>>  Heriot-Watt University is a Scottish charity registered under charity
>>> number SC000278.
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>>
>> ------------------------------
>>  Heriot-Watt University is a Scottish charity registered under charity
>> number SC000278.
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110823/3f506f3d/attachment.html 

From krunci88 at gmail.com  Tue Aug 23 14:55:56 2011
From: krunci88 at gmail.com (krunci)
Date: Tue, 23 Aug 2011 03:55:56 -0700 (PDT)
Subject: [Freeswitch-users] Freeswitch in server mode with OpenFire
Message-ID: <1314096956460-6715666.post@n2.nabble.com>

Hi,

I'm new here on this forum, but I have some problem with configuring
Freeswitch with OpenFire.

First of all, did anybody have succes in configuring Freeswitch with
OpenFire, and that it work properly??

Second, I have configure Freeswitch as a component, and add it in Openfire,
but when I try to establish a call I have some problems:
1. When I call from SIP to XMPP (openfire) on Sip client have ringing all
the time, but on XMPP client side I manage to answer the call, but on SIP
side doesn't change anything. (ringing all the time)
That means I coundn't establish any call. On Pidgin (XMPP side) is written
"Call in progress".  

2. In other way, when I try to call SIP client from XMPP client, on XMPP
side is trying to call SIP side, but on SIP side doesn't change anything.
Like nothing is happening on other side.


So if someone who is expert in openfire and Freeswitch can help me with that
problems?

Thanks

Regards,

Krunci

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-in-server-mode-with-OpenFire-tp6715666p6715666.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From leon at scarlet-internet.nl  Tue Aug 23 19:42:43 2011
From: leon at scarlet-internet.nl (Leon de Rooij)
Date: Tue, 23 Aug 2011 17:42:43 +0200
Subject: [Freeswitch-users] Ignore INFO DTMF
In-Reply-To: <4E535E93.8090602@ewetel.de>
References: <4E535E93.8090602@ewetel.de>
Message-ID: <EE27A5C5-0887-4589-A5FC-54C1E8883B80@scarlet-internet.nl>

Hello Mitja,

I've been looking through that bit of code as well the last few days as we're having problems with a host that negotiates no telephone-event, but wants it anyway.

But for your case, if you want to force SIP INFO, isn't it good enough to just set a channel variable dtmf_info=true on the channel towards your snom phones ?

regards,

Leon



On Aug 23, 2011, at 10:02 AM, Mitja Thomas wrote:

> Hi,
> 
> we are experiencing problems with DTMF between Snom Sip Phones and FS. The DMTF Type is set to SIP INFO (sip info only on Snom Phones). We could nail it down to a missing "telephone-event" parameter in sip sdp:
> INVITE sip:***6 at spklw.x;user:phone SIP/2.0
> Via: SIP/2.0/UDP ip:5060;branch=z9hG4bK-15eyqwai33r1;rport
> From: "Test" <sip:8000 at spklw.x>;tag=qqnbrunjmq
> To: <sip:***6 at spklw.x;user:phone>
> Call-ID: 3c33e6c729b3-t9yol2613q6v
> CSeq: 1 INVITE
> Max-Forwards: 70
> Contact: <sip:8000 at ip:5060;line=9lwqdnrx>;reg-id=1
> X-Serialnumber: 000413262C33
> P-Key-Flags: resolution="31x13", keys="4"
> User-Agent: snom370/8.4.32
> Accept: application/sdp
> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
> Allow-Events: talk, hold, refer, call-info
> Supported: timer, 100rel, replaces, from-change
> Session-Expires: 3600;refresher=uas
> Min-SE: 90
> Content-Type: application/sdp
> Content-Length: 244
> 
> v=0
> o=root 334284706 334284706 IN IP4 phoneip
> s=call
> c=IN IP4 phoneip
> t=0 0
> m=audio 12610 RTP/AVP 8 9
> a=direction:both
> a=rtpmap:8 PCMA/8000
> a=rtpmap:9 G722/8000
> a:ptime:20
> a=rtcp-xr:stat-summary=loss,dup,jitt
> a=sendrecv
> 
> That resulted in sofia_glue.c disabling DTMF. A slight change in sofia_glue.c (Line 4839 in commit 71964f61ac4817ecfc516edaf4b643f41236868d) did the trick for us:
> if (tech_pvt->dtmf_type == DTMF_INFO) { //EWETEL
> } else {                                //EWETEL
>                 switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "Disable 2833 dtmf\n");
>                 switch_channel_set_variable(tech_pvt->channel, "dtmf_type", "none");
>                 tech_pvt->dtmf_type = DTMF_NONE;
>                 te = tech_pvt->recv_te = 0;
> } //EWETEL
> 
> Now the question is: 
> Is this a too restrictive behaviour in FS or a shortcoming from the Snom phones (which would mean we have to submit a bug, wait, wait, wait and then hopefully someday it gets fixed :) )?
> 
> Regards,
> Mitja
> 
> -- 
> Mitja Thomas
> Vertrieb Gesch?ftskunden
> 
> Branchenl?sungen / Service Entwicklung
> 
> Telefon: 	+49 (0) 441 -  8000-4916
> E-Mail: 	mitja.thomas at ewe.de
> 
> 
> EWE TEL GmbH 
> Cloppenburger Stra?e 310
> 26133 Oldenburg
> E-Mail: info at ewe.de
> Internet: www.ewe.de
> 
> Handelsregister Amtsgericht Oldenburg HRB 3723
> Vorsitzender des Aufsichtsrates: Dr. Werner Brinker 
> Gesch?ftsf?hrung: Konrad Meier (Vorsitzender), Dirk Brameier, Ulf Heggenberger, Norbert Westfal
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110823/7ada40ce/attachment.html 

From jack at livecall.com  Tue Aug 23 19:48:19 2011
From: jack at livecall.com (Jack)
Date: Tue, 23 Aug 2011 08:48:19 -0700
Subject: [Freeswitch-users] How to authencticate with mod_rtmp?
In-Reply-To: <BCE15AF6-4991-4D07-983B-6687C5D93CCD@avgs.ca>
References: <CADDV4gzH8gW1cP_K3vMgOtouACRs4PNQSRKC9aVd-m0uwxTjQw@mail.gmail.com>
	<4E52B7B0.4020204@livecall.com>
	<BCE15AF6-4991-4D07-983B-6687C5D93CCD@avgs.ca>
Message-ID: <4E53CBC3.5060008@livecall.com>

Thanks Mathieu,
  That works!
Jack

On 8/23/2011 2:30 AM, Mathieu Rene wrote:
> Hi,
>
> Accounts aren't tied to profiled in any ways. Did you try usIng the 
> full user at domain for the username? Like 1000 at 192.168.10.196 ?
>
> Let me know if that works.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca <mailto:mrene at avgs.ca>
>
>
>
>
> On 2011-08-22, at 10:10 PM, Jack wrote:
>
>> Hi Dmitry,
>>  Did you ever figure the authentication out?   I am experiencing the 
>> same thing now with the latest git.
>> Thanks,
>>  Jack
>>
>> On 7/19/2011 4:53 AM, Dmitry Kravchenko wrote:
>>> Hi!
>>>
>>> It is said in book that user registry is single and centralized for 
>>> all components (p. 64).
>>>
>>> But while trying to login via FLEX telephone to default number of 
>>> 1000 with password of 1234, I get "Authentication failed" message.
>>>
>>> In freeswitch CLI I see message
>>>
>>> 2011-07-19 15:46:51.473828 [WARNING] rtmp.c:227 Authentication 
>>> failed for 1000 at 192.168.10.196 <mailto:1000 at 192.168.10.196>
>>>
>>> where 192.168.10.196 is true IP of my freeswitch machine.
>>>
>>> Connection with X-Lite goes OK with message
>>>
>>> 2011-07-19 15:50:19.473829 [WARNING] sofia_reg.c:1337 SIP auth 
>>> challenge (REGISTER) on sofia profile 'internal' for 
>>> [1000 at 192.168.10.196 <mailto:1000 at 192.168.10.196>] from ip 192.168.10.56
>>>
>>> If I put wrong password into X-Lite then I get
>>>
>>> 2011-07-19 15:50:19.493774 [WARNING] sofia_reg.c:1295 SIP auth 
>>> failure (REGISTER) on sofia profile 'internal' for 
>>> [1000 at 192.168.10.196 <mailto:1000 at 192.168.10.196>] from ip 192.168.10.56
>>>
>>> So my conclusion is that SIP phone is treates as internal profile by 
>>> default. Is it possible to set RTMP profiles as internal too?
>>>
>>> Or may be I should better create exteral accounts to authenticate 
>>> with RTMP?
>>>
>>> Thanks.
>>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com  877-7-4ACLUE
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org 
>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110823/174f36e9/attachment-0001.html 

From anthony.minessale at gmail.com  Tue Aug 23 20:31:46 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 23 Aug 2011 11:31:46 -0500
Subject: [Freeswitch-users] How to authencticate with mod_rtmp?
In-Reply-To: <4E53CBC3.5060008@livecall.com>
References: <CADDV4gzH8gW1cP_K3vMgOtouACRs4PNQSRKC9aVd-m0uwxTjQw@mail.gmail.com>
	<4E52B7B0.4020204@livecall.com>
	<BCE15AF6-4991-4D07-983B-6687C5D93CCD@avgs.ca>
	<4E53CBC3.5060008@livecall.com>
Message-ID: <CAKbxfG-OmDSXyZdL9Fc=VpoUamdgbjr-_h+=OuodHnK3aMaWQQ@mail.gmail.com>

Are all of you guys playing with mod_rtmp working together in the jira forum?

http://jira.freeswitch.org/browse/FS-3368

I want maximum collaboration on this module, it was a condition of its release.



On Tue, Aug 23, 2011 at 10:48 AM, Jack <jack at livecall.com> wrote:
> Thanks Mathieu,
> ?That works!
> Jack
>
> On 8/23/2011 2:30 AM, Mathieu Rene wrote:
>
> Hi,
> Accounts aren't tied to profiled in any ways. Did you try usIng the full
> user at domain for the username? Like 1000 at 192.168.10.196 ?
> Let me know if that works.
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
> On 2011-08-22, at 10:10 PM, Jack wrote:
>
> Hi Dmitry,
> ?Did you ever figure the authentication out??? I am experiencing the same
> thing now with the latest git.
> Thanks,
> ?Jack
>
> On 7/19/2011 4:53 AM, Dmitry Kravchenko wrote:
>
> Hi!
> It is said in book that user registry is single and centralized for all
> components (p. 64).
> But while trying to login via FLEX telephone to default number of 1000 with
> password of 1234, I get "Authentication failed" message.
> In freeswitch CLI I see message
> 2011-07-19 15:46:51.473828 [WARNING] rtmp.c:227 Authentication failed for
> 1000 at 192.168.10.196
> where 192.168.10.196 is true IP of my freeswitch machine.
> Connection with X-Lite goes OK with message
> 2011-07-19 15:50:19.473829 [WARNING] sofia_reg.c:1337 SIP auth challenge
> (REGISTER) on sofia profile 'internal' for [1000 at 192.168.10.196] from ip
> 192.168.10.56
> If I put wrong password into X-Lite then I get
> 2011-07-19 15:50:19.493774 [WARNING] sofia_reg.c:1295 SIP auth failure
> (REGISTER) on sofia profile 'internal' for [1000 at 192.168.10.196] from ip
> 192.168.10.56
> So my conclusion is that SIP phone is treates as internal profile by
> default. Is it possible to set RTMP profiles as internal too?
> Or may be I should better create exteral accounts to authenticate with
> RTMP?
> Thanks.
>
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From jmesquita at freeswitch.org  Tue Aug 23 20:38:13 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Tue, 23 Aug 2011 13:38:13 -0300
Subject: [Freeswitch-users] =?windows-1252?q?FreeSWITCH=99_in_Mexico?=
Message-ID: <CABH8QX3icB5P-OiEzkqMkwcVtBCmcBNGr=P-iLyD_w2nAOmjpg@mail.gmail.com>

Estimados,

Quisiera ante todo pedirles disculpas por enviarles este mensaje off-topic.
Yo estar? dando una charla gratuita de FreeSWITCH? en M?xico(DF) en el dia
30 de Agosto. La charla ser? como un taller en donde instalaremos placas TDM
y crearemos casos reales de instalaciones. La idea es presentar un poco las
posibilidades de FreeSWITCH? y difundir un poco m?s la idea en M?xico. El
evento es gratuito pero con cupos limitados.

El evento es auspiciado por Khomp y Toga Soluciones. Para inscribirse, basta
ingresar al link: http://togakhomp.eventbrite.com/

********************************

Guys,

First of all, sorry for the off topic. I am going to give a speech on
FreeSWITCH? in Mexico this next 28th of August and would like to invite the
mexican community to join me. It will be more like a workshop presenting
real world scenarios using TDM cards provided by Khomp. The event is free
but with limited room.

This is being sponsored by Khomp and Toga Soluciones. If you are interested,
please join us at: http://togakhomp.eventbrite.com/


Thank you all.

Regards,
Jo?o Mesquita
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110823/9e95dcc8/attachment.html 

From msc at freeswitch.org  Tue Aug 23 20:59:00 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 23 Aug 2011 09:59:00 -0700
Subject: [Freeswitch-users] Freeswitch say digits too quick
In-Reply-To: <1314061726.88443.YahooMailMobile@web161007.mail.bf1.yahoo.com>
References: <1314061726.88443.YahooMailMobile@web161007.mail.bf1.yahoo.com>
Message-ID: <CAKzWOxWg9qdmSeUd=ObOx-YyXWAREiPeYXkqvUSAJhv24MeuUg@mail.gmail.com>

If you want to use your sounds exclusively then you could just replace the
existing sound files with your custom sounds.

-MC

On Mon, Aug 22, 2011 at 6:08 PM, Sam <lakersman2006 at yahoo.com> wrote:

> If I wanted to record my own digit sound files is there a way to set the
> directory for the say app?
>
>  ------------------------------
> * From: * Anthony Minessale <anthony.minessale at gmail.com>;
> * To: * FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>;
> * Subject: * Re: [Freeswitch-users] Freeswitch say digits too quick
> * Sent: * Mon, Aug 22, 2011 7:00:27 PM
>
>   if you are on latest, all you could do is modify mod_dptools.c line
> 3628, the 250 on that line is the ms  to delay between each new file
> played by the file string .
>
>
> On Mon, Aug 22, 2011 at 1:36 PM, Sam <lakersman2006 at yahoo.com> wrote:
> > Is there a way to slow down the speed when using the say app to repeat
> > digits like 12345 (one two three four five), right now it is a bit too
> fast.
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110823/8b9d04fb/attachment.html 

From msc at freeswitch.org  Tue Aug 23 21:00:17 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 23 Aug 2011 10:00:17 -0700
Subject: [Freeswitch-users] Freeswitch say digits too quick
In-Reply-To: <1314061726.88443.YahooMailMobile@web161007.mail.bf1.yahoo.com>
References: <1314061726.88443.YahooMailMobile@web161007.mail.bf1.yahoo.com>
Message-ID: <CAKzWOxUQFY4kc6w3qHkBhhgqcsJZQhmRZKRhUaEvEciq0w17rQ@mail.gmail.com>

Oops, forgot to mention: look in conf/lang/en.xml - you can specify the
directory for mod_say_en.

-MC

On Mon, Aug 22, 2011 at 6:08 PM, Sam <lakersman2006 at yahoo.com> wrote:

> If I wanted to record my own digit sound files is there a way to set the
> directory for the say app?
>
>  ------------------------------
> * From: * Anthony Minessale <anthony.minessale at gmail.com>;
> * To: * FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>;
> * Subject: * Re: [Freeswitch-users] Freeswitch say digits too quick
> * Sent: * Mon, Aug 22, 2011 7:00:27 PM
>
>   if you are on latest, all you could do is modify mod_dptools.c line
> 3628, the 250 on that line is the ms  to delay between each new file
> played by the file string .
>
>
> On Mon, Aug 22, 2011 at 1:36 PM, Sam <lakersman2006 at yahoo.com> wrote:
> > Is there a way to slow down the speed when using the say app to repeat
> > digits like 12345 (one two three four five), right now it is a bit too
> fast.
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110823/182737bf/attachment.html 

From msc at freeswitch.org  Tue Aug 23 21:06:59 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 23 Aug 2011 10:06:59 -0700
Subject: [Freeswitch-users] freeswitch php
In-Reply-To: <1314060883.62351.YahooMailNeo@web161008.mail.bf1.yahoo.com>
References: <1313970987.96106.YahooMailNeo@web161008.mail.bf1.yahoo.com>
	<CAKzWOxWTGEYrAwHz20QO0JAYSk7oc9DyLOkVXiJ1=u8v4AAkpA@mail.gmail.com>
	<1314060883.62351.YahooMailNeo@web161008.mail.bf1.yahoo.com>
Message-ID: <CAKzWOxUNFYNRPEknF0uF5gKUZBzfTzOr4R2gCS10SdoQ1fSP8A@mail.gmail.com>

Aha! fs_ivrd. I haven't tinkered w/ fs_ivrd outside of Perl, but it should
work with anything that can read/write STDIN/STDOUT. Give me a few hours to
work on some day job items and then I'll hop in here and take a look. Maybe
you can help me wikify some of this knowledge. ;)

-MC

P.S. - If anyone else has or is using fs_ivrd and has some experience with
it please let me know. I'd like to borrow some of your mental cycles so that
we can document this better.

On Mon, Aug 22, 2011 at 5:54 PM, Sam <lakersman2006 at yahoo.com> wrote:

> OK so if I were to use the ESL lib with PHP, can it still be used with the
> fs_ivrd daemon? I have not seen any ESL examples that is used with the
> fs_ivrd deamon, so I am not exactly to use the ESL lib.
>
> ------------------------------
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Monday, August 22, 2011 5:27 PM
> *Subject:* Re: [Freeswitch-users] freeswitch php
>
> This probably isn't the most efficient way of using the event socket. In
> fact, I'm not sure if this is anything more than a proof of concept. You are
> MUCH better off using the ESL lib with PHP and taking advantage of the
> abstraction you get.
>
> -MC
>
> On Sun, Aug 21, 2011 at 4:56 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> Does anyone know how to retrieve channel variables (ie. uuid, etc.) using
> the php example that was shown in the wiki below?
>
> #!/usr/bin/php -q
>
> <?php
>
> // set a couple of things so we dont kill the system
> ob_implicit_flush(true);
> set_time_limit(30);
>
> // Open stdin so we can read the data in
> $in = fopen("php://stdin", "r");
>
> // Connect
> echo "connect\n\n";
>
> // Answer
> echo "sendmsg\n";
> echo "call-command: execute\n";
> echo "execute-app-name: answer\n\n";
>
> // Play a prompt
> echo "sendmsg\n";
> echo "call-command: execute\n";
> echo "execute-app-name: playback\n";
> echo "execute-app-arg: /usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav\n\n";
>
> // Wait
> sleep(5);
>
> // Hangup
> echo "sendmsg\n";
> echo "call-command: hangup\n\n";
>
> fclose($in);
>
> ?>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110823/9b4e5840/attachment-0001.html 

From acrow at integrafin.co.uk  Tue Aug 23 21:25:25 2011
From: acrow at integrafin.co.uk (Alex Crow)
Date: Tue, 23 Aug 2011 18:25:25 +0100
Subject: [Freeswitch-users] Hold and BLF - disable flashing or pickup for
	held call?
Message-ID: <4E53E285.1020900@integrafin.co.uk>

Hi.

I have some snom 370s, and I noticed that when a monitored extension has 
held a call then the corresponding BLF lamp flashes exactly as if the 
monitored extension is ringing, and it is then possible to steal the 
call from the "holder" by pressing the button.

Ideally I'd like neither of these things to happen. Is it possible to 
disable one or both for a held call?

I have a polycom IP 650 with productivity licence and it shows the 
status of the holding extension as ringing (with the musical notes 
animation on the status display - not great) but at least it doesn't 
steal the call.

Cheers

Alex

-- 
This message is intended only for the addressee and may contain
confidential information.  Unless you are that person, you may not
disclose its contents or use it in any way and are requested to delete
the message along with any attachments and notify us immediately.

"Transact" is operated by Integrated Financial Arrangements plc
Domain House, 5-7 Singer Street, London  EC2A 4BQ
Tel: (020) 7608 4900 Fax: (020) 7608 5300
(Registered office: as above; Registered in England and Wales under number: 3727592)
Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856)



From sg at novetys.com  Tue Aug 23 22:23:00 2011
From: sg at novetys.com (Caez)
Date: Tue, 23 Aug 2011 20:23:00 +0200
Subject: [Freeswitch-users] Problem with freeswith and a Digium card
Message-ID: <CAOu2boE+ZeWmFbJDZ=OB2e+giSeTFJhGWzJuUaDkxgMVVP+SrQ@mail.gmail.com>

Hello,

I have to problems with freeswith and my digium card whith dahdi and freetdm.
I'm a french user.

- I have no signal when the user hangs up.
I have to wait for the timeout before the line is available.
If I call a specific phone, I have to wait for the timeout to 60s
before using the line

- I don't have the phone number for the incoming and outgoing calls.
For the incoming, i have 00000000000
For the outgoing, "Anonymous"


I'm trying to change my tones.conf files, to add some options on
freetdm.conf.xml, but the problem persists

Does anyone have a solution?

Thank you in advance

Sebastien


From acrow at integrafin.co.uk  Tue Aug 23 23:33:12 2011
From: acrow at integrafin.co.uk (Alex Crow)
Date: Tue, 23 Aug 2011 20:33:12 +0100
Subject: [Freeswitch-users] Problem with freeswith and a Digium card
In-Reply-To: <CAOu2boE+ZeWmFbJDZ=OB2e+giSeTFJhGWzJuUaDkxgMVVP+SrQ@mail.gmail.com>
References: <CAOu2boE+ZeWmFbJDZ=OB2e+giSeTFJhGWzJuUaDkxgMVVP+SrQ@mail.gmail.com>
Message-ID: <4E540078.1000204@integrafin.co.uk>

On 23/08/11 19:23, Caez wrote:
> Hello,
>
> I have to problems with freeswith and my digium card whith dahdi and freetdm.
> I'm a french user.
>
> - I have no signal when the user hangs up.
> I have to wait for the timeout before the line is available.
> If I call a specific phone, I have to wait for the timeout to 60s
> before using the line
>
> - I don't have the phone number for the incoming and outgoing calls.
> For the incoming, i have 00000000000
> For the outgoing, "Anonymous"
>
>
> I'm trying to change my tones.conf files, to add some options on
> freetdm.conf.xml, but the problem persists
>
> Does anyone have a solution?
>
> Thank you in advance
>
> Sebastien
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

First, what kind of card is it?

Analog (and then how many FXO/FXS ports), BRI or PRI, manufacturer, etc?

Cheers

Alex

-- 
This message is intended only for the addressee and may contain
confidential information.  Unless you are that person, you may not
disclose its contents or use it in any way and are requested to delete
the message along with any attachments and notify us immediately.

"Transact" is operated by Integrated Financial Arrangements plc
Domain House, 5-7 Singer Street, London  EC2A 4BQ
Tel: (020) 7608 4900 Fax: (020) 7608 5300
(Registered office: as above; Registered in England and Wales under number: 3727592)
Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856)



From dgarcia at anew.com.ve  Tue Aug 23 23:37:30 2011
From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar)
Date: Tue, 23 Aug 2011 15:07:30 -0430
Subject: [Freeswitch-users] Problem with freeswith and a Digium card
In-Reply-To: <CAOu2boE+ZeWmFbJDZ=OB2e+giSeTFJhGWzJuUaDkxgMVVP+SrQ@mail.gmail.com>
References: <CAOu2boE+ZeWmFbJDZ=OB2e+giSeTFJhGWzJuUaDkxgMVVP+SrQ@mail.gmail.com>
Message-ID: <4E54017A.7070504@anew.com.ve>

You are trying to detect hangup or disconnect tone?

If you trying to disconnect tone, I was playing around with this just today,

I have a custom xml to manage my digium card. The file is located in 
/usr/local/freeswitch/conf/dialplan/public/01_freetdm_in.xml

And it is what is has:

<include>
<extension name="freetdm_in">
<condition field="chan_name" expression="FreeTDM/2:1/">
<action application="tone_detect" data="busy 425 r 0 hangup 
'normal_clearing' 3"/>
<action application="set" data="domain_name=$${domain}"/>
<action application="set" data="transfer_ringback=$${fr-ring}"/>
<action application="answer"/>
<action application="transfer" data="6442 XML default"/>
</condition>
</extension>
</include>


I think, you could play with data="busy 425 r 0 hangup 'normal_clearing' 
3" to try to get what you need. Take a look to this page: 
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect

About ANI 000000000 could be because you dont have caller id enable or 
it is not supported by your pstn.


On 8/23/2011 1:53 PM, Caez wrote:
> Hello,
>
> I have to problems with freeswith and my digium card whith dahdi and freetdm.
> I'm a french user.
>
> - I have no signal when the user hangs up.
> I have to wait for the timeout before the line is available.
> If I call a specific phone, I have to wait for the timeout to 60s
> before using the line
>
> - I don't have the phone number for the incoming and outgoing calls.
> For the incoming, i have 00000000000
> For the outgoing, "Anonymous"
>
>
> I'm trying to change my tones.conf files, to add some options on
> freetdm.conf.xml, but the problem persists
>
> Does anyone have a solution?
>
> Thank you in advance
>
> Sebastien
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> -----
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 10.0.1392 / Virus Database: 1520/3852 - Release Date: 08/23/11
>
>


-- 
Atentamente,
*Dario Garc?a*
Consultor.

CCCT, Nivel C2, Sector Yarey, Mz,
Ofc. MZ03a.
Caracas-Venezuela.
Tel?fono: +58 212 9081842
Cel: +58 412 2221515
dgarcia at anew.com.ve
http://www.anew.com.ve
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110823/f0e78f9a/attachment.html 

From petr at petris.info  Tue Aug 23 23:48:39 2011
From: petr at petris.info (Petr Nyklicek)
Date: Tue, 23 Aug 2011 21:48:39 +0200
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 62, Issue 178
In-Reply-To: <mailman.1041.1313998418.29749.freeswitch-users@lists.freeswitch.org>
References: <mailman.1041.1313998418.29749.freeswitch-users@lists.freeswitch.org>
Message-ID: <4E540417.70505@petris.info>

Hi,

thanks for your rep[y.

Ca you give me any hint how can do it ?

I need (z)rtp routing :(


 > ....... although is probably possible at least for ZRTP).
 > Moises Silva



Petr Nykl??ek



From fredyg1965 at gmail.com  Wed Aug 24 00:32:53 2011
From: fredyg1965 at gmail.com (Fredy Gonzales)
Date: Tue, 23 Aug 2011 15:32:53 -0500
Subject: [Freeswitch-users] =?windows-1252?q?FreeSWITCH=99_in_Mexico?=
References: <CABH8QX3icB5P-OiEzkqMkwcVtBCmcBNGr=P-iLyD_w2nAOmjpg@mail.gmail.com>
Message-ID: <DF1741D87A324D3BB05EF0E4B9C35C96@gcg.com.pe>

Felicito esa iniciativa y los esperamos pronto en Per?.

I commend this initiative and soon in Peru.

Saludos

Fredy Gonzales P.
  ----- Original Message ----- 
  From: Jo?o Mesquita 
  To: freeswitch-users at lists.freeswitch.org ; freeswitch-dev at lists.freeswitch.org 
  Sent: Tuesday, August 23, 2011 11:38 AM
  Subject: [Freeswitch-users] FreeSWITCH? in Mexico


  Estimados,


  Quisiera ante todo pedirles disculpas por enviarles este mensaje off-topic. Yo estar? dando una charla gratuita de FreeSWITCH? en M?xico(DF) en el dia 30 de Agosto. La charla ser? como un taller en donde instalaremos placas TDM y crearemos casos reales de instalaciones. La idea es presentar un poco las posibilidades de FreeSWITCH? y difundir un poco m?s la idea en M?xico. El evento es gratuito pero con cupos limitados.


  El evento es auspiciado por Khomp y Toga Soluciones. Para inscribirse, basta ingresar al link: http://togakhomp.eventbrite.com/


  ********************************


  Guys,


  First of all, sorry for the off topic. I am going to give a speech on FreeSWITCH? in Mexico this next 28th of August and would like to invite the mexican community to join me. It will be more like a workshop presenting real world scenarios using TDM cards provided by Khomp. The event is free but with limited room.


  This is being sponsored by Khomp and Toga Soluciones. If you are interested, please join us at: http://togakhomp.eventbrite.com/




  Thank you all.


  Regards,
  Jo?o Mesquita





------------------------------------------------------------------------------



  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110823/561744d1/attachment.html 

From moises.silva at gmail.com  Wed Aug 24 00:41:51 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Tue, 23 Aug 2011 16:41:51 -0400
Subject: [Freeswitch-users] AMR transcoding
In-Reply-To: <4E53A92F.2080504@coppice.org>
References: <4E521A2B.1050808@gmail.com>
	<CAA4nhyBFkosothGquPiAS5Po8OvZ9GTx5MP3FRnN999+w4TqPg@mail.gmail.com>
	<4E525B50.6080404@gmail.com>
	<CAA4nhyA4F9+GztJHh8Y7WJOAUzN+hdX+==FxtOKGZagvF5c_1w@mail.gmail.com>
	<4E53A92F.2080504@coppice.org>
Message-ID: <CAA4nhyCU0bdHr_RQfhuNtwkeMTR=+g82GwxUY_ouFKrfLWkXNA@mail.gmail.com>

Hi Steve, long time no see.

On Tue, Aug 23, 2011 at 9:20 AM, Steve Underwood <steveu at coppice.org> wrote:

>> Sadly, for the specific case of AMR, you still need a license even
>> with D500. It is my understanding that the patent holders of AMR did
>> not allow including the licensing from the manufacturer, they rather
>> want to deal with the end user.

> Can you explain how that works? There appears to be no way for an end
> user to licence AMR.

I believe I have misused the term "end-user". I meant, our card is not
considered an "end-user" product.

We did some research regarding this before we started selling our
transcoding card. We contacted voice age which is responsible for
licensing AMR (http://www.voiceage.com/amr_faqs.php). Because our
product is a card and according to their criteria is not and end user
product (but rather something that others use to build end-user
products), we cannot include the licensing in the card and you have to
contact them for licensing terms.

I believe that if we were shipping a box with the card included (let's
say a PBX) we most likely would be able to include the licensing for
AMR.

Note also that the license for AMR does not "unlock" any features of
the card. The card can do AMR regardless of whether you have a license
or not, but it is the same driving a car without a driver's license,
you can do it, but it is illegal.

Moises Silva
Senior Software Engineer, Software Development Manager
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
L3R 9R6 Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com


From steveu at coppice.org  Wed Aug 24 05:30:38 2011
From: steveu at coppice.org (Steve Underwood)
Date: Wed, 24 Aug 2011 09:30:38 +0800
Subject: [Freeswitch-users] AMR transcoding
In-Reply-To: <CAA4nhyCU0bdHr_RQfhuNtwkeMTR=+g82GwxUY_ouFKrfLWkXNA@mail.gmail.com>
References: <4E521A2B.1050808@gmail.com>	<CAA4nhyBFkosothGquPiAS5Po8OvZ9GTx5MP3FRnN999+w4TqPg@mail.gmail.com>	<4E525B50.6080404@gmail.com>	<CAA4nhyA4F9+GztJHh8Y7WJOAUzN+hdX+==FxtOKGZagvF5c_1w@mail.gmail.com>	<4E53A92F.2080504@coppice.org>
	<CAA4nhyCU0bdHr_RQfhuNtwkeMTR=+g82GwxUY_ouFKrfLWkXNA@mail.gmail.com>
Message-ID: <4E54543E.3050500@coppice.org>

On 08/24/2011 04:41 AM, Moises Silva wrote:
> Hi Steve, long time no see.
>
> On Tue, Aug 23, 2011 at 9:20 AM, Steve Underwood<steveu at coppice.org>  wrote:
>
>>> Sadly, for the specific case of AMR, you still need a license even
>>> with D500. It is my understanding that the patent holders of AMR did
>>> not allow including the licensing from the manufacturer, they rather
>>> want to deal with the end user.
>> Can you explain how that works? There appears to be no way for an end
>> user to licence AMR.
> I believe I have misused the term "end-user". I meant, our card is not
> considered an "end-user" product.
>
> We did some research regarding this before we started selling our
> transcoding card. We contacted voice age which is responsible for
> licensing AMR (http://www.voiceage.com/amr_faqs.php). Because our
> product is a card and according to their criteria is not and end user
> product (but rather something that others use to build end-user
> products), we cannot include the licensing in the card and you have to
> contact them for licensing terms.
>
> I believe that if we were shipping a box with the card included (let's
> say a PBX) we most likely would be able to include the licensing for
> AMR.
>
> Note also that the license for AMR does not "unlock" any features of
> the card. The card can do AMR regardless of whether you have a license
> or not, but it is the same driving a car without a driver's license,
> you can do it, but it is illegal.
As far as I know, you can't even licence AMR properly from Voice Age. I 
understand their patent pool is far from complete. Even their G.729 pool 
licencing, which is far more mature, can leave you getting unexpected 
letters from lawyers.

Steve



From gcd at i.ph  Wed Aug 24 06:36:20 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Wed, 24 Aug 2011 10:36:20 +0800
Subject: [Freeswitch-users] Question about ext-rtp-ip and ext-sip-ip
In-Reply-To: <CAEjbKNQLzTB+d57BwpdrBdsVtabtZjKsf6yvEGZUiZX3wc3DrQ@mail.gmail.com>
References: <CAEjbKNQ3tgWrVrr5uiTVKzv53avDLp3cRoavLuZsOiJfUaW_fg@mail.gmail.com>
	<65727391-DF08-4074-BB7F-BDB766DF7942@freeswitch.org>
	<CAEjbKNSGi2prQqoeLSGbKjtYbqOsD7-hggPeEUcVEWPgtCp2UQ@mail.gmail.com>
	<CAEjbKNTt8Fd=Q6TnGz407j5N4zfe4X8FSLpf2ttmpUydT9YJpA@mail.gmail.com>
	<7C7183C2-3601-47D4-B8DE-D9E292B592D3@freeswitch.org>
	<CAEjbKNT3oQJZ34FZFuq7JRxRV7zo7GaHdaybBRYYrSCKSTY_CQ@mail.gmail.com>
	<CAEjbKNTmqOUa1VyLTh74JfwVVzjxO9Tui+OLGgJPNxxo3wV1Cw@mail.gmail.com>
	<F3681824-2198-41B1-A825-5D9401EACD08@freeswitch.org>
	<CAEjbKNQLzTB+d57BwpdrBdsVtabtZjKsf6yvEGZUiZX3wc3DrQ@mail.gmail.com>
Message-ID: <CAHKqMb7xZEsq3Ws6F8DNSLTzVwUpa5UngJNo6XwpyFrtJt5zNQ@mail.gmail.com>

i'm using dd-wrt and it's working with auto-nat. did u enable UPnP on the
router?
-nandy


On Tue, Aug 23, 2011 at 1:48 AM, Bryan Lemon <bryan at bryanlemon.com> wrote:

> Just a standard dd-wrt router.
>
>
> Thank you,
> Bryan Lemon
> (302) 648-2747
>
>
>
> On Mon, Aug 22, 2011 at 13:27, Brian West <brian at freeswitch.org> wrote:
>
>> What kind of nat / firewall are you traversing?
>>
>> /b
>>
>> On Aug 22, 2011, at 10:20 AM, Bryan Lemon wrote:
>>
>> > Any other input/ideas on this one?
>> >
>> > Thank you,
>> > Bryan Lemon
>> > (302) 648-2747
>> >
>> >
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110824/706fa9ae/attachment.html 

From mkopacki at gmail.com  Wed Aug 24 10:07:46 2011
From: mkopacki at gmail.com (Michal Kopacki)
Date: Wed, 24 Aug 2011 08:07:46 +0200
Subject: [Freeswitch-users] AMR transcoding
In-Reply-To: <4E54543E.3050500@coppice.org>
References: <4E521A2B.1050808@gmail.com>	<CAA4nhyBFkosothGquPiAS5Po8OvZ9GTx5MP3FRnN999+w4TqPg@mail.gmail.com>	<4E525B50.6080404@gmail.com>	<CAA4nhyA4F9+GztJHh8Y7WJOAUzN+hdX+==FxtOKGZagvF5c_1w@mail.gmail.com>	<4E53A92F.2080504@coppice.org>	<CAA4nhyCU0bdHr_RQfhuNtwkeMTR=+g82GwxUY_ouFKrfLWkXNA@mail.gmail.com>
	<4E54543E.3050500@coppice.org>
Message-ID: <4E549532.8040006@gmail.com>

On 2011-08-24 03:30, Steve Underwood wrote:
>
> As far as I know, you can't even licence AMR properly from Voice Age. I
> understand their patent pool is far from complete. Even their G.729 pool
> licencing, which is far more mature, can leave you getting unexpected
> letters from lawyers.
>
> Steve

     In that case how one can do proper licensing ? I didn't found any 
other company selling AMR Licenses.

-- 
Regards,
Michal


From devel at thom.fr.eu.org  Wed Aug 24 10:45:01 2011
From: devel at thom.fr.eu.org (=?UTF-8?Q?Fran=C3=A7ois_Legal?=)
Date: Wed, 24 Aug 2011 08:45:01 +0200
Subject: [Freeswitch-users] Problem with freeswith and a Digium card
In-Reply-To: <CAOu2boE+ZeWmFbJDZ=OB2e+giSeTFJhGWzJuUaDkxgMVVP+SrQ@mail.gmail.com>
References: <CAOu2boE+ZeWmFbJDZ=OB2e+giSeTFJhGWzJuUaDkxgMVVP+SrQ@mail.gmail.com>
Message-ID: <6491f89817632ed80648a716ee338470@thom.fr.eu.org>

  

I suppose this problem happens on analog channels. 

For CID, this
is because the french standard for CID is not the same as the US
standard. 

I committed a patch (Openzap-100 IIRC) for this in Jira some
monthes ago, but it never went through mainstream because lack of
testing. 

For the hang up detection, normally freetdm can now detect
polarity reversal (which in France should signal hook on/off) so maybe
this is not correctly configured in freetdm.conf 

Fran?ois 

On Tue, 23
Aug 2011 20:23:00 +0200, Caez wrote: 

> Hello,
> 
> I have to problems
with freeswith and my digium card whith dahdi and freetdm.
> I'm a
french user.
> 
> - I have no signal when the user hangs up.
> I have to
wait for the timeout before the line is available.
> If I call a
specific phone, I have to wait for the timeout to 60s
> before using the
line
> 
> - I don't have the phone number for the incoming and outgoing
calls.
> For the incoming, i have 00000000000
> For the outgoing,
"Anonymous"
> 
> I'm trying to change my tones.conf files, to add some
options on
> freetdm.conf.xml, but the problem persists
> 
> Does anyone
have a solution?
> 
> Thank you in advance
> 
> Sebastien
> 
>
FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
[1]
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
[2]
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
[3]
> http://www.freeswitch.org [4]

  

Links:
------
[1]
mailto:FreeSWITCH-users at lists.freeswitch.org
[2]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
[3]
http://lists.freeswitch.org/mailman/options/freeswitch-users
[4]
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110824/104361b1/attachment.html 

From pkelly at gmail.com  Wed Aug 24 11:30:45 2011
From: pkelly at gmail.com (Pete Kelly)
Date: Wed, 24 Aug 2011 08:30:45 +0100
Subject: [Freeswitch-users] Making Freeswitch listen on 2 IP addresses
Message-ID: <CAEWH9M-yKO+7ko_wOsxHTEp-Rp7ZRr3+wuMvU=MpquuCFCQ=nA@mail.gmail.com>

Hi

Is it possible to make freeswitch listen on more than one IP for requests
within the same profile?

I would like to assign a VIP to a pair of freeswitch boxes, for the purposes
of failover if one of the boxes should fail or go down. The VIP can then be
used by web applications which have to send XMLRPC requests to freeswitch.

In addition I would like each of the boxes to listen on their "normal" eth0
IP too, so systems like opensips which support load balancing can evenly
distribute load across the two boxes.

My profile.xml file currently has this entry:

<param name="sip-ip" value="$${local_ip_v4}"/>

So I can't see how freeswitch can be told to listen on more than one IP for
a profile.

Is this possible?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110824/0f4573d8/attachment.html 

From avi at avimarcus.net  Wed Aug 24 11:45:47 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 24 Aug 2011 10:45:47 +0300
Subject: [Freeswitch-users] Making Freeswitch listen on 2 IP addresses
In-Reply-To: <CAEWH9M-yKO+7ko_wOsxHTEp-Rp7ZRr3+wuMvU=MpquuCFCQ=nA@mail.gmail.com>
References: <CAEWH9M-yKO+7ko_wOsxHTEp-Rp7ZRr3+wuMvU=MpquuCFCQ=nA@mail.gmail.com>
Message-ID: <CALNZuw=0igBwgXSj15saLstFV+91VzUXr=XiQq0MRaYkGQCPTg@mail.gmail.com>

Sorry, each profile is for exactly one IP and one port (other than a second
port for TLS). Simply copy the profile and change the profile name and IP
settings to make it listen on the second one.

-Avi


On Wed, Aug 24, 2011 at 10:30 AM, Pete Kelly <pkelly at gmail.com> wrote:

> Hi
>
> Is it possible to make freeswitch listen on more than one IP for requests
> within the same profile?
>
> I would like to assign a VIP to a pair of freeswitch boxes, for the
> purposes of failover if one of the boxes should fail or go down. The VIP can
> then be used by web applications which have to send XMLRPC requests to
> freeswitch.
>
> In addition I would like each of the boxes to listen on their "normal" eth0
> IP too, so systems like opensips which support load balancing can evenly
> distribute load across the two boxes.
>
> My profile.xml file currently has this entry:
>
> <param name="sip-ip" value="$${local_ip_v4}"/>
>
> So I can't see how freeswitch can be told to listen on more than one IP for
> a profile.
>
> Is this possible?
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110824/22d822d7/attachment.html 

From michal.bielicki at seventhsignal.de  Wed Aug 24 11:52:32 2011
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Wed, 24 Aug 2011 09:52:32 +0200
Subject: [Freeswitch-users] Making Freeswitch listen on 2 IP addresses
In-Reply-To: <CAEWH9M-yKO+7ko_wOsxHTEp-Rp7ZRr3+wuMvU=MpquuCFCQ=nA@mail.gmail.com>
References: <CAEWH9M-yKO+7ko_wOsxHTEp-Rp7ZRr3+wuMvU=MpquuCFCQ=nA@mail.gmail.com>
Message-ID: <2C2837D4-14B7-4702-AFD4-767CDDA8CAC3@seventhsignal.de>

Why not use 2 profiles ?
Am 24.08.2011 um 09:30 schrieb Pete Kelly:

> Hi
> 
> Is it possible to make freeswitch listen on more than one IP for requests within the same profile?
> 
> I would like to assign a VIP to a pair of freeswitch boxes, for the purposes of failover if one of the boxes should fail or go down. The VIP can then be used by web applications which have to send XMLRPC requests to freeswitch.
> 
> In addition I would like each of the boxes to listen on their "normal" eth0 IP too, so systems like opensips which support load balancing can evenly distribute load across the two boxes.
> 
> My profile.xml file currently has this entry:
> 
> <param name="sip-ip" value="$${local_ip_v4}"/>
> 
> So I can't see how freeswitch can be told to listen on more than one IP for a profile.
> 
> Is this possible?
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Michal Bielicki
Gesch?ftsf?hrer / CEO

Seventh Signal Ltd. & Co. KG
Weigandufer 45, B?ro 115, D-12059 Berlin
Voice: +49 30 60988730

Amtsgericht Charlottenburg HRA 44413 B
Ust.-ID: DE266981999
Gesch?ftsf?hrer: Michal Bielicki
Pers?nlich Haftende Gesellschafterin:
Seventh Signal Ltd, 69 Great Hampton St. Birmingham, 
B18 6EW, GB, Company Nr.: 06889439
WWW.: http://www.seventhsignal.de



----

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110824/0b6d03b7/attachment-0001.html 

From michal.bielicki at seventhsignal.de  Wed Aug 24 11:53:40 2011
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Wed, 24 Aug 2011 09:53:40 +0200
Subject: [Freeswitch-users] AMR transcoding
In-Reply-To: <4E549532.8040006@gmail.com>
References: <4E521A2B.1050808@gmail.com>	<CAA4nhyBFkosothGquPiAS5Po8OvZ9GTx5MP3FRnN999+w4TqPg@mail.gmail.com>	<4E525B50.6080404@gmail.com>	<CAA4nhyA4F9+GztJHh8Y7WJOAUzN+hdX+==FxtOKGZagvF5c_1w@mail.gmail.com>	<4E53A92F.2080504@coppice.org>	<CAA4nhyCU0bdHr_RQfhuNtwkeMTR=+g82GwxUY_ouFKrfLWkXNA@mail.gmail.com>
	<4E54543E.3050500@coppice.org> <4E549532.8040006@gmail.com>
Message-ID: <31C2CD3A-267F-412F-9CDA-3C0179E9865E@seventhsignal.de>


Am 24.08.2011 um 08:07 schrieb Michal Kopacki:

> On 2011-08-24 03:30, Steve Underwood wrote:
>> 
>> As far as I know, you can't even licence AMR properly from Voice Age. I
>> understand their patent pool is far from complete. Even their G.729 pool
>> licencing, which is far more mature, can leave you getting unexpected
>> letters from lawyers.
>> 
>> Steve
> 
>     In that case how one can do proper licensing ? I didn't found any 
> other company selling AMR Licenses.
> 
> -- 
> Regards,
> Michal

Simple answer: one can't.
Complex Answer: one has to find all holders of respective rights and make a contract with each group or individual rights holder.

Sounds like fun, doesn't it ?

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

Michal Bielicki
Gesch?ftsf?hrer / CEO

Seventh Signal Ltd. & Co. KG
Weigandufer 45, B?ro 115, D-12059 Berlin
Voice: +49 30 60988730

Amtsgericht Charlottenburg HRA 44413 B
Ust.-ID: DE266981999
Gesch?ftsf?hrer: Michal Bielicki
Pers?nlich Haftende Gesellschafterin:
Seventh Signal Ltd, 69 Great Hampton St. Birmingham, 
B18 6EW, GB, Company Nr.: 06889439
WWW.: http://www.seventhsignal.de


----


From mitja.thomas1 at ewetel.de  Wed Aug 24 11:56:33 2011
From: mitja.thomas1 at ewetel.de (Mitja Thomas)
Date: Wed, 24 Aug 2011 09:56:33 +0200
Subject: [Freeswitch-users] [solved] Ignore INFO DTMF
In-Reply-To: <EE27A5C5-0887-4589-A5FC-54C1E8883B80@scarlet-internet.nl>
References: <4E535E93.8090602@ewetel.de>
	<EE27A5C5-0887-4589-A5FC-54C1E8883B80@scarlet-internet.nl>
Message-ID: <4E54AEB1.5000005@ewetel.de>

Hello Leon & Avi,

@Leon: sadly that didnt do the trick. dtmf_info=true was already set in 
sip_profile, so setting it again as a channel variable didnt help. As 
far as I understand it (which is admittedly not very well): Although the 
FreeSWITCH is set to accept DTMF_INFO dtmf tones, dtmf gets disabled 
because the Snom phone does not send the right rtp media attribute 
(a=rtpmap 101 telephone-event/8000) in SIP Invite.

Anyhow the liberal-dtmf option (suggested by Avi) solved this problem. 
FS did behave well... liberal and thats exactly what we needed.
Thanks a lot to both of you!

Regards
Mitja

> Hello Mitja,
>
> I've been looking through that bit of code as well the last few days 
> as we're having problems with a host that negotiates no 
> telephone-event, but wants it anyway.
>
> But for your case, if you want to force SIP INFO, isn't it good enough 
> to just set a channel variable dtmf_info=true on the channel towards 
> your snom phones ?
>
> regards,
>
> Leon
>
>
>
> On Aug 23, 2011, at 10:02 AM, Mitja Thomas wrote:
>
>> Hi,
>>
>> we are experiencing problems with DTMF between Snom Sip Phones and 
>> FS. The DMTF Type is set to SIP INFO (sip info only on Snom Phones). 
>> We could nail it down to a missing "telephone-event" parameter in sip 
>> sdp:
>> INVITE sip:***6 at spklw.x;user:phone SIP/2.0
>> Via: SIP/2.0/UDP ip:5060;branch=z9hG4bK-15eyqwai33r1;rport
>> From: "Test" <sip:8000 at spklw.x>;tag=qqnbrunjmq
>> To: <sip:***6 at spklw.x;user:phone>
>> Call-ID: 3c33e6c729b3-t9yol2613q6v
>> CSeq: 1 INVITE
>> Max-Forwards: 70
>> Contact: <sip:8000 at ip:5060;line=9lwqdnrx>;reg-id=1
>> X-Serialnumber: 000413262C33
>> P-Key-Flags: resolution="31x13", keys="4"
>> User-Agent: snom370/8.4.32
>> Accept: application/sdp
>> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, 
>> PRACK, MESSAGE, INFO, UPDATE
>> Allow-Events: talk, hold, refer, call-info
>> Supported: timer, 100rel, replaces, from-change
>> Session-Expires: 3600;refresher=uas
>> Min-SE: 90
>> Content-Type: application/sdp
>> Content-Length: 244
>>
>> v=0
>> o=root 334284706 334284706 IN IP4 phoneip
>> s=call
>> c=IN IP4 phoneip
>> t=0 0
>> m=audio 12610 RTP/AVP 8 9
>> a=direction:both
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:9 G722/8000
>> a:ptime:20
>> a=rtcp-xr:stat-summary=loss,dup,jitt
>> a=sendrecv
>>
>> That resulted in sofia_glue.c disabling DTMF. A slight change in 
>> sofia_glue.c (Line 4839 in commit 
>> 71964f61ac4817ecfc516edaf4b643f41236868d) did the trick for us:
>> if (tech_pvt->dtmf_type == DTMF_INFO) { //EWETEL
>> } else {                                //EWETEL
>>                 
>> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), 
>> SWITCH_LOG_DEBUG, "Disable 2833 dtmf\n");
>>                 switch_channel_set_variable(tech_pvt->channel, 
>> "dtmf_type", "none");
>>                 tech_pvt->dtmf_type = DTMF_NONE;
>>                 te = tech_pvt->recv_te = 0;
>> } //EWETEL
>>
>> Now the question is:
>> Is this a too restrictive behaviour in FS or a shortcoming from the 
>> Snom phones (which would mean we have to submit a bug, wait, wait, 
>> wait and then hopefully someday it gets fixed :) )?
>>
>> Regards,
>> Mitja
>>
>> -- 
>> Mitja Thomas
>> Vertrieb Gesch?ftskunden
>>
>> Branchenl?sungen / Service Entwicklung
>>
>> Telefon: 	+49 (0) 441 -  8000-4916
>> E-Mail: 	mitja.thomas at ewe.de
>>
>>
>> EWE TEL GmbH
>> Cloppenburger Stra?e 310
>> 26133 Oldenburg
>> E-Mail:info at ewe.de
>> Internet:www.ewe.de
>>
>> Handelsregister Amtsgericht Oldenburg HRB 3723
>> Vorsitzender des Aufsichtsrates: Dr. Werner Brinker
>> Gesch?ftsf?hrung: Konrad Meier (Vorsitzender), Dirk Brameier, Ulf Heggenberger, Norbert Westfal
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org 
>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>


-- 
Mitja Thomas
Vertrieb Gesch?ftskunden

Branchenl?sungen / Service Entwicklung

Telefon: 	+49 (0) 441 -  8000-4916
E-Mail: 	mitja.thomas at ewe.de


EWE TEL GmbH
Cloppenburger Stra?e 310
26133 Oldenburg
E-Mail: info at ewe.de
Internet: www.ewe.de

Handelsregister Amtsgericht Oldenburg HRB 3723
Vorsitzender des Aufsichtsrates: Dr. Werner Brinker
Gesch?ftsf?hrung: Konrad Meier (Vorsitzender), Dirk Brameier, Ulf Heggenberger, Norbert Westfal


-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110824/9bf6bf3c/attachment.html 

From sg at novetys.com  Wed Aug 24 12:01:01 2011
From: sg at novetys.com (=?iso-8859-1?Q?S=E9bastien_Gay?=)
Date: Wed, 24 Aug 2011 10:01:01 +0200
Subject: [Freeswitch-users] Problem with freeswith and a Digium card
In-Reply-To: <6491f89817632ed80648a716ee338470@thom.fr.eu.org>
References: <CAOu2boE+ZeWmFbJDZ=OB2e+giSeTFJhGWzJuUaDkxgMVVP+SrQ@mail.gmail.com>
	<6491f89817632ed80648a716ee338470@thom.fr.eu.org>
Message-ID: <07684A03-CF0F-471D-A0FF-7A3195BE2480@novetys.com>

Hi,

Thank you for your answers.

@Alex
The card is a Digium  TDM400P with 1 FXS and 3 FXO :

active=yes
alarms=OK
description=Wildcard TDM400P REV I Board 5
name=WCTDM/4
manufacturer=Digium
devicetype=Wildcard TDM400P REV I
location=PCI Bus 00 Slot 07
basechan=1
totchans=4
irq=17
type=analog
port=1,FXS
port=2,FXO
port=3,FXO
port=4,FXO

@ Fran?ois
I have tried some options in freetdm:
<param name="answer-polarity-reverse" value="true" />
<param name="hangup-polarity-reverse" value="true" />

But the problem persists


@Dario Garcia

I did not know the option tone_detect, I'll do some tests.


Thank you again for your help.

> 
> I suppose this problem happens on analog channels.
> 
> For CID, this is because the french standard for CID is not the same as the US standard.
> 
> I committed a patch (Openzap-100 IIRC) for this in Jira some monthes ago, but it never went through mainstream because lack of testing.
> 
> For the hang up detection, normally freetdm can now detect polarity reversal (which in France should signal hook on/off) so maybe this is not correctly configured in freetdm.conf
> 
>  
> Fran?ois
> 
>  
> On Tue, 23 Aug 2011 20:23:00 +0200, Caez wrote:
> 
>> Hello,
>> 
>> I have to problems with freeswith and my digium card whith dahdi and freetdm.
>> I'm a french user.
>> 
>> - I have no signal when the user hangs up.
>> I have to wait for the timeout before the line is available.
>> If I call a specific phone, I have to wait for the timeout to 60s
>> before using the line
>> 
>> - I don't have the phone number for the incoming and outgoing calls.
>> For the incoming, i have 00000000000
>> For the outgoing, "Anonymous"
>> 
>> 
>> I'm trying to change my tones.conf files, to add some options on
>> freetdm.conf.xml, but the problem persists
>> 
>> Does anyone have a solution?
>> 
>> Thank you in advance
>> 
>> Sebastien
>> 
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>  
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110824/0c16dc60/attachment.html 

From alec.taylor6 at gmail.com  Wed Aug 24 12:13:52 2011
From: alec.taylor6 at gmail.com (Alec Taylor)
Date: Wed, 24 Aug 2011 18:13:52 +1000
Subject: [Freeswitch-users] Making Freeswitch listen on 2 IP addresses
In-Reply-To: <2C2837D4-14B7-4702-AFD4-767CDDA8CAC3@seventhsignal.de>
References: <CAEWH9M-yKO+7ko_wOsxHTEp-Rp7ZRr3+wuMvU=MpquuCFCQ=nA@mail.gmail.com>
	<2C2837D4-14B7-4702-AFD4-767CDDA8CAC3@seventhsignal.de>
Message-ID: <CAO+9iGcghRhxkdMAukA44bH+54Q4Y+2SvRPxG2zUxZ9QG_x4FA@mail.gmail.com>

Why not setup DNS?

On Wed, Aug 24, 2011 at 5:52 PM, Michal Bielicki
<michal.bielicki at seventhsignal.de> wrote:
> Why not use 2 profiles ?
> Am 24.08.2011 um 09:30 schrieb Pete Kelly:
>
> Hi
> Is it possible to make freeswitch listen on more than one IP for requests
> within the same profile?
> I would like to assign a VIP to a pair of freeswitch boxes, for the purposes
> of failover if one of the boxes should fail or go down. The VIP can then be
> used by web applications which have to send XMLRPC requests to freeswitch.
> In addition I would like each of the boxes to listen on their "normal" eth0
> IP too, so systems like opensips which support load balancing can evenly
> distribute load across the two boxes.
> My profile.xml file currently has this entry:
> <param name="sip-ip" value="$${local_ip_v4}"/>
> So I can't see how freeswitch can be told to listen on more than one IP for
> a profile.
> Is this possible?
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> Michal Bielicki
> Gesch?ftsf?hrer / CEO
> Seventh Signal Ltd. & Co. KG
> Weigandufer 45, B?ro 115,?D-12059 Berlin
> Voice: +49 30?60988730
> Amtsgericht Charlottenburg HRA 44413 B
> Ust.-ID: DE266981999
> Gesch?ftsf?hrer: Michal Bielicki
> Pers?nlich Haftende Gesellschafterin:
> Seventh Signal Ltd, 69 Great Hampton St. Birmingham,
> B18 6EW, GB, Company Nr.: 06889439
> WWW.:?http://www.seventhsignal.de
>
> ----
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From sg at novetys.com  Wed Aug 24 12:16:39 2011
From: sg at novetys.com (=?iso-8859-1?Q?S=E9bastien_Gay?=)
Date: Wed, 24 Aug 2011 10:16:39 +0200
Subject: [Freeswitch-users] Problem with freeswith and a Digium card
In-Reply-To: <4E54017A.7070504@anew.com.ve>
References: <CAOu2boE+ZeWmFbJDZ=OB2e+giSeTFJhGWzJuUaDkxgMVVP+SrQ@mail.gmail.com>
	<4E54017A.7070504@anew.com.ve>
Message-ID: <D9DA40A1-F2F5-4D44-BFF6-11BFA99A8BFC@novetys.com>

I think it's good.
I called and the application it is launched.


2011-08-24 10:11:07.817150 [NOTICE] mod_dptools.c:1619 Enabling tone detection 'busy' '425'

As soon as I hung up, he did his job

2011-08-24 10:11:15.997151 [DEBUG] switch_ivr_async.c:2525 TONE busy HIT 1/3
2011-08-24 10:11:17.037144 [DEBUG] switch_ivr_async.c:2525 TONE busy HIT 2/3
2011-08-24 10:11:18.077144 [DEBUG] switch_ivr_async.c:2525 TONE busy HIT 3/3
2011-08-24 10:11:18.077144 [DEBUG] switch_ivr_async.c:2531 TONE busy DETECTED
2011-08-24 10:11:18.077144 [DEBUG] switch_core_session.c:994 Send signal FreeTDM/3:3/ [BREAK]
2011-08-24 10:11:18.077144 [DEBUG] switch_ivr.c:576 FreeTDM/3:3/ Command Execute hangup(normal_clearing)
EXECUTE FreeTDM/3:3/ hangup(normal_clearing)
2011-08-24 10:11:18.077144 [DEBUG] switch_channel.c:2767 (FreeTDM/3:3/) Callstate Change ACTIVE -> HANGUP
2011-08-24 10:11:18.077144 [NOTICE] mod_dptools.c:916 Hangup FreeTDM/3:3/ [CS_EXECUTE] [NORMAL_CLEARING]
Le 23 ao?t 2011 ? 21:37, Saugort Dario Garcia Tovar a ?crit :

I left in the reverse polarity configuration.
I do not know if this solution is unique, but it has the merit of work.

Thank you all

> You are trying to detect hangup or disconnect tone?
> 
> If you trying to disconnect tone, I was playing around with this just today,
> 
> I have a custom xml to manage my digium card. The file is located in /usr/local/freeswitch/conf/dialplan/public/01_freetdm_in.xml
> 
> And it is what is has:
> 
> <include>
>         <extension name="freetdm_in">
>                 <condition field="chan_name" expression="FreeTDM/2:1/">
>                         <action application="tone_detect" data="busy 425 r 0 hangup 'normal_clearing' 3"/>
>                         <action application="set" data="domain_name=$${domain}"/>
>                         <action application="set" data="transfer_ringback=$${fr-ring}"/>
>                         <action application="answer"/>
>                         <action application="transfer" data="6442 XML default"/>
>                 </condition>
>         </extension>
> </include>
> 
> 
> I think, you could play with data="busy 425 r 0 hangup 'normal_clearing' 3" to try to get what you need. Take a look to this page: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect
> 
> About ANI 000000000 could be because you dont have caller id enable or it is not supported by your pstn.
> 
> 
> On 8/23/2011 1:53 PM, Caez wrote:
>> 
>> Hello,
>> 
>> I have to problems with freeswith and my digium card whith dahdi and freetdm.
>> I'm a french user.
>> 
>> - I have no signal when the user hangs up.
>> I have to wait for the timeout before the line is available.
>> If I call a specific phone, I have to wait for the timeout to 60s
>> before using the line
>> 
>> - I don't have the phone number for the incoming and outgoing calls.
>> For the incoming, i have 00000000000
>> For the outgoing, "Anonymous"
>> 
>> 
>> I'm trying to change my tones.conf files, to add some options on
>> freetdm.conf.xml, but the problem persists
>> 
>> Does anyone have a solution?
>> 
>> Thank you in advance
>> 
>> Sebastien
>> 
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> -----
>> No virus found in this message.
>> Checked by AVG - www.avg.com
>> Version: 10.0.1392 / Virus Database: 1520/3852 - Release Date: 08/23/11
>> 
>> 
> 
> 
> -- 
> Atentamente, 
> Dario Garc?a 
> Consultor. 
> 
> CCCT, Nivel C2, Sector Yarey, Mz, 
> Ofc. MZ03a. 
> Caracas-Venezuela. 
> Tel?fono: +58 212 9081842 
> Cel: +58 412 2221515 
> dgarcia at anew.com.ve 
> http://www.anew.com.ve
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110824/4d01867b/attachment.html 

From ankitwalia4u at gmail.com  Wed Aug 24 12:36:04 2011
From: ankitwalia4u at gmail.com (ankIT WALiA)
Date: Wed, 24 Aug 2011 14:06:04 +0530
Subject: [Freeswitch-users] V92 modem with freeswitch
In-Reply-To: <4E1D9276.6070400@anew.com.ve>
References: <CAOcuKucME9wAXkUfVPRWXTzqTs0WhtYyEr=n1KNLUbw0L9PCxw@mail.gmail.com>
	<4E1D9276.6070400@anew.com.ve>
Message-ID: <CAOcuKuc8y102tFum=3Q+wEwa9FfZpjVyiiFR7dYmz6AGgEABtQ@mail.gmail.com>

Thanks for your response. I have tested VOIP mode and SIP which worked
fine.

But I could not find any VOIP service provider in India which is giving me
local number to receive a call except cloud based system like kookoo.in

Alternatively, I thought to buy TDM400P or A200 for my PSTN line. I found
two cards suiting my requirement and budget as of now.

1. A20101 2FXS 2FXO analog card PCI ($ 325 + 5 % tax)

2. 1TDM411BF 4 port modular analog PCI 3.3/5.0V card with 1 Station and 1
Trunk interface( $ 270 + 5 % tax)

I need your advice to select the one. For the time being, I need one FXO and
one FXS but I have one doubt about echo. I read about echo problem, how can
I remove that. I read that there are software for echo cancellation and
there are some hardware for echo cancellation. Please advice.

Thanks
Ankit

On Wed, Jul 13, 2011 at 6:11 PM, Saugort Dario Garcia Tovar <
dgarcia at anew.com.ve> wrote:

> **
> Hi Ankit,
>
> You could use your V.92 56k PCI Modem old hardware but you should have to
> build a driver for Freeswitch, Freeswitch have a list of hardware that have
> been tested.
>
> I recomend you to take a look to freeswitch web site. If you would iike to
> test telephony functions (in VoIP mode, SIP) you dont need telephony
> hardware. But if you would like interface your FS box with a PSTN line/ POTS
> buy hardware from digium or sangoma like TDM400P (digium) or (A200); they
> are quite cheap, around $400 or less.
>
>
>
>
> On 7/13/2011 4:44 AM, ankIT WALiA wrote:
>
> Hi all,
>
>  I found a V.92 56k PCI Modem in my old hardware.
>
> I want to run some basic functionality test with my PSTN phone line and FS.
>
> Can I connect my phone line with FS using this card. If yes how do we configure to use this card.
>
> I am new to telephony. I don't know if whatever I said above make sense.
>
> Thanks
>
> Ankit
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 10.0.1390 / Virus Database: 1516/3761 - Release Date: 07/12/11
>
>
>
> --
> Atentamente,
> *Dario Garc?a*
> Consultor.
>
> CCCT, Nivel C2, Sector Yarey, Mz,
> Ofc. MZ03a.
> Caracas-Venezuela.
> Tel?fono: +58 212 9081842
> Cel: +58 412 2221515
> dgarcia at anew.com.ve
> http://www.anew.com.ve
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Life is like a rose its upto u feel it as its fragrance or thorns
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110824/1f053bc1/attachment.html 

From pkelly at gmail.com  Wed Aug 24 13:10:33 2011
From: pkelly at gmail.com (Pete Kelly)
Date: Wed, 24 Aug 2011 10:10:33 +0100
Subject: [Freeswitch-users] Making Freeswitch listen on 2 IP addresses
In-Reply-To: <CAO+9iGcghRhxkdMAukA44bH+54Q4Y+2SvRPxG2zUxZ9QG_x4FA@mail.gmail.com>
References: <CAEWH9M-yKO+7ko_wOsxHTEp-Rp7ZRr3+wuMvU=MpquuCFCQ=nA@mail.gmail.com>
	<2C2837D4-14B7-4702-AFD4-767CDDA8CAC3@seventhsignal.de>
	<CAO+9iGcghRhxkdMAukA44bH+54Q4Y+2SvRPxG2zUxZ9QG_x4FA@mail.gmail.com>
Message-ID: <CAEWH9M91Q9DbDmNj+DqPvfKi1KNLb+ZiGRe5gkV2zOifszhAgg@mail.gmail.com>

Thanks for the replies, I'm going to give the 2 profiles a go.


On 24 August 2011 09:13, Alec Taylor <alec.taylor6 at gmail.com> wrote:

> Why not setup DNS?
>
> On Wed, Aug 24, 2011 at 5:52 PM, Michal Bielicki
> <michal.bielicki at seventhsignal.de> wrote:
> > Why not use 2 profiles ?
> > Am 24.08.2011 um 09:30 schrieb Pete Kelly:
> >
> > Hi
> > Is it possible to make freeswitch listen on more than one IP for requests
> > within the same profile?
> > I would like to assign a VIP to a pair of freeswitch boxes, for the
> purposes
> > of failover if one of the boxes should fail or go down. The VIP can then
> be
> > used by web applications which have to send XMLRPC requests to
> freeswitch.
> > In addition I would like each of the boxes to listen on their "normal"
> eth0
> > IP too, so systems like opensips which support load balancing can evenly
> > distribute load across the two boxes.
> > My profile.xml file currently has this entry:
> > <param name="sip-ip" value="$${local_ip_v4}"/>
> > So I can't see how freeswitch can be told to listen on more than one IP
> for
> > a profile.
> > Is this possible?
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> > Michal Bielicki
> > Gesch?ftsf?hrer / CEO
> > Seventh Signal Ltd. & Co. KG
> > Weigandufer 45, B?ro 115, D-12059 Berlin
> > Voice: +49 30 60988730
> > Amtsgericht Charlottenburg HRA 44413 B
> > Ust.-ID: DE266981999
> > Gesch?ftsf?hrer: Michal Bielicki
> > Pers?nlich Haftende Gesellschafterin:
> > Seventh Signal Ltd, 69 Great Hampton St. Birmingham,
> > B18 6EW, GB, Company Nr.: 06889439
> > WWW.: http://www.seventhsignal.de
> >
> > ----
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110824/ef0e0460/attachment-0001.html 

From acrow at integrafin.co.uk  Wed Aug 24 13:22:13 2011
From: acrow at integrafin.co.uk (Alex Crow)
Date: Wed, 24 Aug 2011 10:22:13 +0100
Subject: [Freeswitch-users] Problem with freeswith and a Digium card
In-Reply-To: <07684A03-CF0F-471D-A0FF-7A3195BE2480@novetys.com>
References: <CAOu2boE+ZeWmFbJDZ=OB2e+giSeTFJhGWzJuUaDkxgMVVP+SrQ@mail.gmail.com>	<6491f89817632ed80648a716ee338470@thom.fr.eu.org>
	<07684A03-CF0F-471D-A0FF-7A3195BE2480@novetys.com>
Message-ID: <4E54C2C5.9040808@integrafin.co.uk>

On 24/08/11 09:01, S?bastien Gay wrote:
> Hi,
>
> Thank you for your answers.
>
> @Alex
> The card is a Digium  TDM400P with 1 FXS and 3 FXO :
>
> active=yes
> alarms=OK
> description=Wildcard TDM400P REV I Board 5
> name=WCTDM/4
> manufacturer=Digium
> devicetype=Wildcard TDM400P REV I
> location=PCI Bus 00 Slot 07
> basechan=1
> totchans=4
> irq=17
> type=analog
> port=1,FXS
> port=2,FXO
> port=3,FXO
> port=4,FXO
>
> @ Fran?ois
> I have tried some options in freetdm:
> <param name="answer-polarity-reverse" value="*true*" />
>
> <param name="hangup-polarity-reverse" value="*true*" />
>
> But the problem persists
>
>
> @Dario Garcia
>
> I did not know the option tone_detect, I'll do some tests.
>
>
> Thank you again for your help.
>
*
*S?bastien,
*
*Sounds like an issue I have been having. I wanted to have FS only act 
as a voicemail box, with the incoming land line connecting to both the 
FXO port and analog phones via a splitter. I wanted the voicemail to 
only pick up after 30s of ringing, however even if the incoming line had 
stopped ringing (and a polarity reverse was seen) the dialplan would 
still execute the VM and thus record a few seconds of dialtone. I saw 
the polarity reversal when the line stopped ringing but FS still claimed 
it was too close the the previous one even after more than 20s (the 
reversal time limit is set to something like 200ms) and did not hang up.

The VM would also kick in even when the analog phone was on a call after 
being picked up. There appears to be no way to test if an incoming 
analog line is still ringing or not (you can't do tone_detect here as 
the line is onhook).


Dialplan:

<include>
<extension name="public_ftdm">
<condition field="source" expression="freetdm">
<action application="sleep" data="30000"/>
<action application="answer"/>
<action application="voicemail" data="default ${domain_name} 1005"/>
</condition>
</extension>
</include>

However if I transfer the call to a SIP extension, remove the sleep and 
let the extension deal with the VM then it seems to work OK. Obviously 
this means my analog phones only ring for a moment though.

Alex

**

-- 
This message is intended only for the addressee and may contain
confidential information.  Unless you are that person, you may not
disclose its contents or use it in any way and are requested to delete
the message along with any attachments and notify us immediately.

"Transact" is operated by Integrated Financial Arrangements plc
Domain House, 5-7 Singer Street, London  EC2A 4BQ
Tel: (020) 7608 4900 Fax: (020) 7608 5300
(Registered office: as above; Registered in England and Wales under number: 3727592)
Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856)

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110824/ebb9741d/attachment.html 

From juraj.fabo at gmail.com  Wed Aug 24 16:47:55 2011
From: juraj.fabo at gmail.com (jfabo)
Date: Wed, 24 Aug 2011 05:47:55 -0700 (PDT)
Subject: [Freeswitch-users] freetdm api - ftdm_channel_read()
In-Reply-To: <CAA4nhyDc9i=raCOk1mzbJ3vVJYSAAKHGwT0b7wyMbaZrA+Wz=Q@mail.gmail.com>
References: <CAAqVCyTi9m0+JPhjt_sLLkEa97ijFLmETjPoFRUcEQJRmQsN8w@mail.gmail.com>
	<CAA4nhyDc9i=raCOk1mzbJ3vVJYSAAKHGwT0b7wyMbaZrA+Wz=Q@mail.gmail.com>
Message-ID: <1314190075421-6719985.post@n2.nabble.com>

Hi Moises,

On Wed, Jul 27, 2011 at 1:21 PM, Juraj Fabo &lt;juraj.fabo at gmail.com&gt;
wrote:
>> According to return value of sangoma_get_rx_queue_sz() the default rx
>> queue size is 10.

>That is correct. And you can change that in the FreeTDM wanpipe.conf
>configuration to set a different default or by using
>FTDM_COMMAND_SET_RX/TX_QUEUE_SIZE

My target platform is SLES 11. I installed wanpipe-3.5.20 
but did not find the queue setting within wanpipe.conf 
(at least not explicitely as queue size)
However, setting it via libfreetdm API was successful.

>> Actually, why the ftdm_channel_read() does not read them ALL at once
>> and sets the number of read bytes via *datalen output parameter?
>> (assuming provided *data buffer is large enough)
>> In my tests, always 160 bytes were returned in one particular read.

>Did you verify this by providing bigger buffer and datalen?

>The reason is that you typically want to handle voice as soon as is
>available, otherwise you add delay in your audio path, therefore you
>never leave your voice to accumulate in the driver's buffer's.
>FreeSWITCH and all other users of the FreeTDM API work this way.

Yes, I just have different framing on the second network stack = 20ms,
 while on the E1 I have 10ms, that was why I asked about the reading method.
Since read is done by polling, I would like to initiate it from the other 
network stack thread (here I use ACE framework and 
have all streams/mixers triggered from single Reactor/thread)

My problem is that I'm experiencing unexpected latency in the test
environment. 
I have two server, first with sangoma a104d and second with 2xa102de cards
 connected with E1 cables. First server is serving as a 'loopback',
configured 
in telko mode and all it does is that in loop it reads all active channels
 and immediatelly writes read data to the same channel. 
My measurements show latency about 180ms from the write
 on the second server till read of the same speach on the second serve
r in case of single call.
I tried to reduce the queue size as described above, 
however it seems that some frames got lost then.

My intention was to use single thread to handle read/write operations 
on all active channels, which is different than the approach described
 in freetdm/doc/locking.txt (where thread per call leg is mentioned).
I worry most about calling the wait_channel() with timeout = -1 which 
afaik leads to blocking wait on single channel. As mentioned earlier, 
when this is called from single thread prio each particular
ftdm_channel_write()
 it represents danger of blocking and latency. I measure time spent 
in each particular channel wait+write and the result was < 10us.

Maybe rather a simple question, does the application which uses 
libfreetdm need to run thread per call/channel?


>In the future you might want consider asking this questions in the
>freeswitch-dev mailing list to have better chances of a prompt
>response (as long as the question involves C development with
>FreeSWITCH/FreeTDM internal APIs).

should I repost also this?

Thanks

Juraj

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/freetdm-api-ftdm-channel-read-tp6627495p6719985.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From dave.redmore at spigotnetworks.com  Wed Aug 24 19:33:20 2011
From: dave.redmore at spigotnetworks.com (dave.redmore at spigotnetworks.com)
Date: Wed, 24 Aug 2011 10:33:20 -0500 (CDT)
Subject: [Freeswitch-users] Slightly OT - Freeswitch and Vitelity?
In-Reply-To: <32999431.111314199967787.JavaMail.root@zimbra1.spigotsystems.com>
Message-ID: <19689347.131314200000670.JavaMail.root@zimbra1.spigotsystems.com>

Anyone have any experiance with Vitelity as an ITSP? Curious about both interop with Freeswitch and in general. 


Thanks, 


Dave Redmore 
Spigot Networks, Inc. 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110824/4897aed2/attachment.html 

From erwin at merioles.net  Wed Aug 24 16:31:13 2011
From: erwin at merioles.net (Erwin Merioles)
Date: Wed, 24 Aug 2011 20:31:13 +0800
Subject: [Freeswitch-users] FreeSwitch Utilizes 100% of CPU Sometimes When
	User Quits Conference
Message-ID: <002601cc6259$b8b8ec40$2a2ac4c0$@Merioles.net>

Hey guys,

 

I've been having trouble with FreeSwitch for quite some time. We're trying
to use mod_rtmp to add sound to one of our applications, www.321meet.com.
Unfortunately, FreeSwitch's CPU usage spikes when the host ( the first one
to join the conference ), quits or closes the browser window. I've checked
and this always happen when the following line is called -

 

2011-08-24 12:24:28.057200 [NOTICE] rtmp_tcp.c:73 Pollout: true

 

FS Console Log follows :

 

2011-08-24 12:22:59.177294 [NOTICE] mod_rtmp.c:743 New RTMP session
[4a97320c-e50d-4ed3-a59c-4aef799d379d]

2011-08-24 12:22:59.477278 [NOTICE] rtmp_sig.c:121 Sent connect reply

2011-08-24 12:23:15.597196 [INFO] rtmp_sig.c:136 Replied to createStream (1)

2011-08-24 12:23:16.237201 [INFO] rtmp_sig.c:274 Got publish on stream 1.

2011-08-24 12:23:17.177293 [INFO] rtmp_sig.c:136 Replied to createStream (2)

2011-08-24 12:23:17.177293 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled
control packet (type=0x3)

2011-08-24 12:23:17.457195 [WARNING] sofia.c:4403 Ping succeeded
voip9.telsome.com with code 404 - count -1/1/1, state UP

2011-08-24 12:23:17.457195 [INFO] rtmp_sig.c:136 Replied to createStream (3)

2011-08-24 12:23:17.457195 [NOTICE] switch_channel.c:897 New Channel
rtmp/default/3213533 [8e7ccce8-172c-4b91-9183-1ad9d2f0e6dd]

2011-08-24 12:23:17.457195 [ERR] rtmp_sig.c:305 Couldn't create call.

2011-08-24 12:23:17.457195 [WARNING] sofia.c:4403 Ping succeeded testin with
code 404 - count -1/1/1, state UP

2011-08-24 12:23:17.497199 [INFO] mod_dialplan_xml.c:336 Processing
<0000000000>->3213533 in context default

2011-08-24 12:23:17.497199 [NOTICE] mod_rtmp.c:497 Channel
[rtmp/default/3213533] has been answered

2011-08-24 12:23:17.497199 [INFO] mod_conference.c:6644 using channel sound
prefix: /usr/local/freeswitch/sounds/en/us/callie

2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 rtmp/default/3213533
binding '0' to 'mute'

2011-08-24 12:23:17.597289 [INFO] switch_ivr_async.c:164 Digit parser
mod_conference: Setting realm to conf

2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 rtmp/default/3213533
binding '*' to 'deaf mute'

2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 rtmp/default/3213533
binding '9' to 'energy up'

2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 rtmp/default/3213533
binding '8' to 'energy equ'

2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 rtmp/default/3213533
binding '7' to 'energy dn'

2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 rtmp/default/3213533
binding '3' to 'vol talk up'

2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 rtmp/default/3213533
binding '2' to 'vol talk zero'

2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 rtmp/default/3213533
binding '1' to 'vol talk dn'

2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 rtmp/default/3213533
binding '6' to 'vol listen up'

2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 rtmp/default/3213533
binding '5' to 'vol listen zero'

2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 rtmp/default/3213533
binding '4' to 'vol listen dn'

2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 rtmp/default/3213533
binding '#' to 'hangup'

2011-08-24 12:23:17.797195 [INFO] rtmp_sig.c:159 Sending audio

2011-08-24 12:23:17.797195 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled
control packet (type=0x3)

2011-08-24 12:23:18.117290 [INFO] rtmp_sig.c:274 Got publish on stream 3.

2011-08-24 12:23:36.697195 [ERR] rtmp.c:678 Read error

2011-08-24 12:23:36.697195 [NOTICE] mod_rtmp.c:788 RTMP session ended
[4a97320c-e50d-4ed3-a59c-4aef799d379d]

2011-08-24 12:23:36.697195 [NOTICE] mod_rtmp.c:803 Hangup
rtmp/default/3213533 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]

2011-08-24 12:23:36.777294 [NOTICE] switch_core_session.c:1347 Session 1
(rtmp/default/3213533) Ended

2011-08-24 12:23:36.777294 [NOTICE] switch_core_session.c:1349 Close Channel
rtmp/default/3213533 [CS_DESTROY]

2011-08-24 12:23:48.237202 [NOTICE] mod_rtmp.c:743 New RTMP session
[3ff4786f-7cae-4653-879a-ae95a9d50742]

2011-08-24 12:23:48.557204 [NOTICE] rtmp_sig.c:121 Sent connect reply

2011-08-24 12:23:55.357293 [NOTICE] mod_rtmp.c:743 New RTMP session
[af763930-9054-4c76-a0eb-9c351e75949d]

2011-08-24 12:23:55.757290 [NOTICE] rtmp_sig.c:121 Sent connect reply

2011-08-24 12:23:58.957195 [INFO] rtmp_sig.c:136 Replied to createStream (1)

2011-08-24 12:23:59.517204 [INFO] rtmp_sig.c:274 Got publish on stream 1.

2011-08-24 12:24:00.557201 [INFO] rtmp_sig.c:136 Replied to createStream (2)

2011-08-24 12:24:00.557201 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled
control packet (type=0x3)

2011-08-24 12:24:00.557201 [INFO] rtmp_sig.c:136 Replied to createStream (3)

2011-08-24 12:24:00.857222 [NOTICE] switch_channel.c:897 New Channel
rtmp/default/3213533 [5de55413-5cb9-4c18-8b65-9c438833449f]

2011-08-24 12:24:00.857222 [ERR] rtmp_sig.c:305 Couldn't create call.

2011-08-24 12:24:00.857222 [INFO] mod_dialplan_xml.c:336 Processing
<0000000000>->3213533 in context default

2011-08-24 12:24:00.857222 [NOTICE] mod_rtmp.c:497 Channel
[rtmp/default/3213533] has been answered

2011-08-24 12:24:00.857222 [INFO] mod_conference.c:6644 using channel sound
prefix: /usr/local/freeswitch/sounds/en/us/callie

2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 rtmp/default/3213533
binding '0' to 'mute'

2011-08-24 12:24:00.857222 [INFO] switch_ivr_async.c:164 Digit parser
mod_conference: Setting realm to conf

2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 rtmp/default/3213533
binding '*' to 'deaf mute'

2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 rtmp/default/3213533
binding '9' to 'energy up'

2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 rtmp/default/3213533
binding '8' to 'energy equ'

2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 rtmp/default/3213533
binding '7' to 'energy dn'

2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 rtmp/default/3213533
binding '3' to 'vol talk up'

2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 rtmp/default/3213533
binding '2' to 'vol talk zero'

2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 rtmp/default/3213533
binding '1' to 'vol talk dn'

2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 rtmp/default/3213533
binding '6' to 'vol listen up'

2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 rtmp/default/3213533
binding '5' to 'vol listen zero'

2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 rtmp/default/3213533
binding '4' to 'vol listen dn'

2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 rtmp/default/3213533
binding '#' to 'hangup'

2011-08-24 12:24:01.197247 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled
control packet (type=0x3)

2011-08-24 12:24:01.197247 [INFO] rtmp_sig.c:274 Got publish on stream 3.

2011-08-24 12:24:01.197247 [INFO] rtmp_sig.c:159 Sending audio

2011-08-24 12:24:13.637291 [INFO] rtmp_sig.c:136 Replied to createStream (1)

2011-08-24 12:24:14.157289 [INFO] rtmp_sig.c:274 Got publish on stream 1.

2011-08-24 12:24:17.057195 [INFO] rtmp_sig.c:136 Replied to createStream (2)

2011-08-24 12:24:17.057195 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled
control packet (type=0x3)

2011-08-24 12:24:17.057195 [INFO] rtmp_sig.c:136 Replied to createStream (3)

2011-08-24 12:24:17.057195 [NOTICE] switch_channel.c:897 New Channel
rtmp/default/3213533 [4e6c35ce-ec56-4ef1-816a-b94704efbfbd]

2011-08-24 12:24:17.057195 [ERR] rtmp_sig.c:305 Couldn't create call.

2011-08-24 12:24:17.057195 [INFO] mod_dialplan_xml.c:336 Processing
<0000000000>->3213533 in context default

2011-08-24 12:24:17.057195 [NOTICE] mod_rtmp.c:497 Channel
[rtmp/default/3213533] has been answered

2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 rtmp/default/3213533
binding '0' to 'mute'

2011-08-24 12:24:17.057195 [INFO] switch_ivr_async.c:164 Digit parser
mod_conference: Setting realm to conf

2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 rtmp/default/3213533
binding '*' to 'deaf mute'

2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 rtmp/default/3213533
binding '9' to 'energy up'

2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 rtmp/default/3213533
binding '8' to 'energy equ'

2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 rtmp/default/3213533
binding '7' to 'energy dn'

2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 rtmp/default/3213533
binding '3' to 'vol talk up'

2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 rtmp/default/3213533
binding '2' to 'vol talk zero'

2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 rtmp/default/3213533
binding '1' to 'vol talk dn'

2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 rtmp/default/3213533
binding '6' to 'vol listen up'

2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 rtmp/default/3213533
binding '5' to 'vol listen zero'

2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 rtmp/default/3213533
binding '4' to 'vol listen dn'

2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 rtmp/default/3213533
binding '#' to 'hangup'

2011-08-24 12:24:17.497196 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled
control packet (type=0x3)

2011-08-24 12:24:17.497196 [INFO] rtmp_sig.c:274 Got publish on stream 3.

2011-08-24 12:24:17.497196 [INFO] rtmp_sig.c:159 Sending audio

2011-08-24 12:24:27.657196 [ERR] rtmp.c:678 Read error

2011-08-24 12:24:27.657196 [NOTICE] mod_rtmp.c:788 RTMP session ended
[3ff4786f-7cae-4653-879a-ae95a9d50742]

2011-08-24 12:24:27.657196 [NOTICE] mod_rtmp.c:803 Hangup
rtmp/default/3213533 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]

2011-08-24 12:24:27.677201 [NOTICE] switch_core_session.c:1347 Session 2
(rtmp/default/3213533) Ended

2011-08-24 12:24:27.677201 [NOTICE] switch_core_session.c:1349 Close Channel
rtmp/default/3213533 [CS_DESTROY]

2011-08-24 12:24:28.057200 [NOTICE] rtmp_tcp.c:73 Pollout: true

 

Which results to :

 

  PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+  COMMAND

1608 root      -2 -10  191m  20m 6588 R 94.3  3.5   5:39.50 fs

 

Any ideas? Help is VERY much appreciated. TIA!

 

Regards,

 

Erwin D. Merioles

 

merioles.net

+63 922 837 9466 | +63 917 501 1010 | +1 760 670 3241

aY!M : erwin_merioles | Skype : erwin.merioles

This message (including any attachments) contains information that may be
confidential. Unless you are the intended recipient (or is authorized to
receive for the intended recipient), you may not read, print, retain, use,
copy, distribute, or disclose to anyone, any information contained here. If
you have received this in error, please advise the sender by reply e-mail,
and delete all copies of the original message (including attachments).

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110824/61b9877f/attachment-0001.html 

From msc at freeswitch.org  Wed Aug 24 19:39:37 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 24 Aug 2011 08:39:37 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today
Message-ID: <CAKzWOxUtDK5bLDiUrLbN-da4JiX6PGHeOUD=H_8u8EfvZ_73Rg@mail.gmail.com>

Hello all!

Today's agenda is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2011_08_24

Alexandr from the Homer project has been busy updating his software so we're
having him back to talk to us again on the subject. We also have a few
janitorial items to go over.

Thanks!

-Michael
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110824/bb28a1b1/attachment.html 

From anthony.minessale at gmail.com  Wed Aug 24 20:01:02 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 24 Aug 2011 11:01:02 -0500
Subject: [Freeswitch-users] FreeSwitch Utilizes 100% of CPU Sometimes
 When User Quits Conference
In-Reply-To: <002601cc6259$b8b8ec40$2a2ac4c0$@Merioles.net>
References: <002601cc6259$b8b8ec40$2a2ac4c0$@Merioles.net>
Message-ID: <CAKbxfG8SAzOYxvi==cCEcxzq5a0QUiYV3MztxizMC71y3RbtLQ@mail.gmail.com>

Really you should be reporting bugs to http://jira.freeswitch.org
Are you only having this problem with mod_rtmp (its only 2 months old)
Most likely you have created a new condition that the author has not
taken into account.  Ideally you should file it on jira under mod_rtmp
and attach a back trace from a core dump produced by gcore.



On Wed, Aug 24, 2011 at 7:31 AM, Erwin Merioles <erwin at merioles.net> wrote:
> Hey guys,
>
>
>
> I?ve been having trouble with FreeSwitch for quite some time. We?re trying
> to use mod_rtmp to add sound to one of our applications, www.321meet.com.
> Unfortunately, FreeSwitch?s CPU usage spikes when the host ( the first one
> to join the conference ), quits or closes the browser window. I?ve checked
> and this always happen when the following line is called ?
>
>
>
> 2011-08-24 12:24:28.057200 [NOTICE] rtmp_tcp.c:73 Pollout: true
>
>
>
> FS Console Log follows :
>
>
>
> 2011-08-24 12:22:59.177294 [NOTICE] mod_rtmp.c:743 New RTMP session
> [4a97320c-e50d-4ed3-a59c-4aef799d379d]
>
> 2011-08-24 12:22:59.477278 [NOTICE] rtmp_sig.c:121 Sent connect reply
>
> 2011-08-24 12:23:15.597196 [INFO] rtmp_sig.c:136 Replied to createStream (1)
>
> 2011-08-24 12:23:16.237201 [INFO] rtmp_sig.c:274 Got publish on stream 1.
>
> 2011-08-24 12:23:17.177293 [INFO] rtmp_sig.c:136 Replied to createStream (2)
>
> 2011-08-24 12:23:17.177293 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled
> control packet (type=0x3)
>
> 2011-08-24 12:23:17.457195 [WARNING] sofia.c:4403 Ping succeeded
> voip9.telsome.com with code 404 - count -1/1/1, state UP
>
> 2011-08-24 12:23:17.457195 [INFO] rtmp_sig.c:136 Replied to createStream (3)
>
> 2011-08-24 12:23:17.457195 [NOTICE] switch_channel.c:897 New Channel
> rtmp/default/3213533 [8e7ccce8-172c-4b91-9183-1ad9d2f0e6dd]
>
> 2011-08-24 12:23:17.457195 [ERR] rtmp_sig.c:305 Couldn't create call.
>
> 2011-08-24 12:23:17.457195 [WARNING] sofia.c:4403 Ping succeeded testin with
> code 404 - count -1/1/1, state UP
>
> 2011-08-24 12:23:17.497199 [INFO] mod_dialplan_xml.c:336 Processing
> <0000000000>->3213533 in context default
>
> 2011-08-24 12:23:17.497199 [NOTICE] mod_rtmp.c:497 Channel
> [rtmp/default/3213533] has been answered
>
> 2011-08-24 12:23:17.497199 [INFO] mod_conference.c:6644 using channel sound
> prefix: /usr/local/freeswitch/sounds/en/us/callie
>
> 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 rtmp/default/3213533
> binding '0' to 'mute'
>
> 2011-08-24 12:23:17.597289 [INFO] switch_ivr_async.c:164 Digit parser
> mod_conference: Setting realm to conf
>
> 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 rtmp/default/3213533
> binding '*' to 'deaf mute'
>
> 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 rtmp/default/3213533
> binding '9' to 'energy up'
>
> 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 rtmp/default/3213533
> binding '8' to 'energy equ'
>
> 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 rtmp/default/3213533
> binding '7' to 'energy dn'
>
> 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 rtmp/default/3213533
> binding '3' to 'vol talk up'
>
> 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 rtmp/default/3213533
> binding '2' to 'vol talk zero'
>
> 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 rtmp/default/3213533
> binding '1' to 'vol talk dn'
>
> 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 rtmp/default/3213533
> binding '6' to 'vol listen up'
>
> 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 rtmp/default/3213533
> binding '5' to 'vol listen zero'
>
> 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 rtmp/default/3213533
> binding '4' to 'vol listen dn'
>
> 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 rtmp/default/3213533
> binding '#' to 'hangup'
>
> 2011-08-24 12:23:17.797195 [INFO] rtmp_sig.c:159 Sending audio
>
> 2011-08-24 12:23:17.797195 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled
> control packet (type=0x3)
>
> 2011-08-24 12:23:18.117290 [INFO] rtmp_sig.c:274 Got publish on stream 3.
>
> 2011-08-24 12:23:36.697195 [ERR] rtmp.c:678 Read error
>
> 2011-08-24 12:23:36.697195 [NOTICE] mod_rtmp.c:788 RTMP session ended
> [4a97320c-e50d-4ed3-a59c-4aef799d379d]
>
> 2011-08-24 12:23:36.697195 [NOTICE] mod_rtmp.c:803 Hangup
> rtmp/default/3213533 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]
>
> 2011-08-24 12:23:36.777294 [NOTICE] switch_core_session.c:1347 Session 1
> (rtmp/default/3213533) Ended
>
> 2011-08-24 12:23:36.777294 [NOTICE] switch_core_session.c:1349 Close Channel
> rtmp/default/3213533 [CS_DESTROY]
>
> 2011-08-24 12:23:48.237202 [NOTICE] mod_rtmp.c:743 New RTMP session
> [3ff4786f-7cae-4653-879a-ae95a9d50742]
>
> 2011-08-24 12:23:48.557204 [NOTICE] rtmp_sig.c:121 Sent connect reply
>
> 2011-08-24 12:23:55.357293 [NOTICE] mod_rtmp.c:743 New RTMP session
> [af763930-9054-4c76-a0eb-9c351e75949d]
>
> 2011-08-24 12:23:55.757290 [NOTICE] rtmp_sig.c:121 Sent connect reply
>
> 2011-08-24 12:23:58.957195 [INFO] rtmp_sig.c:136 Replied to createStream (1)
>
> 2011-08-24 12:23:59.517204 [INFO] rtmp_sig.c:274 Got publish on stream 1.
>
> 2011-08-24 12:24:00.557201 [INFO] rtmp_sig.c:136 Replied to createStream (2)
>
> 2011-08-24 12:24:00.557201 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled
> control packet (type=0x3)
>
> 2011-08-24 12:24:00.557201 [INFO] rtmp_sig.c:136 Replied to createStream (3)
>
> 2011-08-24 12:24:00.857222 [NOTICE] switch_channel.c:897 New Channel
> rtmp/default/3213533 [5de55413-5cb9-4c18-8b65-9c438833449f]
>
> 2011-08-24 12:24:00.857222 [ERR] rtmp_sig.c:305 Couldn't create call.
>
> 2011-08-24 12:24:00.857222 [INFO] mod_dialplan_xml.c:336 Processing
> <0000000000>->3213533 in context default
>
> 2011-08-24 12:24:00.857222 [NOTICE] mod_rtmp.c:497 Channel
> [rtmp/default/3213533] has been answered
>
> 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:6644 using channel sound
> prefix: /usr/local/freeswitch/sounds/en/us/callie
>
> 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 rtmp/default/3213533
> binding '0' to 'mute'
>
> 2011-08-24 12:24:00.857222 [INFO] switch_ivr_async.c:164 Digit parser
> mod_conference: Setting realm to conf
>
> 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 rtmp/default/3213533
> binding '*' to 'deaf mute'
>
> 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 rtmp/default/3213533
> binding '9' to 'energy up'
>
> 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 rtmp/default/3213533
> binding '8' to 'energy equ'
>
> 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 rtmp/default/3213533
> binding '7' to 'energy dn'
>
> 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 rtmp/default/3213533
> binding '3' to 'vol talk up'
>
> 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 rtmp/default/3213533
> binding '2' to 'vol talk zero'
>
> 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 rtmp/default/3213533
> binding '1' to 'vol talk dn'
>
> 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 rtmp/default/3213533
> binding '6' to 'vol listen up'
>
> 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 rtmp/default/3213533
> binding '5' to 'vol listen zero'
>
> 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 rtmp/default/3213533
> binding '4' to 'vol listen dn'
>
> 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 rtmp/default/3213533
> binding '#' to 'hangup'
>
> 2011-08-24 12:24:01.197247 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled
> control packet (type=0x3)
>
> 2011-08-24 12:24:01.197247 [INFO] rtmp_sig.c:274 Got publish on stream 3.
>
> 2011-08-24 12:24:01.197247 [INFO] rtmp_sig.c:159 Sending audio
>
> 2011-08-24 12:24:13.637291 [INFO] rtmp_sig.c:136 Replied to createStream (1)
>
> 2011-08-24 12:24:14.157289 [INFO] rtmp_sig.c:274 Got publish on stream 1.
>
> 2011-08-24 12:24:17.057195 [INFO] rtmp_sig.c:136 Replied to createStream (2)
>
> 2011-08-24 12:24:17.057195 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled
> control packet (type=0x3)
>
> 2011-08-24 12:24:17.057195 [INFO] rtmp_sig.c:136 Replied to createStream (3)
>
> 2011-08-24 12:24:17.057195 [NOTICE] switch_channel.c:897 New Channel
> rtmp/default/3213533 [4e6c35ce-ec56-4ef1-816a-b94704efbfbd]
>
> 2011-08-24 12:24:17.057195 [ERR] rtmp_sig.c:305 Couldn't create call.
>
> 2011-08-24 12:24:17.057195 [INFO] mod_dialplan_xml.c:336 Processing
> <0000000000>->3213533 in context default
>
> 2011-08-24 12:24:17.057195 [NOTICE] mod_rtmp.c:497 Channel
> [rtmp/default/3213533] has been answered
>
> 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 rtmp/default/3213533
> binding '0' to 'mute'
>
> 2011-08-24 12:24:17.057195 [INFO] switch_ivr_async.c:164 Digit parser
> mod_conference: Setting realm to conf
>
> 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 rtmp/default/3213533
> binding '*' to 'deaf mute'
>
> 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 rtmp/default/3213533
> binding '9' to 'energy up'
>
> 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 rtmp/default/3213533
> binding '8' to 'energy equ'
>
> 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 rtmp/default/3213533
> binding '7' to 'energy dn'
>
> 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 rtmp/default/3213533
> binding '3' to 'vol talk up'
>
> 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 rtmp/default/3213533
> binding '2' to 'vol talk zero'
>
> 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 rtmp/default/3213533
> binding '1' to 'vol talk dn'
>
> 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 rtmp/default/3213533
> binding '6' to 'vol listen up'
>
> 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 rtmp/default/3213533
> binding '5' to 'vol listen zero'
>
> 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 rtmp/default/3213533
> binding '4' to 'vol listen dn'
>
> 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 rtmp/default/3213533
> binding '#' to 'hangup'
>
> 2011-08-24 12:24:17.497196 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled
> control packet (type=0x3)
>
> 2011-08-24 12:24:17.497196 [INFO] rtmp_sig.c:274 Got publish on stream 3.
>
> 2011-08-24 12:24:17.497196 [INFO] rtmp_sig.c:159 Sending audio
>
> 2011-08-24 12:24:27.657196 [ERR] rtmp.c:678 Read error
>
> 2011-08-24 12:24:27.657196 [NOTICE] mod_rtmp.c:788 RTMP session ended
> [3ff4786f-7cae-4653-879a-ae95a9d50742]
>
> 2011-08-24 12:24:27.657196 [NOTICE] mod_rtmp.c:803 Hangup
> rtmp/default/3213533 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]
>
> 2011-08-24 12:24:27.677201 [NOTICE] switch_core_session.c:1347 Session 2
> (rtmp/default/3213533) Ended
>
> 2011-08-24 12:24:27.677201 [NOTICE] switch_core_session.c:1349 Close Channel
> rtmp/default/3213533 [CS_DESTROY]
>
> 2011-08-24 12:24:28.057200 [NOTICE] rtmp_tcp.c:73 Pollout: true
>
>
>
> Which results to :
>
>
>
> ? PID USER????? PR? NI? VIRT? RES? SHR S %CPU %MEM??? TIME+? COMMAND
>
> 1608 root????? -2 -10? 191m? 20m 6588 R 94.3? 3.5?? 5:39.50 fs
>
>
>
> Any ideas? Help is VERY much appreciated. TIA!
>
>
>
> Regards,
>
>
>
> Erwin D. Merioles
>
>
>
> merioles.net
>
> +63 922 837 9466 | +63 917 501 1010 | +1 760 670 3241
>
> aY!M : erwin_merioles | Skype : erwin.merioles
>
> This message (including any attachments) contains information that may be
> confidential. Unless you are the intended recipient (or is authorized to
> receive for the intended recipient), you may not read, print, retain, use,
> copy, distribute, or disclose to anyone, any information contained here. If
> you have received this in error, please advise the sender by reply e-mail,
> and delete all copies of the original message (including attachments).
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From a.afzali2003 at gmail.com  Wed Aug 24 21:45:07 2011
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Wed, 24 Aug 2011 22:15:07 +0430
Subject: [Freeswitch-users] Flex Client Receives Periodic Interrupted Audio
Message-ID: <CABK0fC4fSktwbE5yUU6D+abC0VHJGbs8kv1j346qzrGQwAJyOg@mail.gmail.com>

Hi Guys,

As a taste of RTMP end point, I'm experiencing periodic interrupted
audio. Everything is as defaults in client / server sides except
that "auth-calls = false" and client does not login / logout, It just
makes call. I don't know if it can justified by buffer-len / chunksize
in server side or NetStream.bufferTime in client side.

appreciate all comment.
-- afshin


From justlikeef at gmail.com  Wed Aug 24 22:38:32 2011
From: justlikeef at gmail.com (Rob Hutton)
Date: Wed, 24 Aug 2011 14:38:32 -0400
Subject: [Freeswitch-users] Slightly OT - Freeswitch and Vitelity?
In-Reply-To: <19689347.131314200000670.JavaMail.root@zimbra1.spigotsystems.com>
References: <19689347.131314200000670.JavaMail.root@zimbra1.spigotsystems.com>
Message-ID: <201108241438.32611.justlikeef@gmail.com>

I have set up one switch using them for service and had no problem.  I have no experience on reliability or sound quality.

On Wednesday 24 August 2011 11:33:20 dave.redmore at spigotnetworks.com wrote:
> Anyone have any experiance with Vitelity as an ITSP? Curious about both interop with Freeswitch and in general. 
> 
> 
> Thanks, 
> 
> 
> Dave Redmore 
> Spigot Networks, Inc. 
> 
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110824/cf69f71f/attachment.html 

From mickkemp at gmail.com  Wed Aug 24 20:57:16 2011
From: mickkemp at gmail.com (Michael Kemp)
Date: Wed, 24 Aug 2011 17:57:16 +0100
Subject: [Freeswitch-users] mod_conference - 2-digit DTMF and events
Message-ID: <CAOFe-q7CG0TZ_u0CsNQddbB42j=H41ELbWgzMy69UsfVTkEhMQ@mail.gmail.com>

I have set up a custom caller-controls group in the my conference.conf.xml,
as follows


<caller-controls>

  <group name="mick">

      <control action="event" digits="#1"/>

      <control action="event" digits="#2"/>

      <control action="event" digits="#3"/>

      <control action="mute" digits="#6"/>

  </group>

</caller-controls>


I have a lua script running to detect these events and perform actions
according to which digits it receives.


However, unless I press both the 2nd digit very quickly after the first one
the event is not raised


Is there any way of tuning the inter digit delay for caller-control groups
so that multi-digit DTMF sequences are handled correctly?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110824/ca62ac69/attachment.html 

From brian at freeswitch.org  Thu Aug 25 00:13:41 2011
From: brian at freeswitch.org (Brian West)
Date: Wed, 24 Aug 2011 15:13:41 -0500
Subject: [Freeswitch-users] Flowroute Users
Message-ID: <2317D95A-439D-42FC-8766-1BB9582E981D@freeswitch.org>

https://support.flowroute.com/entries/20385652-inum-availability

Please go voice your support!

/b

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110824/39e25716/attachment-0001.html 

From cjbujold at accra.ca  Thu Aug 25 01:06:38 2011
From: cjbujold at accra.ca (Charles Bujold)
Date: Wed, 24 Aug 2011 18:06:38 -0300
Subject: [Freeswitch-users] Newbie - External Phone setup question
Message-ID: <006401cc62a1$b72a4840$257ed8c0$@accra.ca>

 

Trying to get a phone located in another office registered to the Freeswitch
server.  I created an "external5090.xml" in the sip external folder as per
the wiki and then set the phone to connect to 5090.  But I cannot get the
phone to register.  When I look at wireshark I see a request from the phone
to register and then I see  an error "SIP:403 Forbidden" from the server to
the Phone.    I have created the extension in the Directory folder with the
other internal extensions but I'm wondering if I'm missing something in the
1010extension.xml for it to work. (maybe the extension is located in the
wrong folder, or some identification to say it is not in the local LAN?)

 

I have checked the name and password and they are correct which seems to
indicate the issue is at the server level.

 

Any guidance on how to setup an external phone to Freeswitch would be
appreciated.

 

Thanks

 

CJB

 

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110824/080ce7a9/attachment.html 

From leon at scarlet-internet.nl  Thu Aug 25 01:47:53 2011
From: leon at scarlet-internet.nl (Leon de Rooij)
Date: Wed, 24 Aug 2011 23:47:53 +0200
Subject: [Freeswitch-users] freeswitch crashed - gdb trace
Message-ID: <7F8AFF02-C9B2-4902-A5BC-472CCDD10278@scarlet-internet.nl>

Hi all,

One of our FreeSWITCH servers crashed this morning. It happened when one of our customers suddenly started way more channels than usual (they went from about 100 to a little less than 500 channels). I have a core file and ran fscore_pb as suggested here:

http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#Simple_bash_script_to_make_debug_easy

The output of that script was saved here:

http://pastebin.freeswitch.org/17176

I'll spend all my free time to learn gdb, but for now can anyone help me find out what went wrong ? I'm a bit overwhelmed by the 16k lines that was generated by the script.

Thanks & kind regards,

Leon
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110824/b84fbf09/attachment.html 

From fs-list at communicatefreely.net  Thu Aug 25 01:48:56 2011
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Wed, 24 Aug 2011 17:48:56 -0400
Subject: [Freeswitch-users] Options ping timer adjustment?
Message-ID: <4E5571C8.2020603@communicatefreely.net>

Hello,

Does anyone know of a way to adjust how often to send the options-ping 
packets?  I really only need them to go about once every 30s, and it 
looks like they go every 10s or more.

Thanks!

-Tim


From anthony.minessale at gmail.com  Thu Aug 25 02:57:58 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 24 Aug 2011 17:57:58 -0500
Subject: [Freeswitch-users] Options ping timer adjustment?
In-Reply-To: <4E5571C8.2020603@communicatefreely.net>
References: <4E5571C8.2020603@communicatefreely.net>
Message-ID: <CAKbxfG8aTAHB2vvV7pS+gBF49sCPBpLr+79HUGOjn5bY8qBvRw@mail.gmail.com>

if you mean the one for gateways

<param name="ping" value="30"/>


On Wed, Aug 24, 2011 at 4:48 PM, Tim St. Pierre
<fs-list at communicatefreely.net> wrote:
> Hello,
>
> Does anyone know of a way to adjust how often to send the options-ping
> packets? ?I really only need them to go about once every 30s, and it
> looks like they go every 10s or more.
>
> Thanks!
>
> -Tim
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From adrottenberg at gmail.com  Thu Aug 25 04:23:47 2011
From: adrottenberg at gmail.com (Duvid Rottenberg)
Date: Wed, 24 Aug 2011 20:23:47 -0400
Subject: [Freeswitch-users] Paging with Polycom Phones
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C59EBABB8D4@cooper>
References: <CAC4JBEAF4OMFeD9KV5LSAVZ7DxTG5ceK0Vj3cDEam6kU8hTD3Q@mail.gmail.com>
	<CAKzWOxUMbUYM_+eiQG98OpAYTHdur1GVADHzY1OmfV-E+NBHDw@mail.gmail.com>
	<CAC4JBECX77JNE5htRm2nD8UOtfhVLqK2tnp3N2tnN+ep6DOAqA@mail.gmail.com>
	<CAKzWOxW27QZx=5MLKxtVZaHCr_y0zWpXUEx59_sfmDBu4C_BTA@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59EBABB8D4@cooper>
Message-ID: <CAC4JBEDF1ND_QgwUrT3xrdhC0bkCXW0XyrtyU8VGhJgdGj=G+g@mail.gmail.com>

Thanks all for the responses. I was paging a line via a gateway, it turns
out that the gateway I use was stripping out the sip headers.

On Wed, Aug 17, 2011 at 4:42 PM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> I Also tried with default settings, worked fine also..
>
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [
> freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Michael Collins [
> msc at freeswitch.org]
> Skickat: den 17 augusti 2011 22:29
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] Paging with Polycom Phones
>
> I have plain jane out-of-box Polycoms and they all seem to work. I can't
> tell you if there are any magic settings.
>
> Does anyone else know if there are specific settings on the Polys that need
> to be set in order for the mad-boss to work?
>
> -MC
>
> On Wed, Aug 17, 2011 at 11:33 AM, Duvid Rottenberg <adrottenberg at gmail.com
> <mailto:adrottenberg at gmail.com>> wrote:
> The mad boss example looks like it uses the same commands as the
> conferencing and intercom sample. I copied the  few settings that appear to
> be different, but I'm still having the issue that the polycom phone answers
> and hangs up right away. Did you have to make any changes to your polycom
> config file to get this to work?
>
> Thanks,
> Duvid
>
> On Wed, Aug 17, 2011 at 12:51 PM, Michael Collins <msc at freeswitch.org
> <mailto:msc at freeswitch.org>> wrote:
> Try using the "mad boss" example found in the default dialplan. I've tested
> that with Polycom phones. It's a nice workaround for all these different
> phone vendors who do things so differently.
>
> -MC
>
> On Wed, Aug 17, 2011 at 8:55 AM, Duvid Rottenberg <adrottenberg at gmail.com
> <mailto:adrottenberg at gmail.com>> wrote:
> Has anyone successfully implemented paging (auto-answer) with a polycom
> phone?
> I am using the Conferencing and Intercom sample which sets the
> sip_auto_answer variable to true, however on my polycom phone the result is
> that the phone rings once and hangs up right away (the phone is sending a
> BYE message). I tried adding an Alert-Info header, however it seems that the
> Polycom format (Alert-Info: Ring Answer) isn't compliant with the RFC and I
> couldn't get freeswitch to send the header in this format.
>
> Has anyone else been able to either get polycom phones to work with
> sip_auto_answer or to get freeswitch to send an Alert-Info header in the
> polycom format?
>
> Thanks,
> Duvid Rottenberg
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org<http://www.freeswitch.org/>
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org<http://www.freeswitch.org/>
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:
> FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> !DSPAM:4e4c25df32761815520130!
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110824/ce404f71/attachment.html 

From adrottenberg at gmail.com  Thu Aug 25 04:41:58 2011
From: adrottenberg at gmail.com (Duvid Rottenberg)
Date: Wed, 24 Aug 2011 20:41:58 -0400
Subject: [Freeswitch-users] Parsing Error on conference_set_auto_outcall
Message-ID: <CAC4JBEAfoomQ2vq121NCqA_E3Zj1_gnn9T4hyQVjy4st64pqvg@mail.gmail.com>

I was using the Mad Boss example and I was getting parse errors, I
traced it down to the following line,

<action application="conference_set_auto_outcall" data="${group_call(sales)}"/>

when group_call gets expanded it adds the default dial string for each
member of the group which ends up looking something like this.

[sip_invite_domain=10.195.94.25,presence_id=1001 at xxx.xxx.xxx.xxx]sofia/internal/sip:1001 at xxx.xxx.xxx.xxx,
[sip_invite_domain=10.195.94.25]error/user_not_registered

The parse error I got "Cannot create outgoing channel of type
[presence_id=1003 at 10.195.94.25]error cause: [CHAN_NOT_IMPLEMENTED]"
indicated that it was treating the comma inside the brackets as a
separator between 2 dial strings. I worked around this for now by
editing the default config for the directory to remove presence_id
from the dial string and it's working.

This appears to be a bug in the parser as I was getting this error
with the default configurations.

Thanks,
Duvid Rottenberg


From mkopacki at gmail.com  Thu Aug 25 09:46:32 2011
From: mkopacki at gmail.com (Michal Kopacki)
Date: Thu, 25 Aug 2011 07:46:32 +0200
Subject: [Freeswitch-users] Newbie - External Phone setup question
In-Reply-To: <006401cc62a1$b72a4840$257ed8c0$@accra.ca>
References: <006401cc62a1$b72a4840$257ed8c0$@accra.ca>
Message-ID: <4E55E1B8.1040205@gmail.com>

   Hi,

     check your ACL list. Maybe you have to add IP of this phone directly.

-- 
Regards,
Michal

--------------------------------------------------------------
On 2011-08-24 18:06:38 , Charles Bujold (cjbujold at accra.ca) wrote:
>
> Trying to get a phone located in another office registered to the 
> Freeswitch server.  I created an "external5090.xml" in the sip 
> external folder as per the wiki and then set the phone to connect to 
> 5090.  But I cannot get the phone to register.  When I look at 
> wireshark I see a request from the phone to register and then I see 
>  an error "SIP:403 Forbidden" from the server to the Phone.    I have 
> created the extension in the Directory folder with the other internal 
> extensions but I'm wondering if I'm missing something in the 
> 1010extension.xml for it to work. (maybe the extension is located in 
> the wrong folder, or some identification to say it is not in the local 
> LAN?)
>
> I have checked the name and password and they are correct which seems 
> to indicate the issue is at the server level.
>
> Any guidance on how to setup an external phone to Freeswitch would be 
> appreciated.
>
> Thanks
>
> CJB
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110825/791a7d83/attachment.html 

From a.afzali2003 at gmail.com  Thu Aug 25 11:03:46 2011
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Thu, 25 Aug 2011 11:33:46 +0430
Subject: [Freeswitch-users] Flex Client Receives Periodic Interrupted
	Audio
In-Reply-To: <CABK0fC4fSktwbE5yUU6D+abC0VHJGbs8kv1j346qzrGQwAJyOg@mail.gmail.com>
References: <CABK0fC4fSktwbE5yUU6D+abC0VHJGbs8kv1j346qzrGQwAJyOg@mail.gmail.com>
Message-ID: <CABK0fC5vhHOQjeDmJb2XTRx149XKcdn7wttdpYj0RRSAQ1dFwA@mail.gmail.com>

I think have found the problem! The issue relates to denying use of
microphone by adobe flash !!!

-- afshin

On Wed, Aug 24, 2011 at 10:15 PM, afshin afzali <a.afzali2003 at gmail.com> wrote:
> Hi Guys,
>
> As a taste of RTMP end point, I'm experiencing periodic interrupted
> audio. Everything is as defaults in client / server sides except
> that "auth-calls = false" and client does not login / logout, It just
> makes call. I don't know if it can justified by buffer-len / chunksize
> in server side or NetStream.bufferTime in client side.
>
> appreciate all comment.
> -- afshin
>


From gabe at gundy.org  Thu Aug 25 12:06:30 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Thu, 25 Aug 2011 02:06:30 -0600
Subject: [Freeswitch-users] ESL Inbound - what would be the best known
 way to wait in XML dialplan for any command execution via ESL?
In-Reply-To: <BANLkTi=CXOqpYtNDQYwHpz1Ds9nPSAP_ww@mail.gmail.com>
References: <BANLkTikt2R7Qgtm6LL2jr8CP5X4_0_=aeA@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59DEF56437@cooper>
	<BANLkTi=CXOqpYtNDQYwHpz1Ds9nPSAP_ww@mail.gmail.com>
Message-ID: <CAMwZ-t+L0iwpYpcWi3vkGMM2yk-=V9NJGLx76Dk_RxqC41QYzw@mail.gmail.com>

On Sat, Jun 11, 2011 at 10:07 AM, Anton VG <anton.vazir at gmail.com> wrote:
> PS. Outbound ESL showed itself as a terribly bad scaling solution if
> implemented in python...

What?

I don't mean to troll here, but what does the language (in this case,
Python) have to do with scaling Oubound ESL? Any long-running process
is going to be able to handle the setup and teardown of TCP
connections without much trouble.

One could foul up ESL (in or out) in any language, and maybe you've
see it done poorly in Python, but I'd have to hear/see more before
accepting that it's a particularly bad choice.

I'd love to understand better where you are coming from, if you'd care to share.

Best,
Gabe


From danb.lists at googlemail.com  Thu Aug 25 12:14:07 2011
From: danb.lists at googlemail.com (Dan-Cristian Bogos)
Date: Thu, 25 Aug 2011 11:14:07 +0300
Subject: [Freeswitch-users] H264 packetization with echo app
Message-ID: <CADMA_LByHXMXsuy7vvDT5oYUMjwtbUY15z3AjM=4SKigSFtDSw@mail.gmail.com>

Hey Guys,

I have recently started playing with the echo application for video
calls and found some possible issue (asking here if this is the
desired behavior).

Scenario:
 * Call gets connected with echo application, echoing audio just fine.

 * On reinvite the client sends the following SDP(please note the line
with a=fmtp:102 packetization-mode=1):
{{{
v=0.
o=dan.test 0 1 IN IP4 10.10.10.106.
s=-.
c=IN IP4 10.10.10.106.
t=0 0.
m=audio 48576 RTP/AVP 9 96 97 0 8 98 100 5 6 15 101.
a=rtpmap:9 G722/8000.
a=rtpmap:96 speex/32000.
a=rtpmap:97 speex/16000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:98 iLBC/8000.
a=rtpmap:100 speex/8000.
a=rtpmap:5 DVI4/8000.
a=rtpmap:6 DVI4/16000.
a=rtpmap:15 G728/8000.
a=rtpmap:101 telephone-event/8000.
a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level.
m=video 55774 RTP/AVP 102 99.
a=rtpmap:102 H264/90000.
a=fmtp:102 packetization-mode=1.
a=imageattr:102 send [x=[0-640],y=[0-480]] recv *.
a=rtpmap:99 H264/90000.
a=imageattr:99 send [x=[0-640],y=[0-480]] recv *.
a=nortpproxy:yes.
}}}

 * FreeSWITCH accepts it with 200 OK, returning the following SDP:
{{{
v=0.
o=Webi 1314226434 1314226435 IN IP4 10.10.10.106.
s=Webi.
c=IN IP4 127.0.0.1.
t=0 0.
m=audio 32256 RTP/AVP 9 101.
a=rtpmap:9 G722/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
m=video 18750 RTP/AVP 102.
a=rtpmap:102 H264/90000.
}}}

Issue as seen by me:
FreeSWITCH remaps the 102 from an offer with packetization=1 to a
reply with packetization information missing, therefore defaulting to
0 (default out of RFC).

Question:
* Is this intentional or just some wrong negotiation of packetization
parameter? Is FreeSWITCH not supporting packetization=1?

Based on the negotiated stream the client should send now 102 with
packetization=0 (as preferred by FreeSWITCH in signaling) and
FreeSWITCH should return to the client 99 since 102 with
packetization=1 does not look like being supported.

My FreeSWITCH version is the latest git compiled yesterday:
{{{
/opt/freeswitch/bin/freeswitch -version
FreeSWITCH version: 1.0.head (git-8f15bc7 2011-08-23 14-40-44 -0500)
}}}

Thx in advance for any tip!

DanB


From erwin at merioles.net  Thu Aug 25 13:14:41 2011
From: erwin at merioles.net (Erwin Merioles)
Date: Thu, 25 Aug 2011 17:14:41 +0800
Subject: [Freeswitch-users] FreeSwitch Utilizes 100% of CPU Sometimes
	When User Quits Conference
In-Reply-To: <CAKbxfG8SAzOYxvi==cCEcxzq5a0QUiYV3MztxizMC71y3RbtLQ@mail.gmail.com>
References: <002601cc6259$b8b8ec40$2a2ac4c0$@Merioles.net>
	<CAKbxfG8SAzOYxvi==cCEcxzq5a0QUiYV3MztxizMC71y3RbtLQ@mail.gmail.com>
Message-ID: <007701cc6307$6e569270$4b03b750$@Merioles.net>

Thanks for the reply!

I think I've found the problem. When our "host" ends the meeting, it sends a
disconnect event to all participants. The SWITCH_POLLOUT event happens when
the participants tries to end the meeting simultaneously with the host.

I think it is the server's way of doing a "cleanup" and tries to send any
remaining data to whoever is left in the conference -- which in our case is
none. This results to an infinite loop of tries.

The problem was fixed on our end by delaying the disconnection of the
participants a bit.

Regards,

Erwin D. Merioles

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Thursday, August 25, 2011 12:01 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] FreeSwitch Utilizes 100% of CPU Sometimes
When User Quits Conference

Really you should be reporting bugs to http://jira.freeswitch.org Are you
only having this problem with mod_rtmp (its only 2 months old) Most likely
you have created a new condition that the author has not taken into account.
Ideally you should file it on jira under mod_rtmp and attach a back trace
from a core dump produced by gcore.



On Wed, Aug 24, 2011 at 7:31 AM, Erwin Merioles <erwin at merioles.net> wrote:
> Hey guys,
>
>
>
> I?ve been having trouble with FreeSwitch for quite some time. We?re 
> trying to use mod_rtmp to add sound to one of our applications,
www.321meet.com.
> Unfortunately, FreeSwitch?s CPU usage spikes when the host ( the first 
> one to join the conference ), quits or closes the browser window. I?ve 
> checked and this always happen when the following line is called ?
>
>
>
> 2011-08-24 12:24:28.057200 [NOTICE] rtmp_tcp.c:73 Pollout: true
>
>
>
> FS Console Log follows :
>
>
>
> 2011-08-24 12:22:59.177294 [NOTICE] mod_rtmp.c:743 New RTMP session 
> [4a97320c-e50d-4ed3-a59c-4aef799d379d]
>
> 2011-08-24 12:22:59.477278 [NOTICE] rtmp_sig.c:121 Sent connect reply
>
> 2011-08-24 12:23:15.597196 [INFO] rtmp_sig.c:136 Replied to 
> createStream (1)
>
> 2011-08-24 12:23:16.237201 [INFO] rtmp_sig.c:274 Got publish on stream 1.
>
> 2011-08-24 12:23:17.177293 [INFO] rtmp_sig.c:136 Replied to 
> createStream (2)
>
> 2011-08-24 12:23:17.177293 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled 
> control packet (type=0x3)
>
> 2011-08-24 12:23:17.457195 [WARNING] sofia.c:4403 Ping succeeded 
> voip9.telsome.com with code 404 - count -1/1/1, state UP
>
> 2011-08-24 12:23:17.457195 [INFO] rtmp_sig.c:136 Replied to 
> createStream (3)
>
> 2011-08-24 12:23:17.457195 [NOTICE] switch_channel.c:897 New Channel
> rtmp/default/3213533 [8e7ccce8-172c-4b91-9183-1ad9d2f0e6dd]
>
> 2011-08-24 12:23:17.457195 [ERR] rtmp_sig.c:305 Couldn't create call.
>
> 2011-08-24 12:23:17.457195 [WARNING] sofia.c:4403 Ping succeeded 
> testin with code 404 - count -1/1/1, state UP
>
> 2011-08-24 12:23:17.497199 [INFO] mod_dialplan_xml.c:336 Processing
> <0000000000>->3213533 in context default
>
> 2011-08-24 12:23:17.497199 [NOTICE] mod_rtmp.c:497 Channel 
> [rtmp/default/3213533] has been answered
>
> 2011-08-24 12:23:17.497199 [INFO] mod_conference.c:6644 using channel 
> sound
> prefix: /usr/local/freeswitch/sounds/en/us/callie
>
> 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 
> rtmp/default/3213533 binding '0' to 'mute'
>
> 2011-08-24 12:23:17.597289 [INFO] switch_ivr_async.c:164 Digit parser
> mod_conference: Setting realm to conf
>
> 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 
> rtmp/default/3213533 binding '*' to 'deaf mute'
>
> 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 
> rtmp/default/3213533 binding '9' to 'energy up'
>
> 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 
> rtmp/default/3213533 binding '8' to 'energy equ'
>
> 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 
> rtmp/default/3213533 binding '7' to 'energy dn'
>
> 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 
> rtmp/default/3213533 binding '3' to 'vol talk up'
>
> 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 
> rtmp/default/3213533 binding '2' to 'vol talk zero'
>
> 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 
> rtmp/default/3213533 binding '1' to 'vol talk dn'
>
> 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 
> rtmp/default/3213533 binding '6' to 'vol listen up'
>
> 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 
> rtmp/default/3213533 binding '5' to 'vol listen zero'
>
> 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 
> rtmp/default/3213533 binding '4' to 'vol listen dn'
>
> 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 
> rtmp/default/3213533 binding '#' to 'hangup'
>
> 2011-08-24 12:23:17.797195 [INFO] rtmp_sig.c:159 Sending audio
>
> 2011-08-24 12:23:17.797195 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled 
> control packet (type=0x3)
>
> 2011-08-24 12:23:18.117290 [INFO] rtmp_sig.c:274 Got publish on stream 3.
>
> 2011-08-24 12:23:36.697195 [ERR] rtmp.c:678 Read error
>
> 2011-08-24 12:23:36.697195 [NOTICE] mod_rtmp.c:788 RTMP session ended 
> [4a97320c-e50d-4ed3-a59c-4aef799d379d]
>
> 2011-08-24 12:23:36.697195 [NOTICE] mod_rtmp.c:803 Hangup
> rtmp/default/3213533 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]
>
> 2011-08-24 12:23:36.777294 [NOTICE] switch_core_session.c:1347 Session 
> 1
> (rtmp/default/3213533) Ended
>
> 2011-08-24 12:23:36.777294 [NOTICE] switch_core_session.c:1349 Close 
> Channel
> rtmp/default/3213533 [CS_DESTROY]
>
> 2011-08-24 12:23:48.237202 [NOTICE] mod_rtmp.c:743 New RTMP session 
> [3ff4786f-7cae-4653-879a-ae95a9d50742]
>
> 2011-08-24 12:23:48.557204 [NOTICE] rtmp_sig.c:121 Sent connect reply
>
> 2011-08-24 12:23:55.357293 [NOTICE] mod_rtmp.c:743 New RTMP session 
> [af763930-9054-4c76-a0eb-9c351e75949d]
>
> 2011-08-24 12:23:55.757290 [NOTICE] rtmp_sig.c:121 Sent connect reply
>
> 2011-08-24 12:23:58.957195 [INFO] rtmp_sig.c:136 Replied to 
> createStream (1)
>
> 2011-08-24 12:23:59.517204 [INFO] rtmp_sig.c:274 Got publish on stream 1.
>
> 2011-08-24 12:24:00.557201 [INFO] rtmp_sig.c:136 Replied to 
> createStream (2)
>
> 2011-08-24 12:24:00.557201 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled 
> control packet (type=0x3)
>
> 2011-08-24 12:24:00.557201 [INFO] rtmp_sig.c:136 Replied to 
> createStream (3)
>
> 2011-08-24 12:24:00.857222 [NOTICE] switch_channel.c:897 New Channel
> rtmp/default/3213533 [5de55413-5cb9-4c18-8b65-9c438833449f]
>
> 2011-08-24 12:24:00.857222 [ERR] rtmp_sig.c:305 Couldn't create call.
>
> 2011-08-24 12:24:00.857222 [INFO] mod_dialplan_xml.c:336 Processing
> <0000000000>->3213533 in context default
>
> 2011-08-24 12:24:00.857222 [NOTICE] mod_rtmp.c:497 Channel 
> [rtmp/default/3213533] has been answered
>
> 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:6644 using channel 
> sound
> prefix: /usr/local/freeswitch/sounds/en/us/callie
>
> 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 
> rtmp/default/3213533 binding '0' to 'mute'
>
> 2011-08-24 12:24:00.857222 [INFO] switch_ivr_async.c:164 Digit parser
> mod_conference: Setting realm to conf
>
> 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 
> rtmp/default/3213533 binding '*' to 'deaf mute'
>
> 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 
> rtmp/default/3213533 binding '9' to 'energy up'
>
> 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 
> rtmp/default/3213533 binding '8' to 'energy equ'
>
> 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 
> rtmp/default/3213533 binding '7' to 'energy dn'
>
> 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 
> rtmp/default/3213533 binding '3' to 'vol talk up'
>
> 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 
> rtmp/default/3213533 binding '2' to 'vol talk zero'
>
> 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 
> rtmp/default/3213533 binding '1' to 'vol talk dn'
>
> 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 
> rtmp/default/3213533 binding '6' to 'vol listen up'
>
> 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 
> rtmp/default/3213533 binding '5' to 'vol listen zero'
>
> 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 
> rtmp/default/3213533 binding '4' to 'vol listen dn'
>
> 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 
> rtmp/default/3213533 binding '#' to 'hangup'
>
> 2011-08-24 12:24:01.197247 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled 
> control packet (type=0x3)
>
> 2011-08-24 12:24:01.197247 [INFO] rtmp_sig.c:274 Got publish on stream 3.
>
> 2011-08-24 12:24:01.197247 [INFO] rtmp_sig.c:159 Sending audio
>
> 2011-08-24 12:24:13.637291 [INFO] rtmp_sig.c:136 Replied to 
> createStream (1)
>
> 2011-08-24 12:24:14.157289 [INFO] rtmp_sig.c:274 Got publish on stream 1.
>
> 2011-08-24 12:24:17.057195 [INFO] rtmp_sig.c:136 Replied to 
> createStream (2)
>
> 2011-08-24 12:24:17.057195 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled 
> control packet (type=0x3)
>
> 2011-08-24 12:24:17.057195 [INFO] rtmp_sig.c:136 Replied to 
> createStream (3)
>
> 2011-08-24 12:24:17.057195 [NOTICE] switch_channel.c:897 New Channel
> rtmp/default/3213533 [4e6c35ce-ec56-4ef1-816a-b94704efbfbd]
>
> 2011-08-24 12:24:17.057195 [ERR] rtmp_sig.c:305 Couldn't create call.
>
> 2011-08-24 12:24:17.057195 [INFO] mod_dialplan_xml.c:336 Processing
> <0000000000>->3213533 in context default
>
> 2011-08-24 12:24:17.057195 [NOTICE] mod_rtmp.c:497 Channel 
> [rtmp/default/3213533] has been answered
>
> 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 
> rtmp/default/3213533 binding '0' to 'mute'
>
> 2011-08-24 12:24:17.057195 [INFO] switch_ivr_async.c:164 Digit parser
> mod_conference: Setting realm to conf
>
> 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 
> rtmp/default/3213533 binding '*' to 'deaf mute'
>
> 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 
> rtmp/default/3213533 binding '9' to 'energy up'
>
> 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 
> rtmp/default/3213533 binding '8' to 'energy equ'
>
> 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 
> rtmp/default/3213533 binding '7' to 'energy dn'
>
> 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 
> rtmp/default/3213533 binding '3' to 'vol talk up'
>
> 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 
> rtmp/default/3213533 binding '2' to 'vol talk zero'
>
> 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 
> rtmp/default/3213533 binding '1' to 'vol talk dn'
>
> 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 
> rtmp/default/3213533 binding '6' to 'vol listen up'
>
> 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 
> rtmp/default/3213533 binding '5' to 'vol listen zero'
>
> 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 
> rtmp/default/3213533 binding '4' to 'vol listen dn'
>
> 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 
> rtmp/default/3213533 binding '#' to 'hangup'
>
> 2011-08-24 12:24:17.497196 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled 
> control packet (type=0x3)
>
> 2011-08-24 12:24:17.497196 [INFO] rtmp_sig.c:274 Got publish on stream 3.
>
> 2011-08-24 12:24:17.497196 [INFO] rtmp_sig.c:159 Sending audio
>
> 2011-08-24 12:24:27.657196 [ERR] rtmp.c:678 Read error
>
> 2011-08-24 12:24:27.657196 [NOTICE] mod_rtmp.c:788 RTMP session ended 
> [3ff4786f-7cae-4653-879a-ae95a9d50742]
>
> 2011-08-24 12:24:27.657196 [NOTICE] mod_rtmp.c:803 Hangup
> rtmp/default/3213533 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]
>
> 2011-08-24 12:24:27.677201 [NOTICE] switch_core_session.c:1347 Session 
> 2
> (rtmp/default/3213533) Ended
>
> 2011-08-24 12:24:27.677201 [NOTICE] switch_core_session.c:1349 Close 
> Channel
> rtmp/default/3213533 [CS_DESTROY]
>
> 2011-08-24 12:24:28.057200 [NOTICE] rtmp_tcp.c:73 Pollout: true
>
>
>
> Which results to :
>
>
>
> ? PID USER????? PR? NI? VIRT? RES? SHR S %CPU %MEM??? TIME+? COMMAND
>
> 1608 root????? -2 -10? 191m? 20m 6588 R 94.3? 3.5?? 5:39.50 fs
>
>
>
> Any ideas? Help is VERY much appreciated. TIA!
>
>
>
> Regards,
>
>
>
> Erwin D. Merioles
>
>
>
> merioles.net
>
> +63 922 837 9466 | +63 917 501 1010 | +1 760 670 3241
>
> aY!M : erwin_merioles | Skype : erwin.merioles
>
> This message (including any attachments) contains information that may 
> be confidential. Unless you are the intended recipient (or is 
> authorized to receive for the intended recipient), you may not read, 
> print, retain, use, copy, distribute, or disclose to anyone, any 
> information contained here. If you have received this in error, please 
> advise the sender by reply e-mail, and delete all copies of the original
message (including attachments).
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org
>
>



--
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


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



From alex at jajah.com  Thu Aug 25 13:25:34 2011
From: alex at jajah.com (Alex Massover)
Date: Thu, 25 Aug 2011 12:25:34 +0300
Subject: [Freeswitch-users] Forcing CN offer
Message-ID: <F5B5EF8F78C0BC4BB417E1DBB9D7F7D0180A64A996@jjex01.jajah.dublin>

Hello,

I'm working on the following scenarios:
scenario 1:  A  --> FS --> B
scenario 2: C --> FS --> A

Where A supports CN (a=rtpmap:13 CN/8000) and requires CN negotiation, and B and C do not support and aren't able to negotiate it.

I successfully implemented scenario 1, CN is negotiated between A party and FS, and not negotiated between FS and B.

But scenario 2 doesn't work for me! When C doesn't offer CN in INVITE towards FS, FS also doesn't offer CN in SDP in INVITE towards A.
And nothing helps, tried all the combination of VAD options. Gateway A is in the same internal sip profile in both scenarios.

This is the SDP in INVITE from C to FS:

            Media Description, name and address (m): audio 30224 RTP/AVP 0 8 18 101
            Media Attribute (a): rtpmap:0 PCMU/8000
            Media Attribute (a): rtpmap:8 PCMA/8000
            Media Attribute (a): rtpmap:18 G729/8000
            Media Attribute (a): fmtp:18 annexb=no
            Media Attribute (a): rtpmap:101 telephone-event/8000
            Media Attribute (a): fmtp:101 0-16
            Media Attribute (a): silenceSupp:off - - - -
            Media Attribute (a): ptime:20
            Media Attribute (a): sendrecv


And this is the SDP from FS towards A:
            Media Description, name and address (m): audio 23564 RTP/AVP 0 8 101 13
            Media Attribute (a): rtpmap:101 telephone-event/8000
            Media Attribute (a): fmtp:101 0-16
            Media Attribute (a): ptime:20

FS even puts 13 in m= attribute, but doesn't add a=  attribute for rtpmap:13.


Is that a bug?


--
Best Regards
Alex Massover.

This mail was sent via Mail-SeCure System.


-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110825/f14b3857/attachment.html 

From alex at jajah.com  Thu Aug 25 15:12:51 2011
From: alex at jajah.com (Alex Massover)
Date: Thu, 25 Aug 2011 14:12:51 +0300
Subject: [Freeswitch-users] Forcing CN offer
Message-ID: <F5B5EF8F78C0BC4BB417E1DBB9D7F7D0180A64AA18@jjex01.jajah.dublin>

Hi,

Sorry, the RFC says that a= is not mandatory, m= is enough.
I was just confused why FS in some scenario puts it and in other doesn?t, despite using only G711 at 8000.


From: Alex Massover
Sent: ??? ? 25 ?????? 2011 12:26
To: 'freeswitch-users at lists.freeswitch.org'
Subject: Forcing CN offer

Hello,

I?m working on the following scenarios:
scenario 1:  A  --> FS --> B
scenario 2: C --> FS --> A

Where A supports CN (a=rtpmap:13 CN/8000) and requires CN negotiation, and B and C do not support and aren?t able to negotiate it.

I successfully implemented scenario 1, CN is negotiated between A party and FS, and not negotiated between FS and B.

But scenario 2 doesn?t work for me! When C doesn?t offer CN in INVITE towards FS, FS also doesn?t offer CN in SDP in INVITE towards A.
And nothing helps, tried all the combination of VAD options. Gateway A is in the same internal sip profile in both scenarios.

This is the SDP in INVITE from C to FS:

            Media Description, name and address (m): audio 30224 RTP/AVP 0 8 18 101
            Media Attribute (a): rtpmap:0 PCMU/8000
            Media Attribute (a): rtpmap:8 PCMA/8000
            Media Attribute (a): rtpmap:18 G729/8000
            Media Attribute (a): fmtp:18 annexb=no
            Media Attribute (a): rtpmap:101 telephone-event/8000
            Media Attribute (a): fmtp:101 0-16
            Media Attribute (a): silenceSupp:off - - - -
            Media Attribute (a): ptime:20
            Media Attribute (a): sendrecv


And this is the SDP from FS towards A:
            Media Description, name and address (m): audio 23564 RTP/AVP 0 8 101 13
            Media Attribute (a): rtpmap:101 telephone-event/8000
            Media Attribute (a): fmtp:101 0-16
            Media Attribute (a): ptime:20

FS even puts 13 in m= attribute, but doesn?t add a=  attribute for rtpmap:13.


Is that a bug?


--
Best Regards
Alex Massover.

This mail was sent via Mail-SeCure System.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110825/07f1beaa/attachment.html 

From infos at madovsky.org  Thu Aug 25 18:38:42 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 25 Aug 2011 10:38:42 -0400
Subject: [Freeswitch-users] ringing status and mod_rtmp
Message-ID: <909E89D1B3434E64B75BFC53726CE0A9@e1705>

Hi folks,

is there any callback event for ringing status of leg B with mod_rtmp ?

Thanks

Franck
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110825/c0f0ef82/attachment.html 

From infos at madovsky.org  Thu Aug 25 18:58:39 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 25 Aug 2011 10:58:39 -0400
Subject: [Freeswitch-users] about fax status
Message-ID: <01EA2FD4A1674DC0B3FE1C91C095E989@e1705>

Sometimes fax fails with status like

Received a DCN while waiting for a DIS

what does it mean ?

Thanks

Franck
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110825/9e709c84/attachment.html 

From steveu at coppice.org  Thu Aug 25 20:02:46 2011
From: steveu at coppice.org (Steve Underwood)
Date: Fri, 26 Aug 2011 00:02:46 +0800
Subject: [Freeswitch-users] about fax status
In-Reply-To: <01EA2FD4A1674DC0B3FE1C91C095E989@e1705>
References: <01EA2FD4A1674DC0B3FE1C91C095E989@e1705>
Message-ID: <4E567226.1090109@coppice.org>

On 08/25/2011 10:58 PM, Madovsky wrote:
> Sometimes fax fails with status like
> *Received a DCN while waiting for a DIS*
> what does it mean ?
>
A DCN is a disconnect message. A DIS is what the called party normally 
sends as the first message in a call.

Steve



From anthony.minessale at gmail.com  Thu Aug 25 20:26:25 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 25 Aug 2011 11:26:25 -0500
Subject: [Freeswitch-users] H264 packetization with echo app
In-Reply-To: <CADMA_LByHXMXsuy7vvDT5oYUMjwtbUY15z3AjM=4SKigSFtDSw@mail.gmail.com>
References: <CADMA_LByHXMXsuy7vvDT5oYUMjwtbUY15z3AjM=4SKigSFtDSw@mail.gmail.com>
Message-ID: <CAKbxfG9QGdadbQ5cQFA1vZL85+8VmNYDS8=Ewbq2yrKXy9AtUA@mail.gmail.com>

update to this revision and see if it works:

commit e644e620c89867895cb4e69f834913f5df1070e2
Author: Anthony Minessale <anthm at freeswitch.org>
Date:   Thu Aug 25 09:24:17 2011 -0500

    reflect video fmtp on 1 legged calls



On Thu, Aug 25, 2011 at 3:14 AM, Dan-Cristian Bogos
<danb.lists at googlemail.com> wrote:
> Hey Guys,
>
> I have recently started playing with the echo application for video
> calls and found some possible issue (asking here if this is the
> desired behavior).
>
> Scenario:
> ?* Call gets connected with echo application, echoing audio just fine.
>
> ?* On reinvite the client sends the following SDP(please note the line
> with a=fmtp:102 packetization-mode=1):
> {{{
> v=0.
> o=dan.test 0 1 IN IP4 10.10.10.106.
> s=-.
> c=IN IP4 10.10.10.106.
> t=0 0.
> m=audio 48576 RTP/AVP 9 96 97 0 8 98 100 5 6 15 101.
> a=rtpmap:9 G722/8000.
> a=rtpmap:96 speex/32000.
> a=rtpmap:97 speex/16000.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:98 iLBC/8000.
> a=rtpmap:100 speex/8000.
> a=rtpmap:5 DVI4/8000.
> a=rtpmap:6 DVI4/16000.
> a=rtpmap:15 G728/8000.
> a=rtpmap:101 telephone-event/8000.
> a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level.
> m=video 55774 RTP/AVP 102 99.
> a=rtpmap:102 H264/90000.
> a=fmtp:102 packetization-mode=1.
> a=imageattr:102 send [x=[0-640],y=[0-480]] recv *.
> a=rtpmap:99 H264/90000.
> a=imageattr:99 send [x=[0-640],y=[0-480]] recv *.
> a=nortpproxy:yes.
> }}}
>
> ?* FreeSWITCH accepts it with 200 OK, returning the following SDP:
> {{{
> v=0.
> o=Webi 1314226434 1314226435 IN IP4 10.10.10.106.
> s=Webi.
> c=IN IP4 127.0.0.1.
> t=0 0.
> m=audio 32256 RTP/AVP 9 101.
> a=rtpmap:9 G722/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=silenceSupp:off - - - -.
> a=ptime:20.
> m=video 18750 RTP/AVP 102.
> a=rtpmap:102 H264/90000.
> }}}
>
> Issue as seen by me:
> FreeSWITCH remaps the 102 from an offer with packetization=1 to a
> reply with packetization information missing, therefore defaulting to
> 0 (default out of RFC).
>
> Question:
> * Is this intentional or just some wrong negotiation of packetization
> parameter? Is FreeSWITCH not supporting packetization=1?
>
> Based on the negotiated stream the client should send now 102 with
> packetization=0 (as preferred by FreeSWITCH in signaling) and
> FreeSWITCH should return to the client 99 since 102 with
> packetization=1 does not look like being supported.
>
> My FreeSWITCH version is the latest git compiled yesterday:
> {{{
> /opt/freeswitch/bin/freeswitch -version
> FreeSWITCH version: 1.0.head (git-8f15bc7 2011-08-23 14-40-44 -0500)
> }}}
>
> Thx in advance for any tip!
>
> DanB
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From anthony.minessale at gmail.com  Thu Aug 25 20:31:16 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 25 Aug 2011 11:31:16 -0500
Subject: [Freeswitch-users] Forcing CN offer
In-Reply-To: <F5B5EF8F78C0BC4BB417E1DBB9D7F7D0180A64A996@jjex01.jajah.dublin>
References: <F5B5EF8F78C0BC4BB417E1DBB9D7F7D0180A64A996@jjex01.jajah.dublin>
Message-ID: <CAKbxfG-gO_=ey7qS_rOw_VrTtrVcAHvv+Cp=BrQZpeF_-5KQNA@mail.gmail.com>

if you set the var verbose_sdp=true globally in vars.xml or on a
per_leg basis you will get the extra a= that some things mistakenly
think are mandatory (like some older polycom firmwares) the goal is to
make the sdp as small as possible to prevent going over the MTU

On Thu, Aug 25, 2011 at 4:25 AM, Alex Massover <alex at jajah.com> wrote:
> Hello,
>
>
>
> I?m working on the following scenarios:
>
> scenario 1:? A? ? FS ? B
>
> scenario 2: C ? FS ? A
>
>
>
> Where A supports CN (a=rtpmap:13 CN/8000) and requires CN negotiation, and B
> and C do not support and aren?t able to negotiate it.
>
>
>
> I successfully implemented scenario 1, CN is negotiated between A party and
> FS, and not negotiated between FS and B.
>
>
>
> But scenario 2 doesn?t work for me! When C doesn?t offer CN in INVITE
> towards FS, FS also doesn?t offer CN in SDP in INVITE towards A.
>
> And nothing helps, tried all the combination of VAD options. Gateway A is in
> the same internal sip profile in both scenarios.
>
>
>
> This is the SDP in INVITE from C to FS:
>
>
>
> ??????????? Media Description, name and address (m): audio 30224 RTP/AVP 0 8
> 18 101
>
> ??????????? Media Attribute (a): rtpmap:0 PCMU/8000
>
> ??????????? Media Attribute (a): rtpmap:8 PCMA/8000
>
> ??????????? Media Attribute (a): rtpmap:18 G729/8000
>
> ??????????? Media Attribute (a): fmtp:18 annexb=no
>
> ??????????? Media Attribute (a): rtpmap:101 telephone-event/8000
>
> ??????????? Media Attribute (a): fmtp:101 0-16
>
> ??????????? Media Attribute (a): silenceSupp:off - - - -
>
> ??????????? Media Attribute (a): ptime:20
>
> ??????????? Media Attribute (a): sendrecv
>
>
>
>
>
> And this is the SDP from FS towards A:
>
> ??????????? Media Description, name and address (m): audio 23564 RTP/AVP 0 8
> 101 13
>
> ??????????? Media Attribute (a): rtpmap:101 telephone-event/8000
>
> ??????????? Media Attribute (a): fmtp:101 0-16
>
> ??????????? Media Attribute (a): ptime:20
>
>
>
> FS even puts 13 in m= attribute, but doesn?t add a=? attribute for
> rtpmap:13.
>
>
>
>
>
> Is that a bug?
>
>
>
>
>
> --
>
> Best Regards
>
> Alex Massover.
>
> This mail was sent via Mail-SeCure System.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From adam.kelloway at newpace.ca  Thu Aug 25 22:55:34 2011
From: adam.kelloway at newpace.ca (Adam Kelloway)
Date: Thu, 25 Aug 2011 15:55:34 -0300
Subject: [Freeswitch-users] Session ends unexpectedly during record dial
	plan usage
Message-ID: <4E569AA6.2010008@newpace.ca>

Hi there,

I have a freeswitch installation that I can make sip calls to to listen 
to IVR menus. The sessions last as long as either side does not hang up. 
The exception to this is when I use the 'record' dial plan tool. The sip 
session ends unexpectedly after about 32+ seconds into the recording. 
This happens every time I use the record tool. Note that I have set the 
maximum message length to 120 seconds, so this shouldn't be coming into 
play here (and shouldn't affect the session anyway).

Has anyone ever experienced this, and do you have any suggestions as to 
what might be the cause?

Note that there is no NAT involved here. There are also no Expires or 
Session-Expires header(s) in the sip INVITE or response that would 
affect the length of the session. Indeed, the same type of session can 
continue indefinitely until about 32+ seconds after I invoke the record 
dial plan tool.

Thanks,

Adam



From cjbujold at accra.ca  Fri Aug 26 00:46:21 2011
From: cjbujold at accra.ca (Charles Bujold)
Date: Thu, 25 Aug 2011 17:46:21 -0300
Subject: [Freeswitch-users] Strange problem with Freeswitch and external
	extension
Message-ID: <000901cc6368$0c402220$24c06660$@accra.ca>

I have a remote extension which registers with the Freeswitch server.  The
user can record his greeting in the Freeswitch IVR without a problem.  Yet
if the user calls me I cannot hear him, but he can hear me.

 

I thought at first that it was a NAT issue so we placed the remote telephone
directly on the internet and the same problem remains.  The Freeswitch
server is behind a NAT/router and I can see the packets from the phone
coming into the router and being forwarded to the Freeswitch server.  The
router is not blocking the packets and is forwarding them to the correct
server IP.

 

I them placed wireshark on the Freeswitch server and I can see the packets
leave the Freeswitch server and going to the phone But I cannot see the
Freeswitch server receive the packets from the PFsense router, even though I
can see in the router that the packets are being forwarded properly to the
Freeswitch server IP.  The Freeswitch server is running Ubuntu with no
firewall, so in theory I should see the server accept or reject the packets.
Yet I do not see any in wireshark coming from the phone, only those leaving
from the Freeswitch server.

 

What I do not understand is why, since he is registered,  I cannot hear him,
but yet he can interact with Freeswitch and record a greeting?  Obviously
packets are getting routed properly and Freeswitch recognizes them, some of
the time.

 

Can somebody suggest a possible explanation and solution to this problem?

 

Thanks

 

cjb

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110825/865bead4/attachment.html 

From beffa at ieee.org  Fri Aug 26 02:24:57 2011
From: beffa at ieee.org (Federico Beffa)
Date: Fri, 26 Aug 2011 00:24:57 +0200
Subject: [Freeswitch-users] call_timeout
Message-ID: <CAKrPhPMB9oDhie1ncB0pgCVw4-3o+uVjULPvX26ibLXsDkajPQ@mail.gmail.com>

Hi,

I've setup the dialplan such that when I receive a call from a provider
(public context), the call get transferred to an extension in the default
context (in a similar way as the default config). However, no matter what I
set for call_timeout, the call is always terminated after ca. 19 seconds
with a cause ORIGINATOR_CANCEL. I would like to extend the timeout period.

This is my public entry for the incoming call:

<include>
  <extension name="incoming_call">
    <condition field="destination_number" expression="^XXXXX$">
      <action application="set" data="domain_name=$${domain}"/>
      <action application="export" data="call_timeout=60"/>
      <action application="export" data="bridge_answer_timeout=60"/>
      <action application="export" data="originate_timeout=60"/>
      <action application="ring_ready"/>
      <action application="bridge" data="user/1000@${domain_name}"/>
      <!--<action application="transfer" data="1000 XML default"/>-->
      <action application="hangup"/>
    </condition>
  </extension>
</include>

And this is part of the log:

2011-08-25 23:57:30.920053 [NOTICE] sofia.c:5226 Ring-Ready
sofia/external/0916001220!
2011-08-25 23:57:30.920053 [NOTICE] mod_sofia.c:2366 Ring-Ready
sofia/internal/1001 at xxxxx!
2011-08-25 23:57:30.920053 [NOTICE] switch_ivr_originate.c:481 Ring Ready
sofia/internal/1001 at xxxxx!
2011-08-25 23:57:30.960053 [NOTICE] sofia.c:5226 Ring-Ready sofia/internal/
sip:1000 at 192.168.0.5:31148!
2011-08-25 23:57:30.960053 [NOTICE] switch_ivr_originate.c:481 Ring Ready
sofia/external/0916001220 at 195.190.166.249!
2011-08-25 23:57:50.900063 [NOTICE] sofia.c:5872 Hangup sofia/external/
0916001220 at 195.190.166.249 [CS_EXECUTE] [ORIGINATOR_CANCEL]
2011-08-25 23:57:50.900063 [NOTICE] switch_ivr_originate.c:3144 Hangup
sofia/internal/sip:1000 at 192.168.0.5:31148 [CS_CONSUME_MEDIA]
[ORIGINATOR_CANCEL]
2011-08-25 23:57:50.900063 [NOTICE] switch_ivr_originate.c:2436 Cannot
create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL]
2011-08-25 23:57:50.900063 [INFO] mod_dptools.c:2696 Originate Failed.
Cause: ORIGINATOR_CANCEL
2011-08-25 23:57:50.900063 [NOTICE] sofia.c:5872 Hangup
sofia/external/0916001220 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]
2011-08-25 23:57:50.900063 [NOTICE] switch_core_session.c:1353 Session 11
(sofia/external/0916001220 at 195.190.166.249) Ended
2011-08-25 23:57:50.900063 [NOTICE] switch_core_session.c:1355 Close Channel
sofia/external/0916001220 at 195.190.166.249 [CS_DESTROY]
2011-08-25 23:57:50.900063 [NOTICE] switch_core_session.c:1353 Session 12
(sofia/internal/sip:1000 at 192.168.0.5:31148) Ended
2011-08-25 23:57:50.900063 [NOTICE] switch_core_session.c:1355 Close Channel
sofia/internal/sip:1000 at 192.168.0.5:31148 [CS_DESTROY]
2011-08-25 23:57:50.900063 [NOTICE] switch_core_session.c:1353 Session 10
(sofia/external/0916001220) Ended
2011-08-25 23:57:50.900063 [NOTICE] switch_core_session.c:1355 Close Channel
sofia/external/0916001220 [CS_DESTROY]
2011-08-25 23:57:50.900063 [INFO] mod_dptools.c:2696 Originate Failed.
Cause: NORMAL_TEMPORARY_FAILURE
2011-08-25 23:57:50.900063 [NOTICE] mod_dptools.c:2815 Hangup
sofia/internal/1001 at xxxxx [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]
2011-08-25 23:57:50.900063 [NOTICE] switch_core_session.c:1353 Session 9
(sofia/internal/1001 at xxxxx) Ended
2011-08-25 23:57:50.900063 [NOTICE] switch_core_session.c:1355 Close Channel
sofia/internal/1001 at xxxxx [CS_DESTROY]

I've tryied with "transfer" and with "bridge" also setting [leg_timeout=60],
but no luck. I've changed all timeout related values in default.xml as well.


If the call comes from a phone registered locally, then the timeout works
fine. The problem only exists with calls coming from outside (the public
context). However, the timeout does seems to be related to freeswitch and
not the provider as I have another system running Asterisk and there the
problem (using the same provider accounts) does not exist.

I would really appreciate any suggestion on how to extend calls timeout.

Regards,
Fede
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110826/2259501a/attachment-0001.html 

From danb.lists at googlemail.com  Fri Aug 26 15:05:24 2011
From: danb.lists at googlemail.com (Dan-Cristian Bogos)
Date: Fri, 26 Aug 2011 14:05:24 +0300
Subject: [Freeswitch-users] H264 packetization with echo app
Message-ID: <CADMA_LBrJFGYJBSBi4ABBDZZELadbGmTXhm44PV6Qu8BY-Vtow@mail.gmail.com>

Hey Anthony,

Many thanks for so fast reaction. It works like a charm with your new patches.

Have a good one!
DanB


From pkelly at gmail.com  Fri Aug 26 16:27:06 2011
From: pkelly at gmail.com (Pete Kelly)
Date: Fri, 26 Aug 2011 13:27:06 +0100
Subject: [Freeswitch-users] mod_nibblebill mp3
Message-ID: <CAEWH9M86Js7QesQbJ9B2+AUjF-Y9n0ofNfqmFRqXomrN+=pR9Q@mail.gmail.com>

Hi everyone

I am looking at prepaid solutions at the moment, and mod_nibblebill seems
pretty interesting. However the mp3 linked to from the wiki page is no
longer available:

http://wiki.freeswitch.org/wiki/Mod_nibblebill

Does anyone have a copy at all?

Also, how 'stable' is nibblebill, is it a well used module by people?

I'll probably be starting off with large call volumes (rather than slow
ramping up) so any ideas of usage statistics or advice on usage would be
useful.

Thanks

Pete
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110826/4c7d885d/attachment.html 

From pkelly at gmail.com  Fri Aug 26 16:32:16 2011
From: pkelly at gmail.com (Pete Kelly)
Date: Fri, 26 Aug 2011 13:32:16 +0100
Subject: [Freeswitch-users] Making Freeswitch listen on 2 IP addresses
In-Reply-To: <CAEWH9M91Q9DbDmNj+DqPvfKi1KNLb+ZiGRe5gkV2zOifszhAgg@mail.gmail.com>
References: <CAEWH9M-yKO+7ko_wOsxHTEp-Rp7ZRr3+wuMvU=MpquuCFCQ=nA@mail.gmail.com>
	<2C2837D4-14B7-4702-AFD4-767CDDA8CAC3@seventhsignal.de>
	<CAO+9iGcghRhxkdMAukA44bH+54Q4Y+2SvRPxG2zUxZ9QG_x4FA@mail.gmail.com>
	<CAEWH9M91Q9DbDmNj+DqPvfKi1KNLb+ZiGRe5gkV2zOifszhAgg@mail.gmail.com>
Message-ID: <CAEWH9M_pK1FPmQf4BX2oPuFxM_KpZp6X1aZFmryK7mw-VwT98w@mail.gmail.com>

Worked brilliantly btw, thanks for the replies :)

On 24 August 2011 10:10, Pete Kelly <pkelly at gmail.com> wrote:

> Thanks for the replies, I'm going to give the 2 profiles a go.
>
>
> On 24 August 2011 09:13, Alec Taylor <alec.taylor6 at gmail.com> wrote:
>
>> Why not setup DNS?
>>
>> On Wed, Aug 24, 2011 at 5:52 PM, Michal Bielicki
>> <michal.bielicki at seventhsignal.de> wrote:
>> > Why not use 2 profiles ?
>> > Am 24.08.2011 um 09:30 schrieb Pete Kelly:
>> >
>> > Hi
>> > Is it possible to make freeswitch listen on more than one IP for
>> requests
>> > within the same profile?
>> > I would like to assign a VIP to a pair of freeswitch boxes, for the
>> purposes
>> > of failover if one of the boxes should fail or go down. The VIP can then
>> be
>> > used by web applications which have to send XMLRPC requests to
>> freeswitch.
>> > In addition I would like each of the boxes to listen on their "normal"
>> eth0
>> > IP too, so systems like opensips which support load balancing can evenly
>> > distribute load across the two boxes.
>> > My profile.xml file currently has this entry:
>> > <param name="sip-ip" value="$${local_ip_v4}"/>
>> > So I can't see how freeswitch can be told to listen on more than one IP
>> for
>> > a profile.
>> > Is this possible?
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> > Michal Bielicki
>> > Gesch?ftsf?hrer / CEO
>> > Seventh Signal Ltd. & Co. KG
>> > Weigandufer 45, B?ro 115, D-12059 Berlin
>> > Voice: +49 30 60988730
>> > Amtsgericht Charlottenburg HRA 44413 B
>> > Ust.-ID: DE266981999
>> > Gesch?ftsf?hrer: Michal Bielicki
>> > Pers?nlich Haftende Gesellschafterin:
>> > Seventh Signal Ltd, 69 Great Hampton St. Birmingham,
>> > B18 6EW, GB, Company Nr.: 06889439
>> > WWW.: http://www.seventhsignal.de
>> >
>> > ----
>> >
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110826/0f36de42/attachment.html 

From jeff at jefflenk.com  Fri Aug 26 17:21:28 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Fri, 26 Aug 2011 06:21:28 -0700 (PDT)
Subject: [Freeswitch-users] call_timeout
In-Reply-To: <CAKrPhPMB9oDhie1ncB0pgCVw4-3o+uVjULPvX26ibLXsDkajPQ@mail.gmail.com>
References: <CAKrPhPMB9oDhie1ncB0pgCVw4-3o+uVjULPvX26ibLXsDkajPQ@mail.gmail.com>
Message-ID: <1314364888504-6728720.post@n2.nabble.com>

You should look at the siptrace to see what the other end is doing.

sofia global siptrace on 

attach the siptrace and debug log here for the group.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/call-timeout-tp6726607p6728720.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From mitja.thomas1 at ewetel.de  Fri Aug 26 17:41:16 2011
From: mitja.thomas1 at ewetel.de (Mitja Thomas)
Date: Fri, 26 Aug 2011 15:41:16 +0200
Subject: [Freeswitch-users] Start Conference with first user
Message-ID: <4E57A27C.7070803@ewetel.de>

Hello List,

I try to start a conference in mod_conference when the first member 
attempts to join. This may sound weird, but the conference is set to 
wait for the moderator anyhow and I want to use the enter-sound to play 
something like "There is currently no Moderator in this room. Please 
hold the line". If the moderator joins, the enter-sound setting will be 
set to none from dialplan. I thought I just have to NOT set the mintwo 
member flag, but that didnt do the trick.
When the first member (no moderator) join there is no enter-sound, just 
the moh file. When the second member joins (no moderator) both get the 
enter-sound.
I thought there was a bug with the mintwo flag and I was close to submit 
to JIRA because of that, but looking further into the code Im not sure 
if the mintwo flag is even supposed to behave the way I want to. It just 
kills the conference when less then two users are in it.

So what I want to know, am I expecting the wrong behaviour from the 
mintwo member flag? If so, is there a different approach to tell the 
conference users right at the beginning if there is a moderator present 
or not.

Regards,

Mitja


From chris at ghosttelecom.com  Fri Aug 26 18:08:53 2011
From: chris at ghosttelecom.com (Chris Martineau)
Date: Fri, 26 Aug 2011 15:08:53 +0100
Subject: [Freeswitch-users] rtp natting
Message-ID: <1D10AB188D6CCA46BB4369E3268E36EF309C1A@SVR01.ghosttelecom.local>

Hi,

 

I am new to freeswitch and wish to bolt it into our existing network to
provide conferencing services.

 

Currently we just use an opensips/rtpproxy configuration for simple
proxy switching.

 

However looking at the features of freeswitch it seems to me that I
could replace my entire setup with just the freeswitch platform. Would
that be a fair comment?

 

If so then there a number of things that I am struggling to find clearly
defined in the documentation.

 

1.       Currently our rtp proxy scenario issues a port to come back on
but waits for incoming rtp packets to determine the exact port to return
rtp on when the caller is behind a nat. Does freeswitch do this, does it
need to be configured to work that way? Freeswitch would be on a public
address.

2.       Currently opensips stores all user info in a mysql database and
with 500000+ users is easy to manipulate. How do you deal with such a
large user database in a pure xml environment such as freeswitch?

 

Many thanks for any help you can offer.

 

Regards

 

Chris

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110826/3cce58fe/attachment.html 

From buscom123+fs at gmail.com  Fri Aug 26 21:00:02 2011
From: buscom123+fs at gmail.com (R H)
Date: Fri, 26 Aug 2011 11:00:02 -0600
Subject: [Freeswitch-users] Mod_Managed Mono is not following the system LD
	Path
Message-ID: <CAAtQP55kB_ozxrA0j+Mv+0wPD301gO49ZUN32tvAiK+RqsJ4_g@mail.gmail.com>

Hey Everyone,

I have recently been asked to set up a FreeSwitch server on a company
development server and the development management wants to use Mono C# as
our language for adding logic to the platform.

Following the instructions at http://wiki.freeswitch.org/wiki/Mod_managed I
have been able to successfully compile freeswitch on a Suse 11.4 box, with a
custom installation of Mono 2.8 and I can *successfully execute Demo.csx*.

But here is the problem, the only way I can get Demo.csx to run correctly
without getting a NullReferenceException is to start the server from the
freeswitch mod directory:

    EG.

    $> cd /usr/local/freeswitch/mod/
    $> /usr/local/freeswitch/bin/freeswitch
  *  [SUCCESS!, Demo.csx will run correctly]*

    $> cd / (or any other directory)
    $> /usr/local/freeswitch/bin/freeswitch
    [FAIL, The server starts, calls work, everything is fine EXCEPT Demo.csx
now throws
    *NullReferenceExceptions when I hit it with a call that is routed to the
script by the dial plan]*

- I have tried changing the *dllmap* default to the mono installation, I
have tried adding
/usr/local/freeswitch/mod/*FreeSWITCH.Native.dll.config*with the
following text inside:
    """
    <configuration>
            <dllmap dll="mod_managed.dll" target="mod_managed.so"
os="!windows"/>
    </configuration>
    """

- I have tried adding /usr/local/freeswitch/mod/ to the system ld path
(/etc/ld.so.conf). I also reloaded the ld cache (ldconfig)

- I have tried copying FreeSWITCH.Managed.dll to the existing ld path
(/usr/local/lib/FreeSWITCH.Managed.dll)

- I even created a Console HelloWorld.exe that imports the Freeswitch
libraries and ran it by simply executing:
 $> mono
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110826/7fd5da4c/attachment-0001.html 

From buscom123+fs at gmail.com  Fri Aug 26 21:01:37 2011
From: buscom123+fs at gmail.com (R H)
Date: Fri, 26 Aug 2011 11:01:37 -0600
Subject: [Freeswitch-users] Mod_Managed Mono is not following the system
	LD Path
In-Reply-To: <CAAtQP55kB_ozxrA0j+Mv+0wPD301gO49ZUN32tvAiK+RqsJ4_g@mail.gmail.com>
References: <CAAtQP55kB_ozxrA0j+Mv+0wPD301gO49ZUN32tvAiK+RqsJ4_g@mail.gmail.com>
Message-ID: <CAAtQP57RhsNYx46HczXb=euvPvDttheBZ-2SG_RvbA7sL-uVVA@mail.gmail.com>

Oops, sorry guys, I accidentally sent before finishing. Basically I can get
the library to import using mono from the command line but it wont work in
FreeSwitch without starting the server from the mod directory.

What am I missing?

Thank you,

Ryan

On Fri, Aug 26, 2011 at 11:00 AM, R H <buscom123+fs at gmail.com> wrote:

> Hey Everyone,
>
> I have recently been asked to set up a FreeSwitch server on a company
> development server and the development management wants to use Mono C# as
> our language for adding logic to the platform.
>
> Following the instructions at http://wiki.freeswitch.org/wiki/Mod_managedI have been able to successfully compile freeswitch on a Suse 11.4 box, with
> a custom installation of Mono 2.8 and I can *successfully execute Demo.csx
> *.
>
> But here is the problem, the only way I can get Demo.csx to run correctly
> without getting a NullReferenceException is to start the server from the
> freeswitch mod directory:
>
>     EG.
>
>     $> cd /usr/local/freeswitch/mod/
>     $> /usr/local/freeswitch/bin/freeswitch
>   *  [SUCCESS!, Demo.csx will run correctly]*
>
>     $> cd / (or any other directory)
>     $> /usr/local/freeswitch/bin/freeswitch
>     [FAIL, The server starts, calls work, everything is fine EXCEPT
> Demo.csx now throws
>     *NullReferenceExceptions when I hit it with a call that is routed to
> the script by the dial plan]*
>
> - I have tried changing the *dllmap* default to the mono installation, I
> have tried adding /usr/local/freeswitch/mod/*FreeSWITCH.Native.dll.config*with the following text inside:
>     """
>     <configuration>
>             <dllmap dll="mod_managed.dll" target="mod_managed.so"
> os="!windows"/>
>     </configuration>
>     """
>
> - I have tried adding /usr/local/freeswitch/mod/ to the system ld path
> (/etc/ld.so.conf). I also reloaded the ld cache (ldconfig)
>
> - I have tried copying FreeSWITCH.Managed.dll to the existing ld path
> (/usr/local/lib/FreeSWITCH.Managed.dll)
>
> - I even created a Console HelloWorld.exe that imports the Freeswitch
> libraries and ran it by simply executing:
>  $> mono
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110826/2264b660/attachment.html 

From msc at freeswitch.org  Fri Aug 26 21:31:48 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 26 Aug 2011 10:31:48 -0700
Subject: [Freeswitch-users] Newbie - External Phone setup question
In-Reply-To: <006401cc62a1$b72a4840$257ed8c0$@accra.ca>
References: <006401cc62a1$b72a4840$257ed8c0$@accra.ca>
Message-ID: <CAKzWOxWVTJNraQ+Z4cO4UOxrYjV7TAy=nvS6h=eUf6EdaKSWHA@mail.gmail.com>

How did you create "external5090.xml"? Did you simply copy the external.xml
file and make a few changes? You might want to pastebin the SIP trace of
this dialog and also your external5090.xml file. The folks here are pretty
good at sniffing out the cause of these kinds of issues. Oh, and watch the
console for errors or warnings. Sometimes you'll get a warning about needing
to create a domain and a user, but that only appears on the fs console and
not in the SIP dialog.

-MC

On Wed, Aug 24, 2011 at 2:06 PM, Charles Bujold <cjbujold at accra.ca> wrote:

> ** **
>
> Trying to get a phone located in another office registered to the
> Freeswitch server.  I created an ?external5090.xml? in the sip external
> folder as per the wiki and then set the phone to connect to 5090.  But I
> cannot get the phone to register.  When I look at wireshark I see a request
> from the phone to register and then I see  an error ?SIP:403 Forbidden? from
> the server to the Phone.    I have created the extension in the Directory
> folder with the other internal extensions but I?m wondering if I?m missing
> something in the 1010extension.xml for it to work. (maybe the extension is
> located in the wrong folder, or some identification to say it is not in the
> local LAN?)****
>
> ** **
>
> I have checked the name and password and they are correct which seems to
> indicate the issue is at the server level.****
>
> ** **
>
> Any guidance on how to setup an external phone to Freeswitch would be
> appreciated.****
>
> ** **
>
> Thanks****
>
> ** **
>
> CJB****
>
> ** **
>
> ** **
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110826/940b0dea/attachment.html 

From msc at freeswitch.org  Fri Aug 26 21:33:08 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 26 Aug 2011 10:33:08 -0700
Subject: [Freeswitch-users] call_timeout
In-Reply-To: <1314364888504-6728720.post@n2.nabble.com>
References: <CAKrPhPMB9oDhie1ncB0pgCVw4-3o+uVjULPvX26ibLXsDkajPQ@mail.gmail.com>
	<1314364888504-6728720.post@n2.nabble.com>
Message-ID: <CAKzWOxXM=aPZzRQWN=uH=gLjGC=NPrpVWPGX4QXxcWkay+DN2w@mail.gmail.com>

On Fri, Aug 26, 2011 at 6:21 AM, Jeff Lenk <jeff at jefflenk.com> wrote:

> You should look at the siptrace to see what the other end is doing.
>
> sofia global siptrace on
>
> attach the siptrace and debug log here for the group.
>

Actually if you could put the siptrace and debug on
pastebin.freeswitch.orgthat would be really helpful. If you have a
freeswitch console debug log
then use "FreeSWITCH Log" as the syntax highlighting. It does purty colors.
:)

pastebin.freeswitch.org

Thanks,
MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110826/735ff73f/attachment.html 

From msc at freeswitch.org  Fri Aug 26 21:35:53 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 26 Aug 2011 10:35:53 -0700
Subject: [Freeswitch-users] Parsing Error on conference_set_auto_outcall
In-Reply-To: <CAC4JBEAfoomQ2vq121NCqA_E3Zj1_gnn9T4hyQVjy4st64pqvg@mail.gmail.com>
References: <CAC4JBEAfoomQ2vq121NCqA_E3Zj1_gnn9T4hyQVjy4st64pqvg@mail.gmail.com>
Message-ID: <CAKzWOxUiYznav-iWj=KXmMx4COEE0RAgYD-5LAhxb6GyiZg3SQ@mail.gmail.com>

How recent is your version of FreeSWITCH? Also, which users in that group
were registered and which were not? I'd like to see if others can reproduce
this behavior under similar conditions using the latest HEAD.

-MC

On Wed, Aug 24, 2011 at 5:41 PM, Duvid Rottenberg <adrottenberg at gmail.com>wrote:

> I was using the Mad Boss example and I was getting parse errors, I
> traced it down to the following line,
>
> <action application="conference_set_auto_outcall"
> data="${group_call(sales)}"/>
>
> when group_call gets expanded it adds the default dial string for each
> member of the group which ends up looking something like this.
>
> [sip_invite_domain=10.195.94.25,presence_id=1001 at xxx.xxx.xxx.xxx
> ]sofia/internal/sip:1001 at xxx.xxx.xxx.xxx,
> [sip_invite_domain=10.195.94.25]error/user_not_registered
>
> The parse error I got "Cannot create outgoing channel of type
> [presence_id=1003 at 10.195.94.25]error cause: [CHAN_NOT_IMPLEMENTED]"
> indicated that it was treating the comma inside the brackets as a
> separator between 2 dial strings. I worked around this for now by
> editing the default config for the directory to remove presence_id
> from the dial string and it's working.
>
> This appears to be a bug in the parser as I was getting this error
> with the default configurations.
>
> Thanks,
> Duvid Rottenberg
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110826/333ae89c/attachment.html 

From kris at kriskinc.com  Fri Aug 26 22:01:19 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Fri, 26 Aug 2011 14:01:19 -0400
Subject: [Freeswitch-users] rtp natting
In-Reply-To: <1D10AB188D6CCA46BB4369E3268E36EF309C1A@SVR01.ghosttelecom.local>
References: <1D10AB188D6CCA46BB4369E3268E36EF309C1A@SVR01.ghosttelecom.local>
Message-ID: <CAKDfjgdhB_uMd4pMgUPzdKfsh8J3erE+yKc-Su_bzhcLpcTp6w@mail.gmail.com>

Keep doing what you are doing with OpenSIPS.  Add FreeSWITCH for
things it's good at:

- SBC functionality
- Hosted voice apps (like conferencing)

FreeSWITCH is good at those other things too but the SERs are still
king of scale.

On Fri, Aug 26, 2011 at 10:08 AM, Chris Martineau
<chris at ghosttelecom.com> wrote:
> Hi,
>
>
>
> I am new to freeswitch and wish to bolt it into our existing network to
> provide conferencing services.
>
>
>
> Currently we just use an opensips/rtpproxy configuration for simple proxy
> switching.
>
>
>
> However looking at the features of freeswitch it seems to me that I could
> replace my entire setup with just the freeswitch platform. Would that be a
> fair comment?
>
>
>
> If so then there a number of things that I am struggling to find clearly
> defined in the documentation.
>
>
>
> 1.?????? Currently our rtp proxy scenario issues a port to come back on but
> waits for incoming rtp packets to determine the exact port to return rtp on
> when the caller is behind a nat. Does freeswitch do this, does it need to be
> configured to work that way? Freeswitch would be on a public address.
>
> 2.?????? Currently opensips stores all user info in a mysql database and
> with 500000+ users is easy to manipulate. How do you deal with such a large
> user database in a pure xml environment such as freeswitch?
>
>
>
> Many thanks for any help you can offer.
>
>
>
> Regards
>
>
>
> Chris
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Kristian Kielhofner


From msc at freeswitch.org  Fri Aug 26 22:35:25 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 26 Aug 2011 11:35:25 -0700
Subject: [Freeswitch-users] PRI to VoIP gateway devices
Message-ID: <CAKzWOxV=EAym0GXXJjN-gaCgyLeyPrS_pO2rnPGwR50mv-hA+g@mail.gmail.com>

Hello gang,

If anyone out there has a PRI to VoIP gateway device that we can borrow (or
that you'd like to donate) to the FreeSWITCH project please let me know. We
are looking for something like the Vegastream 400 or the Audiocodes Mediant
600. If you have anything like these that are just collecting dust on a
shelf somewhere please contact me off list.

Thanks!
-Michael
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110826/d184f6c4/attachment-0001.html 

From beffa at ieee.org  Fri Aug 26 23:25:41 2011
From: beffa at ieee.org (Federico Beffa)
Date: Fri, 26 Aug 2011 21:25:41 +0200
Subject: [Freeswitch-users] call_timeout
Message-ID: <CAKrPhPOZJHzb+gFXts3NGr8vU=tfFkh2i+vT29+CN-4gPJVdTQ@mail.gmail.com>

Dear Jeff and Michael,

thanks for the answer. I've put siptrace and log on pastebin:

http://pastebin.freeswitch.org/17205

The case I've monitored was me calling with my mobile phone my desk phone
(Snom 300). The desk phone is registered to Freeswitch and the call goes
through my provider. Note that after ca. 19 secs my mobile phone makes a new
call, while if I try to do the same, but calling with X-lite, then X-lite
just hangs up.

The problem seems to be the CANCEL signal at line 459. Am I right in
interpreting this signal as originating from my provider? If that is the
case, then is there a way to tell my provider's proxy to wait longer?

If my interpretation is correct, then I'm surprised that the same does not
happen with Asterisk, where the call does not terminate (nor is re-started)
for more than 1 minute.

Thanks for any advise!
Fede
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110826/4bb6840d/attachment.html 

From raison at chatsubo.net  Fri Aug 26 23:45:45 2011
From: raison at chatsubo.net (Kevin Raison)
Date: Fri, 26 Aug 2011 12:45:45 -0700
Subject: [Freeswitch-users] odd behavior with freeswitch current
Message-ID: <4E57F7E9.7090004@chatsubo.net>

I am running the most current version of Freeswitch from git on Ubuntu 
Linux 9.10 64 bit.  I have a pair of these systems that are handling no 
more than 12 simultaneous calls each.  Over time, Freeswitch begins to 
take up more and more CPU, and after about 2 weeks, it is using up so 
much CPU that call quality starts to degrade.  It will be churning away 
even when there are no active calls.  A restart of Freeswitch cures the 
problem for a couple of weeks.  Can someone give me pointers on where to 
start debugging this issue?  Or perhaps a better remedy than a restart?


Thanks!

Kevin Raison


From aakashviswam at gmail.com  Fri Aug 26 23:47:19 2011
From: aakashviswam at gmail.com (Aakash)
Date: Fri, 26 Aug 2011 12:47:19 -0700 (PDT)
Subject: [Freeswitch-users] Error SRTP unprotect
Message-ID: <1314388039164-6730234.post@n2.nabble.com>


Hi Guys,

I am using latest git version.Sometimes i am getting an error as below


 [ERR] switch_rtp.c:2559 Error: SRTP unprotect failed with code 7 (auth
check failed)


The calls got disconnected when this error hits.

Thanks,
Aakash

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Error-SRTP-unprotect-tp6730234p6730234.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Sat Aug 27 00:06:56 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 26 Aug 2011 15:06:56 -0500
Subject: [Freeswitch-users] odd behavior with freeswitch current
In-Reply-To: <4E57F7E9.7090004@chatsubo.net>
References: <4E57F7E9.7090004@chatsubo.net>
Message-ID: <CAKbxfG8t-M56V9osw55ih3dXBJJWC46e2RCFssp4qzgn+aD8Uw@mail.gmail.com>

if you are running the most current version of FS but it takes 2 weeks
to notice the problem then the newest your FS could be today is 2
weeks old?   =D are you getting your updates from GIT?

What I recommend is when you have the problem where there is
noticeable CPU usage, run top -H and look at the cpu usage on a
per-thread basis.  When you find the one that is using a lot of CPU,
make note of it.  Then get a gcore and dump it into GDB so you can see
what the thread in question is doing..

you can get a gcore from the fs build root with ./support-d/fscore_pb
gcore <yourname>

Then compare the results on the url provided from that script with the
info from top -H



On Fri, Aug 26, 2011 at 2:45 PM, Kevin Raison <raison at chatsubo.net> wrote:
> I am running the most current version of Freeswitch from git on Ubuntu
> Linux 9.10 64 bit. ?I have a pair of these systems that are handling no
> more than 12 simultaneous calls each. ?Over time, Freeswitch begins to
> take up more and more CPU, and after about 2 weeks, it is using up so
> much CPU that call quality starts to degrade. ?It will be churning away
> even when there are no active calls. ?A restart of Freeswitch cures the
> problem for a couple of weeks. ?Can someone give me pointers on where to
> start debugging this issue? ?Or perhaps a better remedy than a restart?
>
>
> Thanks!
>
> Kevin Raison
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From yungwei at resolvity.com  Sat Aug 27 00:30:17 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Fri, 26 Aug 2011 16:30:17 -0400
Subject: [Freeswitch-users] How to save voicemail msgs to a db
Message-ID: <33095823FD21DF429B481B5163264B79511864FA82@VMBX102.ihostexchange.net>

Hi,

I'm wondering if there's a way to save voicemail messages to a (possibly remote) database. If so, how can that be achieved? 
Thanks.


From gavin.henry at gmail.com  Sat Aug 27 00:37:54 2011
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 26 Aug 2011 21:37:54 +0100
Subject: [Freeswitch-users] call_timeout
In-Reply-To: <CAKrPhPOZJHzb+gFXts3NGr8vU=tfFkh2i+vT29+CN-4gPJVdTQ@mail.gmail.com>
References: <CAKrPhPOZJHzb+gFXts3NGr8vU=tfFkh2i+vT29+CN-4gPJVdTQ@mail.gmail.com>
Message-ID: <CAA8_NKDU3_S5aB+_fzQx-LG7GcBMRw+ZvgNOS_1MpareXbxhiQ@mail.gmail.com>

Can you re-phrase your first paragraph?

On Friday, 26 August 2011, Federico Beffa <beffa at ieee.org> wrote:
> Dear Jeff and Michael,
>
> thanks for the answer. I've put siptrace and log on pastebin:
>
> http://pastebin.freeswitch.org/17205
>
> The case I've monitored was me calling with my mobile phone my desk phone
(Snom 300). The desk phone is registered to Freeswitch and the call goes
through my provider. Note that after ca. 19 secs my mobile phone makes a new
call, while if I try to do the same, but calling with X-lite, then X-lite
just hangs up.

-- 
http://www.suretecsystems.com/services/openldap/
http://www.surevoip.co.uk
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110826/c8a98901/attachment.html 

From gavin.henry at gmail.com  Sat Aug 27 00:39:35 2011
From: gavin.henry at gmail.com (Gavin Henry)
Date: Fri, 26 Aug 2011 21:39:35 +0100
Subject: [Freeswitch-users] rtp natting
In-Reply-To: <1D10AB188D6CCA46BB4369E3268E36EF309C1A@SVR01.ghosttelecom.local>
References: <1D10AB188D6CCA46BB4369E3268E36EF309C1A@SVR01.ghosttelecom.local>
Message-ID: <CAA8_NKDMawG_fvdWY5pOoTfxaUXzj8Z9uMRRmFr7to7j--ZjUA@mail.gmail.com>

500,000+ Very cool.

-- 
http://www.suretecsystems.com/services/openldap/
http://www.surevoip.co.uk
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110826/73183cb2/attachment.html 

From raison at chatsubo.net  Sat Aug 27 00:41:47 2011
From: raison at chatsubo.net (Kevin Raison)
Date: Fri, 26 Aug 2011 13:41:47 -0700
Subject: [Freeswitch-users] odd behavior with freeswitch current
In-Reply-To: <CAKbxfG8t-M56V9osw55ih3dXBJJWC46e2RCFssp4qzgn+aD8Uw@mail.gmail.com>
References: <4E57F7E9.7090004@chatsubo.net>
	<CAKbxfG8t-M56V9osw55ih3dXBJJWC46e2RCFssp4qzgn+aD8Uw@mail.gmail.com>
Message-ID: <4E58050B.5070402@chatsubo.net>

> if you are running the most current version of FS but it takes 2 weeks
> to notice the problem then the newest your FS could be today is 2
> weeks old?   =D are you getting your updates from GIT?

I have gone through several iterations of pulling new updates from git, 
rebuilding and restarting freeswitch and waiting while the CPU usage 
rises.  I will rebuild again today before I restart as well.  Hopefully 
the problem will just go away this time.  Otherwise, I will let you know 
what I find using your method;  thanks for the help!

-Kevin


> What I recommend is when you have the problem where there is
> noticeable CPU usage, run top -H and look at the cpu usage on a
> per-thread basis.  When you find the one that is using a lot of CPU,
> make note of it.  Then get a gcore and dump it into GDB so you can see
> what the thread in question is doing..
>
> you can get a gcore from the fs build root with ./support-d/fscore_pb
> gcore<yourname>
>
> Then compare the results on the url provided from that script with the
> info from top -H


From anthony.minessale at gmail.com  Sat Aug 27 00:55:55 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 26 Aug 2011 15:55:55 -0500
Subject: [Freeswitch-users] odd behavior with freeswitch current
In-Reply-To: <4E58050B.5070402@chatsubo.net>
References: <4E57F7E9.7090004@chatsubo.net>
	<CAKbxfG8t-M56V9osw55ih3dXBJJWC46e2RCFssp4qzgn+aD8Uw@mail.gmail.com>
	<4E58050B.5070402@chatsubo.net>
Message-ID: <CAKbxfG-oZZoObZhj0HeoJySw6RVkrdS7hoHK0hxJ-2RGnzGm7Q@mail.gmail.com>

what about the recipe i gave you to figure it out?
you can just send me the results if you want me to examine it as long
as you can collect it.


On Fri, Aug 26, 2011 at 3:41 PM, Kevin Raison <raison at chatsubo.net> wrote:
>> if you are running the most current version of FS but it takes 2 weeks
>> to notice the problem then the newest your FS could be today is 2
>> weeks old? ? =D are you getting your updates from GIT?
>
> I have gone through several iterations of pulling new updates from git,
> rebuilding and restarting freeswitch and waiting while the CPU usage
> rises. ?I will rebuild again today before I restart as well. ?Hopefully
> the problem will just go away this time. ?Otherwise, I will let you know
> what I find using your method; ?thanks for the help!
>
> -Kevin
>
>
>> What I recommend is when you have the problem where there is
>> noticeable CPU usage, run top -H and look at the cpu usage on a
>> per-thread basis. ?When you find the one that is using a lot of CPU,
>> make note of it. ?Then get a gcore and dump it into GDB so you can see
>> what the thread in question is doing..
>>
>> you can get a gcore from the fs build root with ./support-d/fscore_pb
>> gcore<yourname>
>>
>> Then compare the results on the url provided from that script with the
>> info from top -H
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From beffa at ieee.org  Sat Aug 27 01:15:40 2011
From: beffa at ieee.org (Federico Beffa)
Date: Fri, 26 Aug 2011 23:15:40 +0200
Subject: [Freeswitch-users] call_timeout
Message-ID: <CAKrPhPN69EAO-iCpyayKrsAq7AjRGN8hsmP0gq+vZB5TnM-Z-Q@mail.gmail.com>

The call monitored in the siptrace was as follows:

the call originated from my mobile phone. The destination was my desk
phone, a Snom 300. The Snom 300 is registered with FreeSwitch.
The call from the mobile phone arrives to FreeSwitch through my provider.

After ca. 19 secs the mobile phone seems to hangup and start a new
call (all automatically with no manual intervention).
If instead of calling from my mobile phone, I originate the call from
X-lite (softphone), then, after ca. 19 secs the call is terminated.

I'm trying to figure out how to prevent the call to be terminated or
re-started after these 19 seconds.

Hope this is clearer.
Regards,
Fede

>Can you re-phrase your first paragraph?
>
>On Friday, 26 August 2011, Federico Beffa <beffa at ieee.org> wrote:
>> Dear Jeff and Michael,
>>
>> thanks for the answer. I've put siptrace and log on pastebin:
>>
>> http://pastebin.freeswitch.org/17205
>>
>> The case I've monitored was me calling with my mobile phone my desk phone
>(Snom 300). The desk phone is registered to Freeswitch and the call goes
>through my provider. Note that after ca. 19 secs my mobile phone makes a new
>call, while if I try to do the same, but calling with X-lite, then X-lite
>just hangs up.


From adrottenberg at gmail.com  Sat Aug 27 02:46:03 2011
From: adrottenberg at gmail.com (Duvid Rottenberg)
Date: Fri, 26 Aug 2011 18:46:03 -0400
Subject: [Freeswitch-users] Parsing Error on conference_set_auto_outcall
In-Reply-To: <CAKzWOxUiYznav-iWj=KXmMx4COEE0RAgYD-5LAhxb6GyiZg3SQ@mail.gmail.com>
References: <CAC4JBEAfoomQ2vq121NCqA_E3Zj1_gnn9T4hyQVjy4st64pqvg@mail.gmail.com>
	<CAKzWOxUiYznav-iWj=KXmMx4COEE0RAgYD-5LAhxb6GyiZg3SQ@mail.gmail.com>
Message-ID: <CAC4JBEAX9GaZhEnQkch-jjD4fp0FQs=d-8-_+s+16TwdVusURw@mail.gmail.com>

I pulled the latest from git the begining of this weekl. I am running
it on windows server 2008 (although I don't think that makes a
difference). Only user 1001 was registered.

On Fri, Aug 26, 2011 at 1:35 PM, Michael Collins <msc at freeswitch.org> wrote:
> How recent is your version of FreeSWITCH? Also, which users in that group
> were registered and which were not? I'd like to see if others can reproduce
> this behavior under similar conditions using the latest HEAD.
> -MC
>
> On Wed, Aug 24, 2011 at 5:41 PM, Duvid Rottenberg <adrottenberg at gmail.com>
> wrote:
>>
>> I was using the Mad Boss example and I was getting parse errors, I
>> traced it down to the following line,
>>
>> <action application="conference_set_auto_outcall"
>> data="${group_call(sales)}"/>
>>
>> when group_call gets expanded it adds the default dial string for each
>> member of the group which ends up looking something like this.
>>
>>
>> [sip_invite_domain=10.195.94.25,presence_id=1001 at xxx.xxx.xxx.xxx]sofia/internal/sip:1001 at xxx.xxx.xxx.xxx,
>> [sip_invite_domain=10.195.94.25]error/user_not_registered
>>
>> The parse error I got "Cannot create outgoing channel of type
>> [presence_id=1003 at 10.195.94.25]error cause: [CHAN_NOT_IMPLEMENTED]"
>> indicated that it was treating the comma inside the brackets as a
>> separator between 2 dial strings. I worked around this for now by
>> editing the default config for the directory to remove presence_id
>> from the dial string and it's working.
>>
>> This appears to be a bug in the parser as I was getting this error
>> with the default configurations.
>>
>> Thanks,
>> Duvid Rottenberg
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From robert.hadley at teotech.com  Sat Aug 27 03:41:48 2011
From: robert.hadley at teotech.com (Robert Hadley)
Date: Fri, 26 Aug 2011 16:41:48 -0700
Subject: [Freeswitch-users] How to save voicemail msgs to a db
In-Reply-To: <33095823FD21DF429B481B5163264B79511864FA82@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B79511864FA82@VMBX102.ihostexchange.net>
Message-ID: <E826C94A3C6BE44198FAF4D66B5AB4D81BDC435DF9@VA3DIAXVS351.RED001.local>

Hi,

If you are using odbc to connect to your database, http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core

Then tell mod_voicemail about the DB via this param in autoload_configs/voicemail.conf.xml:

      <param name="odbc-dsn" value="dsn:user:pass"/>

Regards,
Robert


-----Original Message-----
From: Yungwei Chen [mailto:yungwei at resolvity.com] 
Sent: Friday, August 26, 2011 1:30 PM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] How to save voicemail msgs to a db

Hi,

I'm wondering if there's a way to save voicemail messages to a (possibly remote) database. If so, how can that be achieved? 
Thanks.



From jeff at jefflenk.com  Sat Aug 27 07:15:56 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Fri, 26 Aug 2011 20:15:56 -0700 (PDT)
Subject: [Freeswitch-users] call_timeout
In-Reply-To: <CAKrPhPN69EAO-iCpyayKrsAq7AjRGN8hsmP0gq+vZB5TnM-Z-Q@mail.gmail.com>
References: <CAKrPhPN69EAO-iCpyayKrsAq7AjRGN8hsmP0gq+vZB5TnM-Z-Q@mail.gmail.com>
Message-ID: <1314414956577-6731226.post@n2.nabble.com>

Your provider seems to be dropping the call on you.

see -

recv 415 bytes from udp/[195.190.1xx.2xx]:5060 at 18:57:29.322712:
   ------------------------------------------------------------------------
   CANCEL
sip:gw+ticinocom_private at 84.55.2xx.xx:5080;transport=udp;gw=ticinocom_private
SIP/2.0
   Via: SIP/2.0/UDP
195.190.1xx.2xx:5060;branch=z9hG4bK-d8754z-81e78d37b38d1735-1---d8754z-;rport
   Max-Forwards: 70
   To: &lt;sip:41916001220 at 195.190.1xx.2xx&gt;
   From:
"0765681626"&lt;sip:0765681626 at 195.190.1xx.2xx&gt;;tag=36agpbj7t724x3im.o
   Call-ID: 7b9707bd-9e83a87-521dc9e3-3979 at 62.65.1xx.5x
   CSeq: 737 CANCEL
   Content-Length: 0


--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/call-timeout-tp6730468p6731226.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From beffa at ieee.org  Sat Aug 27 12:23:49 2011
From: beffa at ieee.org (Federico Beffa)
Date: Sat, 27 Aug 2011 10:23:49 +0200
Subject: [Freeswitch-users] call_timeout
Message-ID: <CAKrPhPMBmSHQ-o9bYLcK673=3DD3LnjvREFdx1XsUcGDeJNh5A@mail.gmail.com>

Is there a way to force the provider's proxy to wait longer with some
kind of signal? I'm using "ring_ready" before transfer/bridge. Should
I add or replace it with something else?

On another system I have Asterisk installed (I'm planning to replace
it with freeswitch) and for some reason the same provider does not
drop the call for much longer. I guess I will have to trace the
Asterisk call and compare...

Thanks,
Fede

>Your provider seems to be dropping the call on you.
>
>see -
>
>recv 415 bytes from udp/[195.190.1xx.2xx]:5060 at 18:57:29.322712:
>   ------------------------------------------------------------------------
>   CANCEL
>sip:gw+ticinocom_private at 84.55.2xx.xx:5080;transport=udp;gw=ticinocom_private
>SIP/2.0
 >  Via: SIP/2.0/UDP
>195.190.1xx.2xx:5060;branch=z9hG4bK-d8754z-81e78d37b38d1735-1---d8754z-;rport
>   Max-Forwards: 70
>   To: &lt;sip:41916001220 at 195.190.1xx.2xx&gt;
>   From:
>"0765681626"&lt;sip:0765681626 at 195.190.1xx.2xx&gt;;tag=36agpbj7t724x3im.o
>   Call-ID: 7b9707bd-9e83a87-521dc9e3-3979 at 62.65.1xx.5x
>   CSeq: 737 CANCEL
>   Content-Length: 0


From beffa at ieee.org  Sat Aug 27 13:29:09 2011
From: beffa at ieee.org (Federico Beffa)
Date: Sat, 27 Aug 2011 11:29:09 +0200
Subject: [Freeswitch-users] call_timeout
Message-ID: <CAKrPhPMqL24YuZ3dUb3Ngh1VHSn-i_W0Ar_LYruF9TPsLEBE9w@mail.gmail.com>

I've just traced a call on my Asterisk system and ... well, I get
exactly the same behavior. The provider must have changed settings
recently without any notice. In fact my mailbox is never reached
anymore since I'm waiting for 30 seconds, but they CANCEL after 20...

sorry for making you loose time... In any case I've learned how to
trace calls in FreeSwitch :-)
Regards,
Fede


From lakersman2006 at yahoo.com  Sat Aug 27 20:26:38 2011
From: lakersman2006 at yahoo.com (Sam)
Date: Sat, 27 Aug 2011 09:26:38 -0700 (PDT)
Subject: [Freeswitch-users] freeswitch php
In-Reply-To: <CAKzWOxUNFYNRPEknF0uF5gKUZBzfTzOr4R2gCS10SdoQ1fSP8A@mail.gmail.com>
References: <1313970987.96106.YahooMailNeo@web161008.mail.bf1.yahoo.com>
	<CAKzWOxWTGEYrAwHz20QO0JAYSk7oc9DyLOkVXiJ1=u8v4AAkpA@mail.gmail.com>
	<1314060883.62351.YahooMailNeo@web161008.mail.bf1.yahoo.com>
	<CAKzWOxUNFYNRPEknF0uF5gKUZBzfTzOr4R2gCS10SdoQ1fSP8A@mail.gmail.com>
Message-ID: <1314462398.75774.YahooMailNeo@web161009.mail.bf1.yahoo.com>

MC,

Any luck with fs_ivrd with ESL?



________________________________
From: Michael Collins <msc at freeswitch.org>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, August 23, 2011 10:06 AM
Subject: Re: [Freeswitch-users] freeswitch php


Aha! fs_ivrd. I haven't tinkered w/ fs_ivrd outside of Perl, but it should work with anything that can read/write STDIN/STDOUT. Give me a few hours to work on some day job items and then I'll hop in here and take a look. Maybe you can help me wikify some of this knowledge. ;)

-MC

P.S. - If anyone else has or is using fs_ivrd and has some experience with it please let me know. I'd like to borrow some of your mental cycles so that we can document this better.


On Mon, Aug 22, 2011 at 5:54 PM, Sam <lakersman2006 at yahoo.com> wrote:

OK so if I were to use the ESL lib with PHP, can it still be used with the fs_ivrd daemon? I have not seen any ESL examples that is used with the fs_ivrd deamon, so I am not exactly to use the ESL lib.
>
>
>
>
>________________________________
> From: Michael Collins <msc at freeswitch.org>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>Sent: Monday, August 22, 2011 5:27 PM
>Subject: Re: [Freeswitch-users] freeswitch php
>
>
>
>This probably isn't the most efficient way of using the event socket. In fact, I'm not sure if this is anything more than a proof of concept. You are MUCH better off using the ESL lib with PHP and taking advantage of the abstraction you get.
>
>
>-MC
>
>
>On Sun, Aug 21, 2011 at 4:56 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
>Does anyone know how to retrieve channel variables (ie. uuid, etc.) using the php example that was shown in the wiki below?
>>
>>
>>
>>#!/usr/bin/php -q <?php // set a couple of things so we dont kill the system
ob_implicit_flush(true);
set_time_limit(30); // Open stdin so we can read the data in
$in = fopen("php://stdin", "r"); // Connect
echo "connect\n\n"; // Answer
echo "sendmsg\n";
echo "call-command: execute\n";
echo "execute-app-name: answer\n\n"; // Play a prompt
echo "sendmsg\n";
echo "call-command: execute\n";
echo "execute-app-name: playback\n";
echo "execute-app-arg: /usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav\n\n"; // Wait
sleep(5); // Hangup
echo "sendmsg\n";
echo "call-command: hangup\n\n"; fclose($in); ?>
>>
>>FreeSWITCH-users mailing list
>>FreeSWITCH-users at lists.freeswitch.org
>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>http://www.freeswitch.org
>>
>>
>
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>
>
>
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>


FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110827/d74bbe55/attachment.html 

From jeff at jefflenk.com  Sat Aug 27 20:34:46 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Sat, 27 Aug 2011 09:34:46 -0700 (PDT)
Subject: [Freeswitch-users] Parsing Error on conference_set_auto_outcall
In-Reply-To: <CAC4JBEAX9GaZhEnQkch-jjD4fp0FQs=d-8-_+s+16TwdVusURw@mail.gmail.com>
References: <CAC4JBEAfoomQ2vq121NCqA_E3Zj1_gnn9T4hyQVjy4st64pqvg@mail.gmail.com>
	<CAKzWOxUiYznav-iWj=KXmMx4COEE0RAgYD-5LAhxb6GyiZg3SQ@mail.gmail.com>
	<CAC4JBEAX9GaZhEnQkch-jjD4fp0FQs=d-8-_+s+16TwdVusURw@mail.gmail.com>
Message-ID: <1314462886583-6732482.post@n2.nabble.com>

This should probably be reported to Jira so It does not get forgotten. Thanks

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Parsing-Error-on-conference-set-auto-outcall-tp6722868p6732482.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From brian at freeswitch.org  Sat Aug 27 22:52:31 2011
From: brian at freeswitch.org (Brian West)
Date: Sat, 27 Aug 2011 13:52:31 -0500
Subject: [Freeswitch-users] Parsing Error on conference_set_auto_outcall
In-Reply-To: <1314462886583-6732482.post@n2.nabble.com>
References: <CAC4JBEAfoomQ2vq121NCqA_E3Zj1_gnn9T4hyQVjy4st64pqvg@mail.gmail.com>
	<CAKzWOxUiYznav-iWj=KXmMx4COEE0RAgYD-5LAhxb6GyiZg3SQ@mail.gmail.com>
	<CAC4JBEAX9GaZhEnQkch-jjD4fp0FQs=d-8-_+s+16TwdVusURw@mail.gmail.com>
	<1314462886583-6732482.post@n2.nabble.com>
Message-ID: <FD7D0989-B421-4354-B58D-AA43836C4E86@freeswitch.org>

I would like to see the full logs something returned is not properly escaped to be used in an originate string.

/b

On Aug 27, 2011, at 11:34 AM, Jeff Lenk wrote:

> This should probably be reported to Jira so It does not get forgotten. Thanks

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110827/6d23a910/attachment-0001.html 

From djbinter at gmail.com  Sat Aug 27 23:42:50 2011
From: djbinter at gmail.com (DJB International)
Date: Sat, 27 Aug 2011 12:42:50 -0700
Subject: [Freeswitch-users] SYSTEM_SHUTDOWN
Message-ID: <CAAB3jWv9_KJDLR3cWiBq8MjT5gY-c3z2wes2gPF168qGA1=DZg@mail.gmail.com>

I have one FS that set up similar to the others, but the weird thing about
this one is that when I shutdown the FS when there are calls running, it
never generated CDRs of those running calls before the system got shutdown
while the other servers always generate those CDRs with hangup_cause
SYSTEM_SHUTDOWN.

What would be the reason for this particular server not to write the CDRs
when it's shutting down like the other servers?   By the way, I am using
mod_cdr_csv.

Thank you,
Dorn B.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110827/887374aa/attachment.html 

From moises.silva at gmail.com  Sun Aug 28 01:09:38 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Sat, 27 Aug 2011 17:09:38 -0400
Subject: [Freeswitch-users] AMR transcoding
In-Reply-To: <4E549532.8040006@gmail.com>
References: <4E521A2B.1050808@gmail.com>
	<CAA4nhyBFkosothGquPiAS5Po8OvZ9GTx5MP3FRnN999+w4TqPg@mail.gmail.com>
	<4E525B50.6080404@gmail.com>
	<CAA4nhyA4F9+GztJHh8Y7WJOAUzN+hdX+==FxtOKGZagvF5c_1w@mail.gmail.com>
	<4E53A92F.2080504@coppice.org>
	<CAA4nhyCU0bdHr_RQfhuNtwkeMTR=+g82GwxUY_ouFKrfLWkXNA@mail.gmail.com>
	<4E54543E.3050500@coppice.org> <4E549532.8040006@gmail.com>
Message-ID: <CAA4nhyDoDs2Qd=6Zyf5x7wR7mHv-1nWRZH95Js_YT5WyTu0Lgw@mail.gmail.com>

On Wed, Aug 24, 2011 at 2:07 AM, Michal Kopacki <mkopacki at gmail.com> wrote:
> On 2011-08-24 03:30, Steve Underwood wrote:
>>
>> As far as I know, you can't even licence AMR properly from Voice Age. I
>> understand their patent pool is far from complete. Even their G.729 pool
>> licencing, which is far more mature, can leave you getting unexpected
>> letters from lawyers.
>>
>> Steve
>
> ? ? In that case how one can do proper licensing ? I didn't found any
> other company selling AMR Licenses.
>

Did you try calling Voice Age? you should discuss this with them.

Moises Silva
Senior Software Engineer, Software Development Manager
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
L3R 9R6 Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com


From moises.silva at gmail.com  Sun Aug 28 01:27:42 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Sat, 27 Aug 2011 17:27:42 -0400
Subject: [Freeswitch-users] mod_conference - 2-digit DTMF and events
In-Reply-To: <CAOFe-q7CG0TZ_u0CsNQddbB42j=H41ELbWgzMy69UsfVTkEhMQ@mail.gmail.com>
References: <CAOFe-q7CG0TZ_u0CsNQddbB42j=H41ELbWgzMy69UsfVTkEhMQ@mail.gmail.com>
Message-ID: <CAA4nhyCj0FMq7n4TJn4pNDUEK8qwr_hp3AfR8Tr7+nBc+5+TNg@mail.gmail.com>

On Wed, Aug 24, 2011 at 12:57 PM, Michael Kemp <mickkemp at gmail.com> wrote:
> However, unless I press both the 2nd digit very quickly after the first one
> the event is not raised
>
> Is there any way of tuning the inter digit delay for caller-control groups
> so that multi-digit DTMF sequences are handled correctly?

Hi Michael,

I just implemented your feature request. However is not in mainstream
git yet, but you can simply add a remote and fetch the improvements
I've been doing to mod_conference here:

https://github.com/moises-silva/mod_conference-admin/

To merge this changes in your git:
# git remote add -t master moyhub
git://github.com/moises-silva/mod_conference-admin.git
# git fetch moyhub
# git merge moyhub/master

I updated my repository with latest FreeSWITCH git, there should not
be any problems when you merge.

I did not test this new changes to the timeout yet (I tested all the
other ones though), please test them and report feedback in the JIRA
ticket I created to track this improvements:

http://jira.freeswitch.org/browse/FS-3493

Moises Silva
Senior Software Engineer, Software Development Manager
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
L3R 9R6 Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com


From covici at ccs.covici.com  Sun Aug 28 02:40:37 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Sat, 27 Aug 2011 18:40:37 -0400
Subject: [Freeswitch-users] mod_conference - 2-digit DTMF and events
In-Reply-To: <CAA4nhyCj0FMq7n4TJn4pNDUEK8qwr_hp3AfR8Tr7+nBc+5+TNg@mail.gmail.com>
References: <CAOFe-q7CG0TZ_u0CsNQddbB42j=H41ELbWgzMy69UsfVTkEhMQ@mail.gmail.com>
	<CAA4nhyCj0FMq7n4TJn4pNDUEK8qwr_hp3AfR8Tr7+nBc+5+TNg@mail.gmail.com>
Message-ID: <19763.1314484837@ccs.covici.com>

this sounds interesting -- any chance of using the moderator flag or an
admin flag, so you could mute all, but all admin users and then back to
their original state?


Moises Silva <moises.silva at gmail.com> wrote:

> On Wed, Aug 24, 2011 at 12:57 PM, Michael Kemp <mickkemp at gmail.com> wrote:
> > However, unless I press both the 2nd digit very quickly after the first one
> > the event is not raised
> >
> > Is there any way of tuning the inter digit delay for caller-control groups
> > so that multi-digit DTMF sequences are handled correctly?
> 
> Hi Michael,
> 
> I just implemented your feature request. However is not in mainstream
> git yet, but you can simply add a remote and fetch the improvements
> I've been doing to mod_conference here:
> 
> https://github.com/moises-silva/mod_conference-admin/
> 
> To merge this changes in your git:
> # git remote add -t master moyhub
> git://github.com/moises-silva/mod_conference-admin.git
> # git fetch moyhub
> # git merge moyhub/master
> 
> I updated my repository with latest FreeSWITCH git, there should not
> be any problems when you merge.
> 
> I did not test this new changes to the timeout yet (I tested all the
> other ones though), please test them and report feedback in the JIRA
> ticket I created to track this improvements:
> 
> http://jira.freeswitch.org/browse/FS-3493
> 
> Moises Silva
> Senior Software Engineer, Software Development Manager
> Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
> L3R 9R6 Canada
> t. 1 905 474 1990 x128 | e. moy at sangoma.com
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From raison at chatsubo.net  Sun Aug 28 07:04:38 2011
From: raison at chatsubo.net (Kevin Raison)
Date: Sat, 27 Aug 2011 20:04:38 -0700
Subject: [Freeswitch-users] odd behavior with freeswitch current
In-Reply-To: <CAKbxfG8t-M56V9osw55ih3dXBJJWC46e2RCFssp4qzgn+aD8Uw@mail.gmail.com>
References: <4E57F7E9.7090004@chatsubo.net>
	<CAKbxfG8t-M56V9osw55ih3dXBJJWC46e2RCFssp4qzgn+aD8Uw@mail.gmail.com>
Message-ID: <4E59B046.8000601@chatsubo.net>

Thanks, Anthony, but I was able to determine that a 
mod_event_socket-based app was not properly closing connections to 
Freeswitch.  The Freeswitch threads that were eating up CPU were those 
handling the external application's unclosed sessions.  Now that the 
external app is closing connections properly, I am seeing no more 
CLOSED_WAIT connection states, and hopefully as the week progresses, 
Freeswitch won't eat up CPU as it was before.

Cheers and thanks again for the pointers,
Kevin


On 08/26/2011 01:06 PM, Anthony Minessale wrote:
> if you are running the most current version of FS but it takes 2 weeks
> to notice the problem then the newest your FS could be today is 2
> weeks old?   =D are you getting your updates from GIT?
>
> What I recommend is when you have the problem where there is
> noticeable CPU usage, run top -H and look at the cpu usage on a
> per-thread basis.  When you find the one that is using a lot of CPU,
> make note of it.  Then get a gcore and dump it into GDB so you can see
> what the thread in question is doing..
>
> you can get a gcore from the fs build root with ./support-d/fscore_pb
> gcore<yourname>
>
> Then compare the results on the url provided from that script with the
> info from top -H
>
>
>
> On Fri, Aug 26, 2011 at 2:45 PM, Kevin Raison<raison at chatsubo.net>  wrote:
>> I am running the most current version of Freeswitch from git on Ubuntu
>> Linux 9.10 64 bit.  I have a pair of these systems that are handling no
>> more than 12 simultaneous calls each.  Over time, Freeswitch begins to
>> take up more and more CPU, and after about 2 weeks, it is using up so
>> much CPU that call quality starts to degrade.  It will be churning away
>> even when there are no active calls.  A restart of Freeswitch cures the
>> problem for a couple of weeks.  Can someone give me pointers on where to
>> start debugging this issue?  Or perhaps a better remedy than a restart?
>>
>>
>> Thanks!
>>
>> Kevin Raison
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>


From alex at jajah.com  Sun Aug 28 11:46:11 2011
From: alex at jajah.com (Alex Massover)
Date: Sun, 28 Aug 2011 10:46:11 +0300
Subject: [Freeswitch-users] Forcing CN offer
In-Reply-To: <CAKbxfG-gO_=ey7qS_rOw_VrTtrVcAHvv+Cp=BrQZpeF_-5KQNA@mail.gmail.com>
References: <F5B5EF8F78C0BC4BB417E1DBB9D7F7D0180A64A996@jjex01.jajah.dublin>
	<CAKbxfG-gO_=ey7qS_rOw_VrTtrVcAHvv+Cp=BrQZpeF_-5KQNA@mail.gmail.com>
Message-ID: <F5B5EF8F78C0BC4BB417E1DBB9D7F7D0180A75B08F@jjex01.jajah.dublin>

Hello,

That's great, if the other party's stack is not happy without a=, I'll turn on the variable.

Many thanks! 

> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
> users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale
> Sent: ????? 25 ?????? 2011 19:31
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Forcing CN offer
> 
> if you set the var verbose_sdp=true globally in vars.xml or on a
> per_leg basis you will get the extra a= that some things mistakenly
> think are mandatory (like some older polycom firmwares) the goal is to
> make the sdp as small as possible to prevent going over the MTU
> 
> On Thu, Aug 25, 2011 at 4:25 AM, Alex Massover <alex at jajah.com> wrote:
> > Hello,
> >
> >
> >
> > I?m working on the following scenarios:
> >
> > scenario 1:? A? ? FS ? B
> >
> > scenario 2: C ? FS ? A
> >
> >
> >
> > Where A supports CN (a=rtpmap:13 CN/8000) and requires CN
> negotiation, and B
> > and C do not support and aren?t able to negotiate it.
> >
> >
> >
> > I successfully implemented scenario 1, CN is negotiated between A
> party and
> > FS, and not negotiated between FS and B.
> >
> >
> >
> > But scenario 2 doesn?t work for me! When C doesn?t offer CN in INVITE
> > towards FS, FS also doesn?t offer CN in SDP in INVITE towards A.
> >
> > And nothing helps, tried all the combination of VAD options. Gateway
> A is in
> > the same internal sip profile in both scenarios.
> >
> >
> >
> > This is the SDP in INVITE from C to FS:
> >
> >
> >
> > ??????????? Media Description, name and address (m): audio 30224
> RTP/AVP 0 8
> > 18 101
> >
> > ??????????? Media Attribute (a): rtpmap:0 PCMU/8000
> >
> > ??????????? Media Attribute (a): rtpmap:8 PCMA/8000
> >
> > ??????????? Media Attribute (a): rtpmap:18 G729/8000
> >
> > ??????????? Media Attribute (a): fmtp:18 annexb=no
> >
> > ??????????? Media Attribute (a): rtpmap:101 telephone-event/8000
> >
> > ??????????? Media Attribute (a): fmtp:101 0-16
> >
> > ??????????? Media Attribute (a): silenceSupp:off - - - -
> >
> > ??????????? Media Attribute (a): ptime:20
> >
> > ??????????? Media Attribute (a): sendrecv
> >
> >
> >
> >
> >
> > And this is the SDP from FS towards A:
> >
> > ??????????? Media Description, name and address (m): audio 23564
> RTP/AVP 0 8
> > 101 13
> >
> > ??????????? Media Attribute (a): rtpmap:101 telephone-event/8000
> >
> > ??????????? Media Attribute (a): fmtp:101 0-16
> >
> > ??????????? Media Attribute (a): ptime:20
> >
> >
> >
> > FS even puts 13 in m= attribute, but doesn?t add a=? attribute for
> > rtpmap:13.
> >
> >
> >
> >
> >
> > Is that a bug?
> >
> >
> >
> >
> >
> > --
> >
> > Best Regards
> >
> > Alex Massover.
> >
> > This mail was sent via Mail-SeCure System.
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> users
> > http://www.freeswitch.org
> >
> >
> 
> 
> 
> --
> Anthony Minessale II
> 
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> users
> http://www.freeswitch.org
> 
> This mail was received via Mail-SeCure System.
> 


This mail was sent via Mail-SeCure System.

From yehavi.bourvine at gmail.com  Sun Aug 28 13:58:03 2011
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sun, 28 Aug 2011 12:58:03 +0300
Subject: [Freeswitch-users] Delete voicemail greeting?
Message-ID: <CAAtWKeGeM8yggcLeR0s+0vJgEqhrST=00dznu9qwgiBHmgKuAw@mail.gmail.com>

Hello,

  In the voicemail menu (option 5) there is a way to record a new greeting
or select a pre-recorded one. However, I did not find a way to delete the
greeting and return to the deault one (so I am doing this for users by
deleting it from the database).
Is there such an option that I am not aware of?

                                     Thanks! __Yehavi:
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110828/5230acbc/attachment.html 

From steveu at coppice.org  Sun Aug 28 14:30:16 2011
From: steveu at coppice.org (Steve Underwood)
Date: Sun, 28 Aug 2011 18:30:16 +0800
Subject: [Freeswitch-users] AMR transcoding
In-Reply-To: <CAA4nhyDoDs2Qd=6Zyf5x7wR7mHv-1nWRZH95Js_YT5WyTu0Lgw@mail.gmail.com>
References: <4E521A2B.1050808@gmail.com>	<CAA4nhyBFkosothGquPiAS5Po8OvZ9GTx5MP3FRnN999+w4TqPg@mail.gmail.com>	<4E525B50.6080404@gmail.com>	<CAA4nhyA4F9+GztJHh8Y7WJOAUzN+hdX+==FxtOKGZagvF5c_1w@mail.gmail.com>	<4E53A92F.2080504@coppice.org>	<CAA4nhyCU0bdHr_RQfhuNtwkeMTR=+g82GwxUY_ouFKrfLWkXNA@mail.gmail.com>	<4E54543E.3050500@coppice.org>
	<4E549532.8040006@gmail.com>
	<CAA4nhyDoDs2Qd=6Zyf5x7wR7mHv-1nWRZH95Js_YT5WyTu0Lgw@mail.gmail.com>
Message-ID: <4E5A18B8.5030901@coppice.org>

On 08/28/2011 05:09 AM, Moises Silva wrote:
> On Wed, Aug 24, 2011 at 2:07 AM, Michal Kopacki<mkopacki at gmail.com>  wrote:
>> On 2011-08-24 03:30, Steve Underwood wrote:
>>> As far as I know, you can't even licence AMR properly from Voice Age. I
>>> understand their patent pool is far from complete. Even their G.729 pool
>>> licencing, which is far more mature, can leave you getting unexpected
>>> letters from lawyers.
>>>
>>> Steve
>>      In that case how one can do proper licensing ? I didn't found any
>> other company selling AMR Licenses.
>>
> Did you try calling Voice Age? you should discuss this with them.
Be warned. Voice Age may not properly inform you about patents outside 
their pool. This is understandable, if you think about it. To do so 
would be expressing an opinion about a possibly touchy legal issue.

Steve



From adrottenberg at gmail.com  Sun Aug 28 18:37:16 2011
From: adrottenberg at gmail.com (Duvid Rottenberg)
Date: Sun, 28 Aug 2011 10:37:16 -0400
Subject: [Freeswitch-users] Parsing Error on conference_set_auto_outcall
In-Reply-To: <FD7D0989-B421-4354-B58D-AA43836C4E86@freeswitch.org>
References: <CAC4JBEAfoomQ2vq121NCqA_E3Zj1_gnn9T4hyQVjy4st64pqvg@mail.gmail.com>
	<CAKzWOxUiYznav-iWj=KXmMx4COEE0RAgYD-5LAhxb6GyiZg3SQ@mail.gmail.com>
	<CAC4JBEAX9GaZhEnQkch-jjD4fp0FQs=d-8-_+s+16TwdVusURw@mail.gmail.com>
	<1314462886583-6732482.post@n2.nabble.com>
	<FD7D0989-B421-4354-B58D-AA43836C4E86@freeswitch.org>
Message-ID: <CAC4JBEDwXfCAkDUn=pY7ugU8Cg_=sxRd3+QxYV9O7v=p3Fwq9A@mail.gmail.com>

I put the logs in the pastebin at http://pastebin.freeswitch.org/17220


On Sat, Aug 27, 2011 at 2:52 PM, Brian West <brian at freeswitch.org> wrote:
> I would like to see the full logs something returned is not properly escaped
> to be used in an originate string.
> /b
> On Aug 27, 2011, at 11:34 AM, Jeff Lenk wrote:
>
> This should probably be reported to Jira so It does not get forgotten.
> 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 curriegrad2004 at gmail.com  Sun Aug 28 18:51:54 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sun, 28 Aug 2011 07:51:54 -0700
Subject: [Freeswitch-users] AMR transcoding
In-Reply-To: <4E5A18B8.5030901@coppice.org>
References: <4E521A2B.1050808@gmail.com>
	<CAA4nhyBFkosothGquPiAS5Po8OvZ9GTx5MP3FRnN999+w4TqPg@mail.gmail.com>
	<4E525B50.6080404@gmail.com>
	<CAA4nhyA4F9+GztJHh8Y7WJOAUzN+hdX+==FxtOKGZagvF5c_1w@mail.gmail.com>
	<4E53A92F.2080504@coppice.org>
	<CAA4nhyCU0bdHr_RQfhuNtwkeMTR=+g82GwxUY_ouFKrfLWkXNA@mail.gmail.com>
	<4E54543E.3050500@coppice.org> <4E549532.8040006@gmail.com>
	<CAA4nhyDoDs2Qd=6Zyf5x7wR7mHv-1nWRZH95Js_YT5WyTu0Lgw@mail.gmail.com>
	<4E5A18B8.5030901@coppice.org>
Message-ID: <CAMWiMy_jcbaUmmimtKi25iaenYUMofLs51QfvVmVePx7YS9JCQ@mail.gmail.com>

Steve,

Do you have any proof to back this up? Because if not it just sounds
like a bunch of FUD to me.

On Sun, Aug 28, 2011 at 3:30 AM, Steve Underwood <steveu at coppice.org> wrote:
> On 08/28/2011 05:09 AM, Moises Silva wrote:
>> On Wed, Aug 24, 2011 at 2:07 AM, Michal Kopacki<mkopacki at gmail.com> ?wrote:
>>> On 2011-08-24 03:30, Steve Underwood wrote:
>>>> As far as I know, you can't even licence AMR properly from Voice Age. I
>>>> understand their patent pool is far from complete. Even their G.729 pool
>>>> licencing, which is far more mature, can leave you getting unexpected
>>>> letters from lawyers.
>>>>
>>>> Steve
>>> ? ? ?In that case how one can do proper licensing ? I didn't found any
>>> other company selling AMR Licenses.
>>>
>> Did you try calling Voice Age? you should discuss this with them.
> Be warned. Voice Age may not properly inform you about patents outside
> their pool. This is understandable, if you think about it. To do so
> would be expressing an opinion about a possibly touchy legal issue.
>
> 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
>


From steveu at coppice.org  Sun Aug 28 19:39:57 2011
From: steveu at coppice.org (Steve Underwood)
Date: Sun, 28 Aug 2011 23:39:57 +0800
Subject: [Freeswitch-users] AMR transcoding
In-Reply-To: <CAMWiMy_jcbaUmmimtKi25iaenYUMofLs51QfvVmVePx7YS9JCQ@mail.gmail.com>
References: <4E521A2B.1050808@gmail.com>	<CAA4nhyBFkosothGquPiAS5Po8OvZ9GTx5MP3FRnN999+w4TqPg@mail.gmail.com>	<4E525B50.6080404@gmail.com>	<CAA4nhyA4F9+GztJHh8Y7WJOAUzN+hdX+==FxtOKGZagvF5c_1w@mail.gmail.com>	<4E53A92F.2080504@coppice.org>	<CAA4nhyCU0bdHr_RQfhuNtwkeMTR=+g82GwxUY_ouFKrfLWkXNA@mail.gmail.com>	<4E54543E.3050500@coppice.org>
	<4E549532.8040006@gmail.com>	<CAA4nhyDoDs2Qd=6Zyf5x7wR7mHv-1nWRZH95Js_YT5WyTu0Lgw@mail.gmail.com>	<4E5A18B8.5030901@coppice.org>
	<CAMWiMy_jcbaUmmimtKi25iaenYUMofLs51QfvVmVePx7YS9JCQ@mail.gmail.com>
Message-ID: <4E5A614D.2030200@coppice.org>

On 08/28/2011 10:51 PM, curriegrad2004 wrote:
> Steve,
>
> Do you have any proof to back this up? Because if not it just sounds
> like a bunch of FUD to me.
In what way is this FUD? Voice Age do not claim to represent everyone 
with a patent relevant to the codes they have pools for. From experience 
we know more about the handling of the G.729 codec. Voice Age says they 
represent most of the relevant patent holders. They point you to a 
couple of the others, which they don't represent. If you use more than 
the basic features of G.729 you will find lawyers letters arriving about 
patents Voice Age didn't mention.
> On Sun, Aug 28, 2011 at 3:30 AM, Steve Underwood<steveu at coppice.org>  wrote:
>> On 08/28/2011 05:09 AM, Moises Silva wrote:
>>> On Wed, Aug 24, 2011 at 2:07 AM, Michal Kopacki<mkopacki at gmail.com>    wrote:
>>>> On 2011-08-24 03:30, Steve Underwood wrote:
>>>>> As far as I know, you can't even licence AMR properly from Voice Age. I
>>>>> understand their patent pool is far from complete. Even their G.729 pool
>>>>> licencing, which is far more mature, can leave you getting unexpected
>>>>> letters from lawyers.
>>>>>
>>>>> Steve
>>>>       In that case how one can do proper licensing ? I didn't found any
>>>> other company selling AMR Licenses.
>>>>
>>> Did you try calling Voice Age? you should discuss this with them.
>> Be warned. Voice Age may not properly inform you about patents outside
>> their pool. This is understandable, if you think about it. To do so
>> would be expressing an opinion about a possibly touchy legal issue.
>>
>> Steve
>>
Steve



From ijurado at econcept.es  Sun Aug 28 21:43:46 2011
From: ijurado at econcept.es (Isaac Jurado)
Date: Sun, 28 Aug 2011 19:43:46 +0200
Subject: [Freeswitch-users] Delete voicemail greeting?
In-Reply-To: <CAAtWKeGeM8yggcLeR0s+0vJgEqhrST=00dznu9qwgiBHmgKuAw@mail.gmail.com>
References: <CAAtWKeGeM8yggcLeR0s+0vJgEqhrST=00dznu9qwgiBHmgKuAw@mail.gmail.com>
Message-ID: <CABj_V-=aRC_SzHHKYB+SFX=eDaREuJtMLyGMK+kkQdJ8XA6qWw@mail.gmail.com>

On Sun, Aug 28, 2011 at 11:58 AM, Yehavi Bourvine
<yehavi.bourvine at gmail.com> wrote:
>
> Hello,
>
> ? In the voicemail menu (option 5) there is a way to record a new
> greeting or select a pre-recorded one. However, I did not find a way
> to delete the greeting and return to the deault one (so I am doing
> this for users by deleting it from the database).  Is there such an
> option that I am not aware of?

skip_greeting?

http://wiki.freeswitch.org/wiki/Mod_voicemail#skip_greeting

I hope it helps.  Cheers.

-- 
Isaac Jurado
Internet Busines Solutions eConcept


From ijurado at econcept.es  Sun Aug 28 21:45:09 2011
From: ijurado at econcept.es (Isaac Jurado)
Date: Sun, 28 Aug 2011 19:45:09 +0200
Subject: [Freeswitch-users] Delete voicemail greeting?
In-Reply-To: <CABj_V-=aRC_SzHHKYB+SFX=eDaREuJtMLyGMK+kkQdJ8XA6qWw@mail.gmail.com>
References: <CAAtWKeGeM8yggcLeR0s+0vJgEqhrST=00dznu9qwgiBHmgKuAw@mail.gmail.com>
	<CABj_V-=aRC_SzHHKYB+SFX=eDaREuJtMLyGMK+kkQdJ8XA6qWw@mail.gmail.com>
Message-ID: <CABj_V-mBC_MDcGjvt0=Paj+DX2fSZcZ7ZUwfU3Oe1RMxDRy-8A@mail.gmail.com>

On Sun, Aug 28, 2011 at 7:43 PM, Isaac Jurado <ijurado at econcept.es> wrote:
> On Sun, Aug 28, 2011 at 11:58 AM, Yehavi Bourvine <yehavi.bourvine at gmail.com> wrote:
>>
>> Hello,
>>
>> ? In the voicemail menu (option 5) there is a way to record a new
>> greeting or select a pre-recorded one. However, I did not find a way
>> to delete the greeting and return to the deault one (so I am doing
>> this for users by deleting it from the database). ?Is there such an
>> option that I am not aware of?
>
> skip_greeting?
>
> http://wiki.freeswitch.org/wiki/Mod_voicemail#skip_greeting

Sorry, that's not what you were asking for.  I misunderstood.

-- 
Isaac Jurado
Internet Busines Solutions eConcept


From justlikeef at gmail.com  Mon Aug 29 00:13:18 2011
From: justlikeef at gmail.com (Rob Hutton)
Date: Sun, 28 Aug 2011 16:13:18 -0400
Subject: [Freeswitch-users] Multiple Tone Detect
Message-ID: <201108281613.18871.justlikeef@gmail.com>

I am trying to get tone detection set up so that a single number can be used for both voice and fax.  Tone detection needs to run concurrent to the normal call processing, so that a call proceeds normally unless the fax tones are heard.

On this wiki page (http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_fax_detect) it gives the format:

<action application="tone_detect" data="fax 1100 r +5000 transfer fax XML default" />

and an associated dialplan entry for the call to be transfered to that actually handles the fax reception.

On this wiki page (http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect) it says that the format from the first page is not the preferred method, and that the format should be something like:

<action application="set" data="execute_on_tone_detect=transfer fax XML default"/>
<action application="set" data="tone_detect_hits=1"/>
<action application="set" data="execute_on_media=tone_detect mytone 1100 r +5000"/>

My question is, using the first format, I could set up multiple detections, eg. one for fax tones that sends/receives a fax, one for answering machines tone that hangs up, etc.

Is this possible with the preferred syntax?


From justlikeef at gmail.com  Mon Aug 29 00:37:37 2011
From: justlikeef at gmail.com (Rob Hutton)
Date: Sun, 28 Aug 2011 16:37:37 -0400
Subject: [Freeswitch-users] Looking up variable from config
Message-ID: <201108281637.37738.justlikeef@gmail.com>

Is there any way of looking up a variable from the XML config from the dialplan?

For instance, I know what number is dialed from ${destination_number} and I know that the directory user entry attribute of ID matches the number because I have it set up that way.  Can I lookup the value email-addr parameter from the user and, say, set a variable to the value? Something like an XPATH lookup?


From brian at freeswitch.org  Mon Aug 29 01:07:31 2011
From: brian at freeswitch.org (Brian West)
Date: Sun, 28 Aug 2011 16:07:31 -0500
Subject: [Freeswitch-users] Looking up variable from config
In-Reply-To: <201108281637.37738.justlikeef@gmail.com>
References: <201108281637.37738.justlikeef@gmail.com>
Message-ID: <85E2FC60-DBA1-43B8-B57D-20572A096D64@freeswitch.org>

http://wiki.freeswitch.org/wiki/Mod_commands#user_data

${user_data()}

/b


On Aug 28, 2011, at 3:37 PM, Rob Hutton wrote:

> Is there any way of looking up a variable from the XML config from the dialplan?
> 
> For instance, I know what number is dialed from ${destination_number} and I know that the directory user entry attribute of ID matches the number because I have it set up that way.  Can I lookup the value email-addr parameter from the user and, say, set a variable to the value? Something like an XPATH lookup?
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From joe.jflemmings at gmail.com  Sun Aug 28 12:34:21 2011
From: joe.jflemmings at gmail.com (Joe Flemmings)
Date: Sun, 28 Aug 2011 01:34:21 -0700
Subject: [Freeswitch-users] Error Compiling esl - php
Message-ID: <CALH5ZrZ7jrP0ZsV-WOSC1FPH9ez63PkoDxGufA24vboHKf94bA@mail.gmail.com>

I'm getting the following error compiling esl for php

make phpmod
make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x"
CFLAGS="-I/usr/src/freeswitch-snapshot/libs/esl/src/include -DHAVE_EDITLINE
-g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror
-Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes"
CXXFLAGS="-I/usr/src/freeswitch-snapshot/libs/esl/src/include
-DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
-Wno-unused-variable" CXX_CFLAGS="" -C php
make[1]: Entering directory `/usr/src/freeswitch-snapshot/libs/esl/php'
g++  -I/usr/src/freeswitch-snapshot/libs/esl/src/include -DHAVE_EDITLINE -g
-ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable
-I/usr/include/php -I/usr/include/php/main -I/usr/include/php/TSRM
-I/usr/include/php/Zend -I/usr/include/php/ext
-I/usr/include/php/ext/date/lib -Wno-unused-label -Wno-unused-function -c
esl_wrap.cpp -o esl_wrap.o
cc1plus: warnings being treated as errors
esl_wrap.cpp:2591: error: deprecated conversion from string constant to
???char*???
make[1]: *** [esl_wrap.o] Error 1
make[1]: Leaving directory `/usr/src/freeswitch-snapshot/libs/esl/php'
make: *** [phpmod] Error 2
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110828/147728a0/attachment-0001.html 

From joe.jflemmings at gmail.com  Mon Aug 29 01:36:05 2011
From: joe.jflemmings at gmail.com (Joe Flemmings)
Date: Sun, 28 Aug 2011 14:36:05 -0700
Subject: [Freeswitch-users] Error Compiling esl - php
In-Reply-To: <CALH5ZrZ7jrP0ZsV-WOSC1FPH9ez63PkoDxGufA24vboHKf94bA@mail.gmail.com>
References: <CALH5ZrZ7jrP0ZsV-WOSC1FPH9ez63PkoDxGufA24vboHKf94bA@mail.gmail.com>
Message-ID: <CALH5Zrbx67O_w_ztdg39=xpyycKOop+caOvjNtitdPo89gOVYA@mail.gmail.com>

I'm getting the following error compiling esl for php

make phpmod
make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x"
CFLAGS="-I/usr/src/freeswitch-snapshot/libs/esl/src/include -DHAVE_EDITLINE
-g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror
-Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes"
CXXFLAGS="-I/usr/src/freeswitch-snapshot/libs/esl/src/include
-DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
-Wno-unused-variable" CXX_CFLAGS="" -C php
make[1]: Entering directory `/usr/src/freeswitch-snapshot/libs/esl/php'
g++  -I/usr/src/freeswitch-snapshot/libs/esl/src/include -DHAVE_EDITLINE -g
-ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable
-I/usr/include/php -I/usr/include/php/main -I/usr/include/php/TSRM
-I/usr/include/php/Zend -I/usr/include/php/ext
-I/usr/include/php/ext/date/lib -Wno-unused-label -Wno-unused-function -c
esl_wrap.cpp -o esl_wrap.o
cc1plus: warnings being treated as errors
esl_wrap.cpp:2591: error: deprecated conversion from string constant to
char*
make[1]: *** [esl_wrap.o] Error 1
make[1]: Leaving directory `/usr/src/freeswitch-snapshot/libs/esl/php'
make: *** [phpmod] Error 2
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110828/44532cf8/attachment-0001.html 

From dfoulkes at ihug.co.nz  Mon Aug 29 04:43:37 2011
From: dfoulkes at ihug.co.nz (D Foulkes)
Date: Mon, 29 Aug 2011 12:43:37 +1200
Subject: [Freeswitch-users] Error compling jsxml.c
Message-ID: <4E5AE0B9.7040307@ihug.co.nz>

Hi, I am still fairly new to freeswitch...

Having problems with a setup on a Ubuntu 10.10, so I decided to do a 
rebuild from scratch. Following recommended instructions on the wiki 
installation guide (using git clone) I get the following error when 
running the compile:

make && make install
...

libtool: compile: gcc -DOSSP -DXP_UNIX -DEXPORT_JS_API 
-DJS_HAS_FILE_OBJECT=1 -DJS_HAS_XML_SUPPORT=1 -DJS_THREADSAFE=1 
-Insprpub/dist/include/nspr -Insprpub/pr/include -DHAVE_CONFIG_H -Isrc 
-O2 -DNDEBUG -pipe -c src/jsxml.c -fPIC -DPIC -o src/.libs/jsxml.o
src/jsxml.c: In function ?PutProperty?:
src/jsxml.c:4571: error: expected expression before ?>? token
src/jsxml.c:4571: error: stray ?\37? in program
src/jsxml.c:4571: error: stray ?`? in program
src/jsxml.c:4572: error: stray ?\37? in program
src/jsxml.c:4572: error: stray ?\37? in program
src/jsxml.c:4572: error: stray ?\37? in program
src/jsxml.c:4572: error: stray ?\37? in program
src/jsxml.c:4572: error: stray ?\37? in program
src/jsxml.c:4573: error: stray ?\37? in program
src/jsxml.c:4573: error: stray ?\37? in program
src/jsxml.c:4587: error: ?JSXML? has no member named ?xml?
src/jsxml.c:4587: error: stray ?`? in program
src/jsxml.c:4587: error: expected ?;? before ?value?
src/jsxml.c:4588: error: expected expression before ?goto?
src/jsxml.c:4591: error: expected expression before ?/? token
src/jsxml.c:4592: error: expected statement before ?)? token
src/jsxml.c:4592: warning: missing terminating ' character
src/jsxml.c:4592: error: missing terminating ' character
src/jsxml.c:4593: error: ?JSXMLQName? has no member named ?localOame?
src/jsxml.c:4593: error: expected expression before ?)? token
make[5]: *** [src/jsxml.lo] Error 1
make[4]: *** [/usr/local/src/freeswitch/libs/js/libjs.la] Error 2
make[3]: *** [mod_spidermonkey-all] Error 1
make[2]: *** [all-recursive] Error 1
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2

Any help would be appreciated. Thanks,

Daniel


From member at linkedin.com  Mon Aug 29 05:50:48 2011
From: member at linkedin.com (Martin Armstrong via LinkedIn)
Date: Mon, 29 Aug 2011 01:50:48 +0000 (UTC)
Subject: [Freeswitch-users] Invitation to connect on LinkedIn
Message-ID: <1977956802.3028892.1314582648670.JavaMail.app@ela4-app0132.prod>

LinkedIn
------------




    Martin Armstrong requested to add you as a connection on LinkedIn:
  
------------------------------------------

Zohair,

I'd like to add you to my professional network on LinkedIn.

- Martin

Accept invitation from Martin Armstrong
http://www.linkedin.com/e/kwhdv8-grwskfh0-4q/GrULditvq6UNXRBIaIGnrCzzv0aYgRNIau2nD4Rpg0D0u0BXjFSyiLl/blk/I1648266809_3/1BpC5vrmRLoRZcjkkZt5YCpnlOt3RApnhMpmdzgmhxrSNBszYPnPAMe3oSczwQdz59bPtnnSlFmB5QbPANd3kQe3kMc3gLrCBxbOYWrSlI/EML_comm_afe/

View invitation from Martin Armstrong
http://www.linkedin.com/e/kwhdv8-grwskfh0-4q/GrULditvq6UNXRBIaIGnrCzzv0aYgRNIau2nD4Rpg0D0u0BXjFSyiLl/blk/I1648266809_3/3dvej0UdzoOe3gSckALqnpPbOYWrSlI/svi/ 
------------------------------------------

DID YOU KNOW LinkedIn can help you find the right service providers using recommendations from your trusted network? Using LinkedIn Services, you can take the risky guesswork out of selecting service providers by reading the recommendations of credible, trustworthy members of your network. 
http://www.linkedin.com/e/kwhdv8-grwskfh0-4q/svp/inv-25/
 
-- 
(c) 2011, LinkedIn Corporation
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110829/2d57154f/attachment.html 

From member at linkedin.com  Mon Aug 29 05:52:42 2011
From: member at linkedin.com (Martin Armstrong via LinkedIn)
Date: Mon, 29 Aug 2011 01:52:42 +0000 (UTC)
Subject: [Freeswitch-users] Invitation to connect on LinkedIn
Message-ID: <1649474201.3049560.1314582762091.JavaMail.app@ela4-app0135.prod>

LinkedIn
------------




    Martin Armstrong requested to add you as a connection on LinkedIn:
  
------------------------------------------

Zohair,

I'd like to add you to my professional network on LinkedIn.

- Martin

Accept invitation from Martin Armstrong
http://www.linkedin.com/e/kwhdv8-grwsmuzr-35/GrULditvq6UNXRBIaIGnrCzzv0aYgRNIau2nD4Rpg0D0u0BXjFSyiLl/blk/I1648269943_3/1BpC5vrmRLoRZcjkkZt5YCpnlOt3RApnhMpmdzgmhxrSNBszYPnPcQejASczwQdz59bPtnnSlFmB5QbPANd3kQe3kMc3gLrCBxbOYWrSlI/EML_comm_afe/

View invitation from Martin Armstrong
http://www.linkedin.com/e/kwhdv8-grwsmuzr-35/GrULditvq6UNXRBIaIGnrCzzv0aYgRNIau2nD4Rpg0D0u0BXjFSyiLl/blk/I1648269943_3/3dvcPgVejoOe3gSckALqnpPbOYWrSlI/svi/ 
------------------------------------------

DID YOU KNOW LinkedIn can help you find the right service providers using recommendations from your trusted network? Using LinkedIn Services, you can take the risky guesswork out of selecting service providers by reading the recommendations of credible, trustworthy members of your network. 
http://www.linkedin.com/e/kwhdv8-grwsmuzr-35/svp/inv-25/
 
-- 
(c) 2011, LinkedIn Corporation
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110829/bd235f95/attachment.html 

From kbdfck at gmail.com  Mon Aug 29 11:36:29 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Mon, 29 Aug 2011 11:36:29 +0400
Subject: [Freeswitch-users] Inband DTMF and bind_meta_app problem again
Message-ID: <CA+GmxbYZ1PHwf60O6+bqaypfOyMuXMkLTm7=BFC4=nmuW_Ng=g@mail.gmail.com>

Hi All
After recent updates of DTMF negotiation or some associated stuff we can't
use bind_meta_app with inband DTMF. bind_meta_app ignores bindings, although
start_dtmf and spandsp_start_dtmf detect DTMF. We use git-fb6e979 2011-08-26
04-48-33 +0000.


<anti-action application="set" data="dtmf_type=${pbx_call_dtmf_type}"/>
<anti-action application="spandsp_start_dtmf"/>
<anti-action application="bind_meta_app" data="7 a s
execute_extension::att_xfer XML features"/>

EXECUTE sofia/local/test1 at 85.114.2.200 bind_meta_app(7 a s
execute_extension::att_xfer XML features)

2011-08-29 11:39:09.371414 [INFO] switch_ivr_async.c:3066 Bound A-Leg: *7
execute_extension::att_xfer XML features
...cut...
 2011-08-29 11:40:15.751415 [DEBUG] mod_spandsp_dsp.c:56 DTMF BEGIN
DETECTED: [*]
2011-08-29 11:40:15.751415 [DEBUG] switch_ivr_bridge.c:391 Send signal
sofia/external/89215572714 [BREAK]
2011-08-29 11:40:15.831571 [DEBUG] mod_spandsp_dsp.c:68 DTMF END DETECTED:
[*], duration = 80 ms
2011-08-29 11:40:15.951420 [DEBUG] mod_spandsp_dsp.c:56 DTMF BEGIN DETECTED:
[7]
2011-08-29 11:40:15.951420 [DEBUG] switch_ivr_bridge.c:391 Send signal
sofia/external/89215572714 [BREAK]
2011-08-29 11:40:16.051411 [DEBUG] mod_spandsp_dsp.c:68 DTMF END DETECTED:
[7], duration = 100 ms
2011-08-29 11:40:17.391410 [DEBUG] mod_spandsp_dsp.c:56 DTMF BEGIN DETECTED:
[*]
2011-08-29 11:40:17.391410 [DEBUG] switch_ivr_bridge.c:391 Send signal
sofia/external/89215572714 [BREAK]
2011-08-29 11:40:17.471409 [DEBUG] mod_spandsp_dsp.c:68 DTMF END DETECTED:
[*], duration = 80 ms
2011-08-29 11:40:17.571412 [DEBUG] mod_spandsp_dsp.c:56 DTMF BEGIN DETECTED:
[7]
2011-08-29 11:40:17.571412 [DEBUG] switch_ivr_bridge.c:391 Send signal
sofia/external/89215572714 [BREAK]
2011-08-29 11:40:17.631411 [DEBUG] mod_spandsp_dsp.c:68 DTMF END DETECTED:
[7], duration = 60 ms

But no defined application is launched.


-- 
Best regards,

Dmitry Sytchev,
IT Engineer
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110829/5aa30ad0/attachment.html 

From john at whitesmiths.com  Mon Aug 29 06:49:19 2011
From: john at whitesmiths.com (John O'Brien)
Date: Mon, 29 Aug 2011 12:49:19 +1000
Subject: [Freeswitch-users] Build problem with mod_python CentOS 5.5 Python
	2.7
Message-ID: <583A1777-E07B-47C2-919B-B92E2CCF9E79@whitesmiths.com>

Hi,

I am trying to install mod_python on CentOS 5.5 x86_64.

Have downloaded the source of Python 2.7.2 built and installed it

# cd Python-2.7.2
# ./configure
# make
# make install

All good!!!

Have downloaded source of FreeSWITCH 1.0.6.
Edited /usr/local/src/freeswitch/modules.conf to enable mod_python

# cd /usr/local/src/freeswitch/
# ./configure
# make

Creating mod_python.so...
/usr/bin/ld: /usr/local/lib/libpython2.7.a(abstract.o): relocation R_X86_64_32 against `a local symbol' can not be used when making a shared object; recompile with -fPIC
/usr/local/lib/libpython2.7.a: could not read symbols: Bad value
collect2: ld returned 1 exit status
g++ -I/usr/local/include/python2.7 -I/usr/local/include/python2.7 -fPIC -fno-strict-aliasing -m64 -DNDEBUG -g -fwrapv -O3 -Wall -Wstrict-prototypes -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -shared -o .libs/mod_python.so -shared -Wl,-x .libs/mod_python.o -m64 freeswitch_python.o mod_python_wrap.o  /usr/local/src/freeswitch/.libs/libfreeswitch.so -L/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib -lpq /usr/local/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/local/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid -lrt -lcrypt -L/usr/local/src/freeswitch/libs/srtp -L/usr/kerberos/lib64 -lssl -lcrypto -lz -lncurses -L/usr/local/lib -lpthread -ldl -lutil -lm -lpython2.7   -Wl,--rpath -Wl,/usr/local/freeswitch/lib -Wl,--rpath -Wl,/usr/local/freeswitch/mod
make[1]: *** [mod_python.so] Error 1
make: *** [all] Error 1

The problem appears to be the python library libpython2.7.a was not been built with PIC flag.

Have tried various incantations to build a compatible Python library without success.

Does anyone know how to get mod_python to work with Python 2.7?

Regards,

John


From lrmipsum0 at gmail.com  Mon Aug 29 17:34:32 2011
From: lrmipsum0 at gmail.com (Tom Fayette)
Date: Mon, 29 Aug 2011 15:34:32 +0200
Subject: [Freeswitch-users] Problem with receiving a NOTIFY after
 sending a SUBSCRIBE request
In-Reply-To: <CAKzWOxXvx4HSVnMUCvd49sspkgv1NfCtHjXqs74ge2o7Cg0f_w@mail.gmail.com>
References: <CANSA4prMK6zbUX4xBQdBKkFS1qxuEg0cRacSpsJY=Jn51B_e6w@mail.gmail.com>
	<CAKzWOxXvx4HSVnMUCvd49sspkgv1NfCtHjXqs74ge2o7Cg0f_w@mail.gmail.com>
Message-ID: <CANSA4pqgp4aXEPtDB1kHaA9YRqdZ5jyJeEbYUDh6hbWT5qJq5A@mail.gmail.com>

My guess would be that in the field, he has already deployed some servers
which expect the To: field to be composed like that :) When I'll have a
chance, I'll ask.

Anyway, I have tweaked the sofia_presence.c in a rather ugly way, hard
coding the reference from the Contact: header

--- src/mod/endpoints/mod_sofia/sofia_presence.c        2011-08-18
05:55:20.000000000 -0400
+++ src/mod/endpoints/mod_sofia/sofia_presence.c_tweaked
2011-08-18 05:51:04.000000000 -0400
@@ -2160,11 +2160,13 @@
        }

        if (to) {
-               to_str = switch_mprintf("sip:%s@%s",
to->a_url->url_user, to->a_url->url_host);
+               //to_str = switch_mprintf("sip:%s@%s",
to->a_url->url_user, to->a_url->url_host);
+               to_str = switch_mprintf("sip:%s@%s", contact_user,
to->a_url->url_host);
        }

        if (to) {
-               to_user = to->a_url->url_user;
+               //to_user = to->a_url->url_user;
+               to_user = contact_user;
                to_host = to->a_url->url_host;
        }


So far I haven't seen any downside of this hack.

Regards,
Tom

On Wed, Aug 17, 2011 at 6:59 PM, Michael Collins <msc at freeswitch.org> wrote:

> This is interesting. I'm looking at the SIP book by Alan Johnston (one of
> the guys who authored the SIP spec) and I can't see any indication that the
> To: field can contain an invalid URI like that.
>
> Question: why, exactly, does the client feel that it needs to violate such
> a basic principle?
>
> -MC
>
> On Tue, Aug 16, 2011 at 7:58 AM, Lorem Ipsum <lrmipsum0 at gmail.com> wrote:
>
>> Hello,
>>
>> I'm testing a SIP stack for an embedded device. The device, among other
>> things, is capable of informing a user about pending messages on the
>> voicemail. It does that by subscribing to the message-summary. Below some
>> wireshark traces (172.16.30.68 is my device, 172.16.31.10 is FreeSWITCH):
>>
>> REGISTER sip:172.16.31.10 SIP/2.0
>> Via: SIP/2.0/UDP 172.16.30.68:5080
>> ;rport;branch=z9hG4bKPj051b000000035ea40edf
>> Route: <sip:172.16.31.10:5060;lr>
>> Max-Forwards: 70
>> From: <sip:399510002 at 172.16.31.10>;tag=051b000000025ea40edf
>> To: <sip:399510002 at 172.16.31.10>
>> Call-ID: 051b000000015ea40edf
>> CSeq: 1 REGISTER
>> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY
>> User-Agent: My_Sip_Device
>> Contact: <sip:399510002 at 172.16.30.68:5080>;transport=udp
>> Content-Length:  0
>>
>> SIP/2.0 401 Unauthorized
>> Via: SIP/2.0/UDP 172.16.30.68:5080
>> ;rport=5080;branch=z9hG4bKPj051b000000035ea40edf
>> From: <sip:399510002 at 172.16.31.10>;tag=051b000000025ea40edf
>> To: <sip:399510002 at 172.16.31.10>;tag=mQrcXUcvrtUmS
>> Call-ID: 051b000000015ea40edf
>> CSeq: 1 REGISTER
>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-38e3f5f 2011-08-09 03-09-19
>> -0400
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
>> REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> Supported: timer, precondition, path, replaces
>> WWW-Authenticate: Digest realm="172.16.31.10",
>> nonce="3139f996-c814-11e0-93d2-05aa0ee343d6", algorithm=MD5, qop="auth"
>> Content-Length: 0
>>
>> REGISTER sip:172.16.31.10 SIP/2.0
>> Via: SIP/2.0/UDP 172.16.30.68:5080
>> ;rport;branch=z9hG4bKPj051b000000065ea40edf
>> Route: <sip:172.16.31.10:5060;lr>
>> Max-Forwards: 70
>> From: <sip:399510002 at 172.16.31.10>;tag=051b000000045ea40edf
>> To: <sip:399510002 at 172.16.31.10>
>> Call-ID: 051b000000015ea40edf
>> CSeq: 2 REGISTER
>> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY
>> User-Agent: My_Sip_Device
>> Contact: <sip:399510002 at 172.16.30.68:5080>;transport=udp
>> Authorization: Digest username="399510002", realm="172.16.31.10",
>> nonce="3139f996-c814-11e0-93d2-05aa0ee343d6", uri="sip:172.16.31.10",
>> response="1e95409a562c074cbe6df148a85107ef", algorithm=MD5,
>> cnonce="051b000000055ea40edf", qop=auth, nc=00000001
>> Content-Length:  0
>>
>> SIP/2.0 200 OK
>> Via: SIP/2.0/UDP 172.16.30.68:5080
>> ;rport=5080;branch=z9hG4bKPj051b000000065ea40edf
>> From: <sip:399510002 at 172.16.31.10>;tag=051b000000045ea40edf
>> To: <sip:399510002 at 172.16.31.10>;tag=N0H5ypXZN3H7m
>> Call-ID: 051b000000015ea40edf
>> CSeq: 2 REGISTER
>> Contact: <sip:399510002 at 172.16.30.68:5080>;transport=udp;expires=180
>> Date: Tue, 16 Aug 2011 14:29:47 GMT
>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-38e3f5f 2011-08-09 03-09-19
>> -0400
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
>> REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> Supported: timer, precondition, path, replaces
>> Content-Length: 0
>>
>>
>> SUBSCRIBE sip:172.16.31.10 SIP/2.0
>> Via: SIP/2.0/UDP 172.16.30.68:5080
>> ;rport;branch=z9hG4bKPj051b0000000a5ea40edf
>> Max-Forwards: 69
>> From: "399510002" <sip:399510002 at 172.16.31.10>;tag=051b000000085ea40edf
>> To: <sip:172.16.31.10>
>> Contact: <sip:399510002 at 172.16.30.68:5080>
>> Call-ID: 051b000000095ea40edf
>> CSeq: 1 SUBSCRIBE
>> Event: message-summary
>> Accept: application/simple-message-summary
>> Allow-Events: message-summary
>> User-Agent: My_Sip_Device
>> X-Serialnumber: LMZ091218000026
>> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY
>> Route: <sip:172.16.31.10:5060;lr>
>> Content-Length:  0
>>
>> SIP/2.0 200 OK
>> Via: SIP/2.0/UDP 172.16.30.68:5080
>> ;rport=5080;branch=z9hG4bKPj051b0000000a5ea40edf
>> From: "399510002" <sip:399510002 at 172.16.31.10>;tag=051b000000085ea40edf
>> To: <sip:172.16.31.10>;tag=p9ay0He3jc8Sg
>> Call-ID: 051b000000095ea40edf
>> CSeq: 1 SUBSCRIBE
>> Contact: <sip:@172.16.31.10:5060>
>> Expires: 60
>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-38e3f5f 2011-08-09 03-09-19
>> -0400
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
>> REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> Supported: timer, precondition, path, replaces
>> Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
>> include-session-description, presence.winfo, message-summary, refer
>> Subscription-State: active;expires=60
>> Content-Length: 0
>>
>>
>> After sending the 200 OK, FreeSWITCH does not send the NOTIFY.
>> If you look at the Contact header of the answer to the SUBSCRIBE you will
>> notice that the part before the "@" is missing. I guess this is because
>> SUBSCRIBE request does not contain the whole URI, just the host part. That
>> is because our customer wants it done this way; the request line should look
>> like this:
>> SUBSCRIBE sip:voicemail_server SIP/2.0
>>
>> and the To: header should look like this:
>> To: <sip:voicemail_server>
>>
>> My question is: how can I make FreeSWITCH (the NOTIFY part anyway, other
>> things are working OK) work with such a device? Thanks.
>>
>> Regards,
>> Tom
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110829/65b20e83/attachment-0001.html 

From yungwei at resolvity.com  Mon Aug 29 17:53:37 2011
From: yungwei at resolvity.com (Yungwei Chen)
Date: Mon, 29 Aug 2011 09:53:37 -0400
Subject: [Freeswitch-users] How to save voicemail msgs to a db
In-Reply-To: <E826C94A3C6BE44198FAF4D66B5AB4D81BDC435DF9@VA3DIAXVS351.RED001.local>
References: <33095823FD21DF429B481B5163264B79511864FA82@VMBX102.ihostexchange.net>
	<E826C94A3C6BE44198FAF4D66B5AB4D81BDC435DF9@VA3DIAXVS351.RED001.local>
Message-ID: <33095823FD21DF429B481B5163264B79511864FB4B@VMBX102.ihostexchange.net>

Thanks for your feedback.
Can the voicemail msgs (*.wav) be saved to the db as well?

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Robert Hadley
Sent: Friday, August 26, 2011 6:42 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] How to save voicemail msgs to a db

Hi,

If you are using odbc to connect to your database, http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core

Then tell mod_voicemail about the DB via this param in autoload_configs/voicemail.conf.xml:

      <param name="odbc-dsn" value="dsn:user:pass"/>

Regards,
Robert


-----Original Message-----
From: Yungwei Chen 
Sent: Friday, August 26, 2011 1:30 PM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] How to save voicemail msgs to a db

Hi,

I'm wondering if there's a way to save voicemail messages to a (possibly remote) database. If so, how can that be achieved? 
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 moises.silva at gmail.com  Mon Aug 29 18:12:20 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Mon, 29 Aug 2011 10:12:20 -0400
Subject: [Freeswitch-users] freetdm api - ftdm_channel_read()
In-Reply-To: <1314190075421-6719985.post@n2.nabble.com>
References: <CAAqVCyTi9m0+JPhjt_sLLkEa97ijFLmETjPoFRUcEQJRmQsN8w@mail.gmail.com>
	<CAA4nhyDc9i=raCOk1mzbJ3vVJYSAAKHGwT0b7wyMbaZrA+Wz=Q@mail.gmail.com>
	<1314190075421-6719985.post@n2.nabble.com>
Message-ID: <CAA4nhyB_ds1fj=ruKGvog744LaeHHa_iVESkH7Ep16eqjEMwiQ@mail.gmail.com>

On Wed, Aug 24, 2011 at 8:47 AM, jfabo <juraj.fabo at gmail.com> wrote:
> Hi Moises,
> My target platform is SLES 11. I installed wanpipe-3.5.20
> but did not find the queue setting within wanpipe.conf
> (at least not explicitely as queue size)
> However, setting it via libfreetdm API was successful.

Do not confuse it with /etc/wanpipe/wanpipex.conf, the wanpipe.conf
file I am talking about is the freetdm wanpipe.conf file, the sample
file is found in conf/wanpipe.conf in the freetdm source code
directory.

> Yes, I just have different framing on the second network stack = 20ms,
> ?while on the E1 I have 10ms, that was why I asked about the reading method.
> Since read is done by polling, I would like to initiate it from the other
> network stack thread (here I use ACE framework and
> have all streams/mixers triggered from single Reactor/thread)

You are going to be better off by setting the timing to be the same as
the network stack.

> My intention was to use single thread to handle read/write operations
> on all active channels, which is different than the approach described
> ?in freetdm/doc/locking.txt (where thread per call leg is mentioned).
> I worry most about calling the wait_channel() with timeout = -1 which
> afaik leads to blocking wait on single channel. As mentioned earlier,
> when this is called from single thread prio each particular
> ftdm_channel_write()
> ?it represents danger of blocking and latency. I measure time spent
> in each particular channel wait+write and the result was < 10us.

If you're following this approach, you can safely assume all channels
in a card are ready to read/write at the same time, so polling in just
one channel is enough (just make sure the channel you poll on is not
alarmed, ie, physical line disconnected). The card delivers data to
the all the channels in the same card under the same hardware clock
(either coming from the telco or the internal oscillator). The
approach other applications follow is one thread per span, then you
poll in a single channel and read/write for all of them when waken up.

> Maybe rather a simple question, does the application which uses
> libfreetdm need to run thread per call/channel?

No, we have at least one application that even work in a completely
different mode (span mode), where all the IO read/write is done for
the whole span in a single operation in a single thread, unfortunately
to do this you need to modify ftmod_wanpipe or code your own I/O
module making use of libsangoma span mode. Probably not worth it yet.

> should I repost also this?

You can move your future replies there.

Moises Silva
Senior Software Engineer, Software Development Manager
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
L3R 9R6 Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com


From moises.silva at gmail.com  Mon Aug 29 18:13:32 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Mon, 29 Aug 2011 10:13:32 -0400
Subject: [Freeswitch-users] mod_conference - 2-digit DTMF and events
In-Reply-To: <19763.1314484837@ccs.covici.com>
References: <CAOFe-q7CG0TZ_u0CsNQddbB42j=H41ELbWgzMy69UsfVTkEhMQ@mail.gmail.com>
	<CAA4nhyCj0FMq7n4TJn4pNDUEK8qwr_hp3AfR8Tr7+nBc+5+TNg@mail.gmail.com>
	<19763.1314484837@ccs.covici.com>
Message-ID: <CAA4nhyCMxrUN8MzA6WTa+8wUKOpyvxud5c5A_4Zhjqo+roT+vA@mail.gmail.com>

On Sat, Aug 27, 2011 at 6:40 PM,  <covici at ccs.covici.com> wrote:
> this sounds interesting -- any chance of using the moderator flag or an
> admin flag, so you could mute all, but all admin users and then back to
> their original state?

It is doable for sure, given that me or someone else finds the time to
implement it.

Moises Silva
Senior Software Engineer, Software Development Manager
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON
L3R 9R6 Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com


From rhuddleston at gmail.com  Mon Aug 29 18:25:57 2011
From: rhuddleston at gmail.com (Robert Huddleston)
Date: Mon, 29 Aug 2011 10:25:57 -0400
Subject: [Freeswitch-users] CallWithUs
Message-ID: <005001cc6657$925b32e0$b71198a0$@com>

I already reviewed the wiki - does anyone have experience with CallWithUs
and using IP Auth / not registration based?

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110829/350975e7/attachment.html 

From aakashviswam at gmail.com  Mon Aug 29 18:33:26 2011
From: aakashviswam at gmail.com (Aakash)
Date: Mon, 29 Aug 2011 07:33:26 -0700 (PDT)
Subject: [Freeswitch-users] Dialed number identification service
Message-ID: <1314628406963-6737827.post@n2.nabble.com>

Hi,

Is this possible - Dialed number identification service in freeswitch.

For eg:

I have configured my two inbound number to the same extension(eg 1000).When
calling party dials any of the two inbound number should able to show the
DNIS to the routed extension(eg 1000).


Thanks
Aakash



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Dialed-number-identification-service-tp6737827p6737827.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From jerre at j-cope.com  Mon Aug 29 18:09:35 2011
From: jerre at j-cope.com (Jerre Cope)
Date: Mon, 29 Aug 2011 09:09:35 -0500
Subject: [Freeswitch-users] Error compling jsxml.c
In-Reply-To: <mailman.1827.1314625137.29749.freeswitch-users@lists.freeswitch.org>
References: <mailman.1827.1314625137.29749.freeswitch-users@lists.freeswitch.org>
Message-ID: <4E5B9D9F.2030907@j-cope.com>

Be sure you have these packages installed:

    * git-core subversion build-essential autoconf automake libtool
      libncurses5 libncurses5-dev libjpeg62-dev libcurl4-gnutls-dev
      libglobus-openssl-dev libglobus-openssl-module-dev
    * for perl scripts will need: libcurses-perl




On 08/29/2011 08:38 AM, freeswitch-users-request at lists.freeswitch.org 
wrote:
> Send FreeSWITCH-users mailing list submissions to
> 	freeswitch-users at lists.freeswitch.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
> 	http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> or, via email, send a message with subject or body 'help' to
> 	freeswitch-users-request at lists.freeswitch.org
>
> You can reach the person managing the list at
> 	freeswitch-users-owner at lists.freeswitch.org
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of FreeSWITCH-users digest..."
>
>
> Today's Topics:
>
>     1. Error compling jsxml.c (D Foulkes)
>     2. Invitation to connect on LinkedIn (Martin Armstrong via LinkedIn)
>     3. Invitation to connect on LinkedIn (Martin Armstrong via LinkedIn)
>     4. Inband DTMF and bind_meta_app problem again (Dmitry Sytchev)
>     5. Build problem with mod_python CentOS 5.5 Python	2.7 (John O'Brien)
>     6. Re: Problem with receiving a NOTIFY after sending a SUBSCRIBE
>        request (Tom Fayette)
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110829/509973ce/attachment.html 

From justlikeef at gmail.com  Mon Aug 29 19:07:52 2011
From: justlikeef at gmail.com (Rob Hutton)
Date: Mon, 29 Aug 2011 11:07:52 -0400
Subject: [Freeswitch-users] Looking up variable from config
In-Reply-To: <85E2FC60-DBA1-43B8-B57D-20572A096D64@freeswitch.org>
References: <201108281637.37738.justlikeef@gmail.com>
	<85E2FC60-DBA1-43B8-B57D-20572A096D64@freeswitch.org>
Message-ID: <201108291107.53211.justlikeef@gmail.com>

Thank you very much...

On Sunday 28 August 2011 17:07:31 Brian West wrote:
> http://wiki.freeswitch.org/wiki/Mod_commands#user_data
> 
> ${user_data()}
> 
> /b
> 
> 
> On Aug 28, 2011, at 3:37 PM, Rob Hutton wrote:
> 
> > Is there any way of looking up a variable from the XML config from the dialplan?
> > 
> > For instance, I know what number is dialed from ${destination_number} and I know that the directory user entry attribute of ID matches the number because I have it set up that way.  Can I lookup the value email-addr parameter from the user and, say, set a variable to the value? Something like an XPATH lookup?
> > 
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


From darcy at primrose.ws  Mon Aug 29 19:23:38 2011
From: darcy at primrose.ws (Darcy)
Date: Mon, 29 Aug 2011 11:23:38 -0400
Subject: [Freeswitch-users] CallWithUs
In-Reply-To: <005001cc6657$925b32e0$b71198a0$@com>
References: <005001cc6657$925b32e0$b71198a0$@com>
Message-ID: <441F1D5786364B62BA13EBD8AB085999@DWP>

We used callwithus using the freeswitch basically as a tandem.  I believe you need to send them your ip address for them to put in their system, then everything else is just standard config.
Make sure their IP address is in your acl list.
This is what I used in my dial plan.
<action application="bridge" data=mailto:data=sofia/${use_profile}/1$1 at carrier.callwithus.com/>

They asked us to do the following:
* CALLERID: set to blank
* HOST: Your SIP server IP address
* PORT: Your port (5060 for default port)
  Send calls to carrier.callwithus.com, port 5060. 

We stopped using them because to many of their routes kept going down and we did not get proper responses to declined calls, so customers got dead air.

Darcy

From: Robert Huddleston 
Sent: Monday, August 29, 2011 10:25 AM
To: 'FreeSWITCH Users Help' 
Subject: [Freeswitch-users] CallWithUs

I already reviewed the wiki ? does anyone have experience with CallWithUs and using IP Auth / not registration based?

 



--------------------------------------------------------------------------------

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110829/a3878fa4/attachment-0001.html 

From alex at jajah.com  Mon Aug 29 19:36:04 2011
From: alex at jajah.com (Alex Massover)
Date: Mon, 29 Aug 2011 18:36:04 +0300
Subject: [Freeswitch-users] Condition based on custom sip header
Message-ID: <F5B5EF8F78C0BC4BB417E1DBB9D7F7D0180A75B7CD@jjex01.jajah.dublin>

Hi,

I'm trying to implement dialplan condition based on custom SIP header ("Header1: value1"). As far as I understood there's no way to read custom SIP header which isn't X- header with sip_h_. Is that correct?

What are the other options, please? Is lua's getHeader() suitable for this?

--
Best Regards,
Alex Massover


This mail was sent via Mail-SeCure System.


-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110829/7b973ff8/attachment.html 

From rhuddleston at gmail.com  Mon Aug 29 19:44:25 2011
From: rhuddleston at gmail.com (Robert Huddleston)
Date: Mon, 29 Aug 2011 11:44:25 -0400
Subject: [Freeswitch-users] CallWithUs
In-Reply-To: <441F1D5786364B62BA13EBD8AB085999@DWP>
References: <005001cc6657$925b32e0$b71198a0$@com>
	<441F1D5786364B62BA13EBD8AB085999@DWP>
Message-ID: <008d01cc6662$88857ea0$99907be0$@com>

Nice J

 

I got it working. I guess my confusion was more with their A2Billing platform/portal and how to get a DID to route without using SIP registration. In their interface I had to configure the DID as SIP forwarding and then add an ACL in Freeswitch.

 

As far as the invalid responses ? I?ve been there a couple of times? It?s only happened to me on exotic destinations like Haiti following the Earthquake.

 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Darcy
Sent: Monday, August 29, 2011 11:24 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] CallWithUs

 

We used callwithus using the freeswitch basically as a tandem.  I believe you need to send them your ip address for them to put in their system, then everything else is just standard config.

Make sure their IP address is in your acl list.

This is what I used in my dial plan.

<action application="bridge" data=mailto:data=sofia/${use_profile}/1$1 at carrier.callwithus.com <mailto:data=sofia/$%7buse_profile%7d/1$1 at carrier.callwithus.com> />

 

They asked us to do the following:

* CALLERID: set to blank

* HOST: Your SIP server IP address

* PORT: Your port (5060 for default port)

  Send calls to carrier.callwithus.com, port 5060. 

 

We stopped using them because to many of their routes kept going down and we did not get proper responses to declined calls, so customers got dead air.

 

Darcy

 

From: Robert Huddleston <mailto:rhuddleston at gmail.com>  

Sent: Monday, August 29, 2011 10:25 AM

To: 'FreeSWITCH Users Help' <mailto:freeswitch-users at lists.freeswitch.org>  

Subject: [Freeswitch-users] CallWithUs

 

I already reviewed the wiki ? does anyone have experience with CallWithUs and using IP Auth / not registration based?

 

  _____  


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

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110829/ab2b29ba/attachment.html 

From msc at freeswitch.org  Mon Aug 29 19:47:51 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 29 Aug 2011 08:47:51 -0700
Subject: [Freeswitch-users] Start Conference with first user
In-Reply-To: <4E57A27C.7070803@ewetel.de>
References: <4E57A27C.7070803@ewetel.de>
Message-ID: <CAKzWOxUj0zQUBGVdcJ2cC2JMwN_FAQbP6+7qzJRv-aCf7wqBUw@mail.gmail.com>

I believe the mintwo flag is not what you want. Instead you probably want
the controls that let you change the enter sound:

conference <conf> enter_sound on|off|none|file <filename>

So when the conference is first created it will use whatever is in the
conference profile for the enter sound and then when the conference
moderator joins you can use the api trick disable or change the enter sound.
Let's say the name of the conference is "conference-3000" and you just want
to turn off the enter sound.  You could do something like this in the
dialplan right before sending the moderator into the conference:

<action application="set" data="api=${conference conference-3000 enter_sound
off}"/>

Hope that helps!

-MC

On Fri, Aug 26, 2011 at 6:41 AM, Mitja Thomas <mitja.thomas1 at ewetel.de>wrote:

> Hello List,
>
> I try to start a conference in mod_conference when the first member
> attempts to join. This may sound weird, but the conference is set to
> wait for the moderator anyhow and I want to use the enter-sound to play
> something like "There is currently no Moderator in this room. Please
> hold the line". If the moderator joins, the enter-sound setting will be
> set to none from dialplan. I thought I just have to NOT set the mintwo
> member flag, but that didnt do the trick.
> When the first member (no moderator) join there is no enter-sound, just
> the moh file. When the second member joins (no moderator) both get the
> enter-sound.
> I thought there was a bug with the mintwo flag and I was close to submit
> to JIRA because of that, but looking further into the code Im not sure
> if the mintwo flag is even supposed to behave the way I want to. It just
> kills the conference when less then two users are in it.
>
> So what I want to know, am I expecting the wrong behaviour from the
> mintwo member flag? If so, is there a different approach to tell the
> conference users right at the beginning if there is a moderator present
> or not.
>
> Regards,
>
> Mitja
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110829/75988a07/attachment.html 

From msc at freeswitch.org  Mon Aug 29 21:06:37 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 29 Aug 2011 10:06:37 -0700
Subject: [Freeswitch-users] Inband DTMF and bind_meta_app problem again
In-Reply-To: <CA+GmxbYZ1PHwf60O6+bqaypfOyMuXMkLTm7=BFC4=nmuW_Ng=g@mail.gmail.com>
References: <CA+GmxbYZ1PHwf60O6+bqaypfOyMuXMkLTm7=BFC4=nmuW_Ng=g@mail.gmail.com>
Message-ID: <CAKzWOxXZeDRui4n2A4THoX8dE7ochZHS0QM0hizRH1n_A-md1Q@mail.gmail.com>

Best thing to do would be to open a jira and do your best to narrow down the
commit where it went from working to non-working. Also, do a test with
latest git to confirm that it hasn't been fixed in the last 10 minutes. Tony
has been known to do that. ;)

-MC

On Mon, Aug 29, 2011 at 12:36 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:

> Hi All
> After recent updates of DTMF negotiation or some associated stuff we can't
> use bind_meta_app with inband DTMF. bind_meta_app ignores bindings, although
> start_dtmf and spandsp_start_dtmf detect DTMF. We use git-fb6e979 2011-08-26
> 04-48-33 +0000.
>
>
> <anti-action application="set" data="dtmf_type=${pbx_call_dtmf_type}"/>
> <anti-action application="spandsp_start_dtmf"/>
> <anti-action application="bind_meta_app" data="7 a s
> execute_extension::att_xfer XML features"/>
>
> EXECUTE sofia/local/test1 at 85.114.2.200 bind_meta_app(7 a s
> execute_extension::att_xfer XML features)
>
> 2011-08-29 11:39:09.371414 [INFO] switch_ivr_async.c:3066 Bound A-Leg: *7
> execute_extension::att_xfer XML features
> ...cut...
>  2011-08-29 11:40:15.751415 [DEBUG] mod_spandsp_dsp.c:56 DTMF BEGIN
> DETECTED: [*]
> 2011-08-29 11:40:15.751415 [DEBUG] switch_ivr_bridge.c:391 Send signal
> sofia/external/89215572714 [BREAK]
> 2011-08-29 11:40:15.831571 [DEBUG] mod_spandsp_dsp.c:68 DTMF END DETECTED:
> [*], duration = 80 ms
> 2011-08-29 11:40:15.951420 [DEBUG] mod_spandsp_dsp.c:56 DTMF BEGIN
> DETECTED: [7]
> 2011-08-29 11:40:15.951420 [DEBUG] switch_ivr_bridge.c:391 Send signal
> sofia/external/89215572714 [BREAK]
> 2011-08-29 11:40:16.051411 [DEBUG] mod_spandsp_dsp.c:68 DTMF END DETECTED:
> [7], duration = 100 ms
> 2011-08-29 11:40:17.391410 [DEBUG] mod_spandsp_dsp.c:56 DTMF BEGIN
> DETECTED: [*]
> 2011-08-29 11:40:17.391410 [DEBUG] switch_ivr_bridge.c:391 Send signal
> sofia/external/89215572714 [BREAK]
> 2011-08-29 11:40:17.471409 [DEBUG] mod_spandsp_dsp.c:68 DTMF END DETECTED:
> [*], duration = 80 ms
> 2011-08-29 11:40:17.571412 [DEBUG] mod_spandsp_dsp.c:56 DTMF BEGIN
> DETECTED: [7]
> 2011-08-29 11:40:17.571412 [DEBUG] switch_ivr_bridge.c:391 Send signal
> sofia/external/89215572714 [BREAK]
> 2011-08-29 11:40:17.631411 [DEBUG] mod_spandsp_dsp.c:68 DTMF END DETECTED:
> [7], duration = 60 ms
>
> But no defined application is launched.
>
>
> --
> Best regards,
>
> Dmitry Sytchev,
> IT Engineer
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110829/6fcd6d29/attachment-0001.html 

From msc at freeswitch.org  Mon Aug 29 21:20:33 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 29 Aug 2011 10:20:33 -0700
Subject: [Freeswitch-users] Dialed number identification service
In-Reply-To: <1314628406963-6737827.post@n2.nabble.com>
References: <1314628406963-6737827.post@n2.nabble.com>
Message-ID: <CAKzWOxWQePmA=HYSQpUo+DT2Ud4wnPRX6WGUwbKQsZ7SYQeXfg@mail.gmail.com>

What kind of phone is this? In many cases the phone will display the "from"
information automatically.

-MC

On Mon, Aug 29, 2011 at 7:33 AM, Aakash <aakashviswam at gmail.com> wrote:

> Hi,
>
> Is this possible - Dialed number identification service in freeswitch.
>
> For eg:
>
> I have configured my two inbound number to the same extension(eg 1000).When
> calling party dials any of the two inbound number should able to show the
> DNIS to the routed extension(eg 1000).
>
>
> Thanks
> Aakash
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Dialed-number-identification-service-tp6737827p6737827.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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110829/6ef78e69/attachment.html 

From xing2kin at yahoo.com  Mon Aug 29 21:24:31 2011
From: xing2kin at yahoo.com (king2kin)
Date: Mon, 29 Aug 2011 10:24:31 -0700 (PDT)
Subject: [Freeswitch-users] How to start IVR script via dialplan by making
	an outbound call
Message-ID: <1314638671.27927.YahooMailClassic@web39701.mail.mud.yahoo.com>

Hi folks,

It's easy to start to run IVR script by an inbound call on FreeSwitch. However, I don't know to start to run IVR script in the following case:

I'd like to run an external client application which will ask FreeSwitch to make an outbound call to a mobile phone number, when the phone rings, someone picks it up and then she/he starts to interact with FreeSwitch normally like an inbount-call initiates to run IVR script.

Could anyone please give me some advice?

Thanks

x.k.


From msc at freeswitch.org  Mon Aug 29 21:29:36 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 29 Aug 2011 10:29:36 -0700
Subject: [Freeswitch-users] Session ends unexpectedly during record dial
 plan usage
In-Reply-To: <4E569AA6.2010008@newpace.ca>
References: <4E569AA6.2010008@newpace.ca>
Message-ID: <CAKzWOxWk_9Janh4ZEjpx2J1NKFcQGf0oy97zZC8w14xku130Wg@mail.gmail.com>

Go ahead and get a console debug log on this along with a SIP trace. Drop it
in pastebin. Hopefully it contains some clues as to what is happening.
-MC

On Thu, Aug 25, 2011 at 11:55 AM, Adam Kelloway <adam.kelloway at newpace.ca>wrote:

> Hi there,
>
> I have a freeswitch installation that I can make sip calls to to listen
> to IVR menus. The sessions last as long as either side does not hang up.
> The exception to this is when I use the 'record' dial plan tool. The sip
> session ends unexpectedly after about 32+ seconds into the recording.
> This happens every time I use the record tool. Note that I have set the
> maximum message length to 120 seconds, so this shouldn't be coming into
> play here (and shouldn't affect the session anyway).
>
> Has anyone ever experienced this, and do you have any suggestions as to
> what might be the cause?
>
> Note that there is no NAT involved here. There are also no Expires or
> Session-Expires header(s) in the sip INVITE or response that would
> affect the length of the session. Indeed, the same type of session can
> continue indefinitely until about 32+ seconds after I invoke the record
> dial plan tool.
>
> Thanks,
>
> Adam
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110829/8b64c41c/attachment.html 

From brad at tech21.com  Mon Aug 29 21:32:41 2011
From: brad at tech21.com (Brad Mina)
Date: Mon, 29 Aug 2011 10:32:41 -0700
Subject: [Freeswitch-users] How to start IVR script via dialplan by
 making an outbound call
In-Reply-To: <1314638671.27927.YahooMailClassic@web39701.mail.mud.yahoo.com>
References: <1314638671.27927.YahooMailClassic@web39701.mail.mud.yahoo.com>
Message-ID: <CAGRr059W+oh8nkAK+p+gim8a32SW=1pN-VYLwtRGD5bH9HKj5w@mail.gmail.com>

If I understand this correctly, you want an automated dialing script which
will spawn an IVR application upon the callee answering?

On Mon, Aug 29, 2011 at 10:24 AM, king2kin <xing2kin at yahoo.com> wrote:

> Hi folks,
>
> It's easy to start to run IVR script by an inbound call on FreeSwitch.
> However, I don't know to start to run IVR script in the following case:
>
> I'd like to run an external client application which will ask FreeSwitch to
> make an outbound call to a mobile phone number, when the phone rings,
> someone picks it up and then she/he starts to interact with FreeSwitch
> normally like an inbount-call initiates to run IVR script.
>
> Could anyone please give me some advice?
>
> Thanks
>
> x.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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110829/38b8d41b/attachment.html 

From aakashviswam at gmail.com  Mon Aug 29 21:33:38 2011
From: aakashviswam at gmail.com (Aakash)
Date: Mon, 29 Aug 2011 10:33:38 -0700 (PDT)
Subject: [Freeswitch-users] Dialed number identification service
In-Reply-To: <CAKzWOxWQePmA=HYSQpUo+DT2Ud4wnPRX6WGUwbKQsZ7SYQeXfg@mail.gmail.com>
References: <1314628406963-6737827.post@n2.nabble.com>
	<CAKzWOxWQePmA=HYSQpUo+DT2Ud4wnPRX6WGUwbKQsZ7SYQeXfg@mail.gmail.com>
Message-ID: <1314639218946-6738612.post@n2.nabble.com>

We are using Snom,Polycom and Aastra.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Dialed-number-identification-service-tp6737827p6738612.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Mon Aug 29 21:33:23 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 29 Aug 2011 10:33:23 -0700
Subject: [Freeswitch-users] Strange problem with Freeswitch and external
	extension
In-Reply-To: <000901cc6368$0c402220$24c06660$@accra.ca>
References: <000901cc6368$0c402220$24c06660$@accra.ca>
Message-ID: <CAKzWOxWgqnmFpdrsfvDBd7K3b20Wm3TvcGVGi6_7oia+LkWRuw@mail.gmail.com>

What kind of phone is it? Just curious. Also, be sure look at the SDP ports
and compare them to the actual ports to which RTP is being sent by pfsense.
It's entirely possible that there's a mismatch somewhere. If you can nail
down what is actually happening vs. what should be happening then it will be
easier to know where to look next. My money is on the phone or the pfsense
router.

-MC

On Thu, Aug 25, 2011 at 1:46 PM, Charles Bujold <cjbujold at accra.ca> wrote:

> I have a remote extension which registers with the Freeswitch server.  The
> user can record his greeting in the Freeswitch IVR without a problem.  Yet
> if the user calls me I cannot hear him, but he can hear me.****
>
> ** **
>
> I thought at first that it was a NAT issue so we placed the remote
> telephone directly on the internet and the same problem remains.  The
> Freeswitch server is behind a NAT/router and I can see the packets from the
> phone coming into the router and being forwarded to the Freeswitch server.
> The router is not blocking the packets and is forwarding them to the correct
> server IP.****
>
> ** **
>
> I them placed wireshark on the Freeswitch server and I can see the packets
> leave the Freeswitch server and going to the phone But I cannot see the
> Freeswitch server receive the packets from the PFsense router, even though I
> can see in the router that the packets are being forwarded properly to the
> Freeswitch server IP.  The Freeswitch server is running Ubuntu with no
> firewall, so in theory I should see the server accept or reject the
> packets.  Yet I do not see any in wireshark coming from the phone, only
> those leaving from the Freeswitch server.****
>
> ** **
>
> What I do not understand is why, since he is registered,  I cannot hear
> him, but yet he can interact with Freeswitch and record a greeting?
> Obviously packets are getting routed properly and Freeswitch recognizes
> them, some of the time.****
>
> ** **
>
> Can somebody suggest a possible explanation and solution to this problem?*
> ***
>
> ** **
>
> Thanks****
>
> ** **
>
> cjb****
>
> ** **
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110829/6ff74a91/attachment.html 

From robert.hadley at teotech.com  Mon Aug 29 21:36:52 2011
From: robert.hadley at teotech.com (Robert Hadley)
Date: Mon, 29 Aug 2011 10:36:52 -0700
Subject: [Freeswitch-users] How to save voicemail msgs to a db
In-Reply-To: <33095823FD21DF429B481B5163264B79511864FB4B@VMBX102.ihostexchange.net>
References: <33095823FD21DF429B481B5163264B79511864FA82@VMBX102.ihostexchange.net>
	<E826C94A3C6BE44198FAF4D66B5AB4D81BDC435DF9@VA3DIAXVS351.RED001.local>
	<33095823FD21DF429B481B5163264B79511864FB4B@VMBX102.ihostexchange.net>
Message-ID: <E826C94A3C6BE44198FAF4D66B5AB4D81BDD817201@VA3DIAXVS351.RED001.local>


No, the path to the file is saved in the DB, but the file is stored in a folder.
There is a default storage folder or you can customize it via this parameter:
      <!--<param name="storage-dir" value="/tmp"/>-->

Robert

-----Original Message-----
From: Yungwei Chen [mailto:yungwei at resolvity.com] 
Sent: Monday, August 29, 2011 6:54 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] How to save voicemail msgs to a db

Thanks for your feedback.
Can the voicemail msgs (*.wav) be saved to the db as well?

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Robert Hadley
Sent: Friday, August 26, 2011 6:42 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] How to save voicemail msgs to a db

Hi,

If you are using odbc to connect to your database, http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core

Then tell mod_voicemail about the DB via this param in autoload_configs/voicemail.conf.xml:

      <param name="odbc-dsn" value="dsn:user:pass"/>

Regards,
Robert


-----Original Message-----
From: Yungwei Chen 
Sent: Friday, August 26, 2011 1:30 PM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] How to save voicemail msgs to a db

Hi,

I'm wondering if there's a way to save voicemail messages to a (possibly remote) database. If so, how can that be achieved? 
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 msc at freeswitch.org  Mon Aug 29 21:59:16 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 29 Aug 2011 10:59:16 -0700
Subject: [Freeswitch-users] Multiple Tone Detect
In-Reply-To: <201108281613.18871.justlikeef@gmail.com>
References: <201108281613.18871.justlikeef@gmail.com>
Message-ID: <CAKzWOxXGmU2G4Eqz4FX2tV223Cb4dW1zSg_-qnnaLA4kEcUngA@mail.gmail.com>

If you need multiple tone detects then definitely use the first method. I've
done as many as 6 tone_detects on a single call and it works well.
-MC

On Sun, Aug 28, 2011 at 1:13 PM, Rob Hutton <justlikeef at gmail.com> wrote:

> I am trying to get tone detection set up so that a single number can be
> used for both voice and fax.  Tone detection needs to run concurrent to the
> normal call processing, so that a call proceeds normally unless the fax
> tones are heard.
>
> On this wiki page (
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_fax_detect) it gives
> the format:
>
> <action application="tone_detect" data="fax 1100 r +5000 transfer fax XML
> default" />
>
> and an associated dialplan entry for the call to be transfered to that
> actually handles the fax reception.
>
> On this wiki page (
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect) it says
> that the format from the first page is not the preferred method, and that
> the format should be something like:
>
> <action application="set" data="execute_on_tone_detect=transfer fax XML
> default"/>
> <action application="set" data="tone_detect_hits=1"/>
> <action application="set" data="execute_on_media=tone_detect mytone 1100 r
> +5000"/>
>
> My question is, using the first format, I could set up multiple detections,
> eg. one for fax tones that sends/receives a fax, one for answering machines
> tone that hangs up, etc.
>
> Is this possible with the preferred syntax?
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110829/d68e713e/attachment.html 

From anthony.minessale at gmail.com  Mon Aug 29 22:13:41 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 29 Aug 2011 13:13:41 -0500
Subject: [Freeswitch-users] Multiple Tone Detect
In-Reply-To: <CAKzWOxXGmU2G4Eqz4FX2tV223Cb4dW1zSg_-qnnaLA4kEcUngA@mail.gmail.com>
References: <201108281613.18871.justlikeef@gmail.com>
	<CAKzWOxXGmU2G4Eqz4FX2tV223Cb4dW1zSg_-qnnaLA4kEcUngA@mail.gmail.com>
Message-ID: <CAKbxfG-UHXsRstwp0Ut3zz-BQDxumyfZXZSCLWoRBNqAt=-qvQ@mail.gmail.com>

there is a specific app for fax detecting inside mod spandsp called
spandsp_start_fax_detect


it takes about the same input args namely <app> [<arg>][ <timeout>]
It uses the official fax identification code inside spandsp and works
much better than the single tone detect way.



On Mon, Aug 29, 2011 at 12:59 PM, Michael Collins <msc at freeswitch.org> wrote:
> If you need multiple tone detects then definitely use the first method. I've
> done as many as 6 tone_detects on a single call and it works well.
> -MC
>
> On Sun, Aug 28, 2011 at 1:13 PM, Rob Hutton <justlikeef at gmail.com> wrote:
>>
>> I am trying to get tone detection set up so that a single number can be
>> used for both voice and fax. ?Tone detection needs to run concurrent to the
>> normal call processing, so that a call proceeds normally unless the fax
>> tones are heard.
>>
>> On this wiki page
>> (http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_fax_detect) it gives
>> the format:
>>
>> <action application="tone_detect" data="fax 1100 r +5000 transfer fax XML
>> default" />
>>
>> and an associated dialplan entry for the call to be transfered to that
>> actually handles the fax reception.
>>
>> On this wiki page
>> (http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect) it says
>> that the format from the first page is not the preferred method, and that
>> the format should be something like:
>>
>> <action application="set" data="execute_on_tone_detect=transfer fax XML
>> default"/>
>> <action application="set" data="tone_detect_hits=1"/>
>> <action application="set" data="execute_on_media=tone_detect mytone 1100 r
>> +5000"/>
>>
>> My question is, using the first format, I could set up multiple
>> detections, eg. one for fax tones that sends/receives a fax, one for
>> answering machines tone that hangs up, etc.
>>
>> Is this possible with the preferred syntax?
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From anthony.minessale at gmail.com  Mon Aug 29 22:15:33 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 29 Aug 2011 13:15:33 -0500
Subject: [Freeswitch-users] Error compling jsxml.c
In-Reply-To: <4E5AE0B9.7040307@ihug.co.nz>
References: <4E5AE0B9.7040307@ihug.co.nz>
Message-ID: <CAKbxfG9coLLTPH2AXinETtKSGQbcqco17vuNZHPK6p-tWt8pfA@mail.gmail.com>

looks like a bad merge?

git checkout HEAD ./libs/js/src/jsxml.c

?


On Sun, Aug 28, 2011 at 7:43 PM, D Foulkes <dfoulkes at ihug.co.nz> wrote:
> Hi, I am still fairly new to freeswitch...
>
> Having problems with a setup on a Ubuntu 10.10, so I decided to do a
> rebuild from scratch. Following recommended instructions on the wiki
> installation guide (using git clone) I get the following error when
> running the compile:
>
> make && make install
> ...
>
> libtool: compile: gcc -DOSSP -DXP_UNIX -DEXPORT_JS_API
> -DJS_HAS_FILE_OBJECT=1 -DJS_HAS_XML_SUPPORT=1 -DJS_THREADSAFE=1
> -Insprpub/dist/include/nspr -Insprpub/pr/include -DHAVE_CONFIG_H -Isrc
> -O2 -DNDEBUG -pipe -c src/jsxml.c -fPIC -DPIC -o src/.libs/jsxml.o
> src/jsxml.c: In function ?PutProperty?:
> src/jsxml.c:4571: error: expected expression before ?>? token
> src/jsxml.c:4571: error: stray ?\37? in program
> src/jsxml.c:4571: error: stray ?`? in program
> src/jsxml.c:4572: error: stray ?\37? in program
> src/jsxml.c:4572: error: stray ?\37? in program
> src/jsxml.c:4572: error: stray ?\37? in program
> src/jsxml.c:4572: error: stray ?\37? in program
> src/jsxml.c:4572: error: stray ?\37? in program
> src/jsxml.c:4573: error: stray ?\37? in program
> src/jsxml.c:4573: error: stray ?\37? in program
> src/jsxml.c:4587: error: ?JSXML? has no member named ?xml?
> src/jsxml.c:4587: error: stray ?`? in program
> src/jsxml.c:4587: error: expected ?;? before ?value?
> src/jsxml.c:4588: error: expected expression before ?goto?
> src/jsxml.c:4591: error: expected expression before ?/? token
> src/jsxml.c:4592: error: expected statement before ?)? token
> src/jsxml.c:4592: warning: missing terminating ' character
> src/jsxml.c:4592: error: missing terminating ' character
> src/jsxml.c:4593: error: ?JSXMLQName? has no member named ?localOame?
> src/jsxml.c:4593: error: expected expression before ?)? token
> make[5]: *** [src/jsxml.lo] Error 1
> make[4]: *** [/usr/local/src/freeswitch/libs/js/libjs.la] Error 2
> make[3]: *** [mod_spidermonkey-all] Error 1
> make[2]: *** [all-recursive] Error 1
> make[1]: *** [all-recursive] Error 1
> make: *** [all] Error 2
>
> Any help would be appreciated. Thanks,
>
> Daniel
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From msc at freeswitch.org  Mon Aug 29 22:16:44 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 29 Aug 2011 11:16:44 -0700
Subject: [Freeswitch-users] freeswitch php
In-Reply-To: <1314462398.75774.YahooMailNeo@web161009.mail.bf1.yahoo.com>
References: <1313970987.96106.YahooMailNeo@web161008.mail.bf1.yahoo.com>
	<CAKzWOxWTGEYrAwHz20QO0JAYSk7oc9DyLOkVXiJ1=u8v4AAkpA@mail.gmail.com>
	<1314060883.62351.YahooMailNeo@web161008.mail.bf1.yahoo.com>
	<CAKzWOxUNFYNRPEknF0uF5gKUZBzfTzOr4R2gCS10SdoQ1fSP8A@mail.gmail.com>
	<1314462398.75774.YahooMailNeo@web161009.mail.bf1.yahoo.com>
Message-ID: <CAKzWOxVyCAFwsuxwDwnNGWvOwxapqktX3sH+wY3KNA1nhprcOA@mail.gmail.com>

Sam,

I got this to work without any issues. I think there may be two things for
you to check:

#1 - make sure that you set your script to be executable
#2 - make sure that you have php-cli installed

Other than that, as long as fs_ivrd is running then it should work. For the
record, I copied the wiki exactly except I used port 9090 instead of 8084.

-MC

On Sat, Aug 27, 2011 at 9:26 AM, Sam <lakersman2006 at yahoo.com> wrote:

> MC,
>
> Any luck with fs_ivrd with ESL?
>
> ------------------------------
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Tuesday, August 23, 2011 10:06 AM
>
> *Subject:* Re: [Freeswitch-users] freeswitch php
>
> Aha! fs_ivrd. I haven't tinkered w/ fs_ivrd outside of Perl, but it should
> work with anything that can read/write STDIN/STDOUT. Give me a few hours to
> work on some day job items and then I'll hop in here and take a look. Maybe
> you can help me wikify some of this knowledge. ;)
>
> -MC
>
> P.S. - If anyone else has or is using fs_ivrd and has some experience with
> it please let me know. I'd like to borrow some of your mental cycles so that
> we can document this better.
>
> On Mon, Aug 22, 2011 at 5:54 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> OK so if I were to use the ESL lib with PHP, can it still be used with the
> fs_ivrd daemon? I have not seen any ESL examples that is used with the
> fs_ivrd deamon, so I am not exactly to use the ESL lib.
>
> ------------------------------
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Monday, August 22, 2011 5:27 PM
> *Subject:* Re: [Freeswitch-users] freeswitch php
>
> This probably isn't the most efficient way of using the event socket. In
> fact, I'm not sure if this is anything more than a proof of concept. You are
> MUCH better off using the ESL lib with PHP and taking advantage of the
> abstraction you get.
>
> -MC
>
> On Sun, Aug 21, 2011 at 4:56 PM, Sam <lakersman2006 at yahoo.com> wrote:
>
> Does anyone know how to retrieve channel variables (ie. uuid, etc.) using
> the php example that was shown in the wiki below?
>
> #!/usr/bin/php -q
>
> <?php
>
> // set a couple of things so we dont kill the system
> ob_implicit_flush(true);
> set_time_limit(30);
>
> // Open stdin so we can read the data in
> $in = fopen("php://stdin", "r");
>
> // Connect
> echo "connect\n\n";
>
> // Answer
> echo "sendmsg\n";
> echo "call-command: execute\n";
> echo "execute-app-name: answer\n\n";
>
> // Play a prompt
> echo "sendmsg\n";
> echo "call-command: execute\n";
> echo "execute-app-name: playback\n";
> echo "execute-app-arg: /usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav\n\n";
>
> // Wait
> sleep(5);
>
> // Hangup
> echo "sendmsg\n";
> echo "call-command: hangup\n\n";
>
> fclose($in);
>
> ?>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110829/28205eed/attachment-0001.html 

From dfoulkes at ihug.co.nz  Mon Aug 29 23:11:28 2011
From: dfoulkes at ihug.co.nz (daniel)
Date: Mon, 29 Aug 2011 12:11:28 -0700 (PDT)
Subject: [Freeswitch-users] Error compling jsxml.c
In-Reply-To: <CAKbxfG9coLLTPH2AXinETtKSGQbcqco17vuNZHPK6p-tWt8pfA@mail.gmail.com>
References: <4E5AE0B9.7040307@ihug.co.nz>
	<CAKbxfG9coLLTPH2AXinETtKSGQbcqco17vuNZHPK6p-tWt8pfA@mail.gmail.com>
Message-ID: <1314645088523-6739059.post@n2.nabble.com>

Thanks.  That did seem to be the problem.  I fixed it (before your post) by
deleting the GIT repository and doing another git clone...   Should have
tried that earlier but thought GIT did thorough error checking and so
couldn't see how a file would be corrupted.



Anthony Minessale wrote:
> 
> looks like a bad merge?
> 
> git checkout HEAD ./libs/js/src/jsxml.c
> 
> ?
> 
> 
> On Sun, Aug 28, 2011 at 7:43 PM, D Foulkes &lt;dfoulkes at ihug.co.nz&gt;
> wrote:
>> Hi, I am still fairly new to freeswitch...
>>
>> Having problems with a setup on a Ubuntu 10.10, so I decided to do a
>> rebuild from scratch. Following recommended instructions on the wiki
>> installation guide (using git clone) I get the following error when
>> running the compile:
>>
>> make && make install
>> ...
>>
>> libtool: compile: gcc -DOSSP -DXP_UNIX -DEXPORT_JS_API
>> -DJS_HAS_FILE_OBJECT=1 -DJS_HAS_XML_SUPPORT=1 -DJS_THREADSAFE=1
>> -Insprpub/dist/include/nspr -Insprpub/pr/include -DHAVE_CONFIG_H -Isrc
>> -O2 -DNDEBUG -pipe -c src/jsxml.c -fPIC -DPIC -o src/.libs/jsxml.o
>> src/jsxml.c: In function ?PutProperty?:
>> src/jsxml.c:4571: error: expected expression before ?>? token
>> src/jsxml.c:4571: error: stray ?\37? in program
>> src/jsxml.c:4571: error: stray ?`? in program
>> src/jsxml.c:4572: error: stray ?\37? in program
>> src/jsxml.c:4572: error: stray ?\37? in program
>> src/jsxml.c:4572: error: stray ?\37? in program
>> src/jsxml.c:4572: error: stray ?\37? in program
>> src/jsxml.c:4572: error: stray ?\37? in program
>> src/jsxml.c:4573: error: stray ?\37? in program
>> src/jsxml.c:4573: error: stray ?\37? in program
>> src/jsxml.c:4587: error: ?JSXML? has no member named ?xml?
>> src/jsxml.c:4587: error: stray ?`? in program
>> src/jsxml.c:4587: error: expected ?;? before ?value?
>> src/jsxml.c:4588: error: expected expression before ?goto?
>> src/jsxml.c:4591: error: expected expression before ?/? token
>> src/jsxml.c:4592: error: expected statement before ?)? token
>> src/jsxml.c:4592: warning: missing terminating ' character
>> src/jsxml.c:4592: error: missing terminating ' character
>> src/jsxml.c:4593: error: ?JSXMLQName? has no member named ?localOame?
>> src/jsxml.c:4593: error: expected expression before ?)? token
>> make[5]: *** [src/jsxml.lo] Error 1
>> make[4]: *** [/usr/local/src/freeswitch/libs/js/libjs.la] Error 2
>> make[3]: *** [mod_spidermonkey-all] Error 1
>> make[2]: *** [all-recursive] Error 1
>> make[1]: *** [all-recursive] Error 1
>> make: *** [all] Error 2
>>
>> Any help would be appreciated. Thanks,
>>
>> Daniel
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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://freeswitch-users.2379917.n2.nabble.com/Error-compling-jsxml-c-tp6736056p6739059.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From justlikeef at gmail.com  Mon Aug 29 23:36:26 2011
From: justlikeef at gmail.com (Rob Hutton)
Date: Mon, 29 Aug 2011 15:36:26 -0400
Subject: [Freeswitch-users] Dialed number identification service
In-Reply-To: <CAKzWOxWQePmA=HYSQpUo+DT2Ud4wnPRX6WGUwbKQsZ7SYQeXfg@mail.gmail.com>
References: <1314628406963-6737827.post@n2.nabble.com>
	<CAKzWOxWQePmA=HYSQpUo+DT2Ud4wnPRX6WGUwbKQsZ7SYQeXfg@mail.gmail.com>
Message-ID: <201108291536.26499.justlikeef@gmail.com>

I beleive he is asking for the original disposition to be displayed on the phone.  IOW, if you have 5551212 and 5552222 assigned to the same internal user/device, it will show the number that the remote user dialed, the 555xxxx.

Or whatever the DNIS digits are...
 
On Monday 29 August 2011 13:20:33 Michael Collins wrote:
> What kind of phone is this? In many cases the phone will display the "from"
> information automatically.
> 
> -MC
> 
> On Mon, Aug 29, 2011 at 7:33 AM, Aakash <aakashviswam at gmail.com> wrote:
> 
> > Hi,
> >
> > Is this possible - Dialed number identification service in freeswitch.
> >
> > For eg:
> >
> > I have configured my two inbound number to the same extension(eg 1000).When
> > calling party dials any of the two inbound number should able to show the
> > DNIS to the routed extension(eg 1000).
> >
> >
> > Thanks
> > Aakash
> >
> >
> >
> > --
> > View this message in context:
> > http://freeswitch-users.2379917.n2.nabble.com/Dialed-number-identification-service-tp6737827p6737827.html
> > Sent from the freeswitch-users mailing list archive at Nabble.com.
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 


From aakashviswam at gmail.com  Mon Aug 29 23:52:40 2011
From: aakashviswam at gmail.com (Aakash)
Date: Mon, 29 Aug 2011 12:52:40 -0700 (PDT)
Subject: [Freeswitch-users] Dialed number identification service
In-Reply-To: <201108291536.26499.justlikeef@gmail.com>
References: <1314628406963-6737827.post@n2.nabble.com>
	<CAKzWOxWQePmA=HYSQpUo+DT2Ud4wnPRX6WGUwbKQsZ7SYQeXfg@mail.gmail.com>
	<201108291536.26499.justlikeef@gmail.com>
Message-ID: <1314647560536-6739256.post@n2.nabble.com>

Yes Rob...You are right..Just i need to display the DNIS digits in the sip
phone.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Dialed-number-identification-service-tp6737827p6739256.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Tue Aug 30 00:38:17 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 29 Aug 2011 15:38:17 -0500
Subject: [Freeswitch-users] Session ends unexpectedly during record dial
 plan usage
In-Reply-To: <CAKzWOxWk_9Janh4ZEjpx2J1NKFcQGf0oy97zZC8w14xku130Wg@mail.gmail.com>
References: <4E569AA6.2010008@newpace.ca>
	<CAKzWOxWk_9Janh4ZEjpx2J1NKFcQGf0oy97zZC8w14xku130Wg@mail.gmail.com>
Message-ID: <CAKbxfG-Krk-MKXZZdTcTJEyf628siaMvGHTy1PQ_nC+1DnYoKQ@mail.gmail.com>

/me punches MSC in the arm.....

On Mon, Aug 29, 2011 at 12:29 PM, Michael Collins <msc at freeswitch.org> wrote:
> Go ahead and get a console debug log on this along with a SIP trace. Drop it
> in pastebin. Hopefully it contains some clues as to what is happening.
> -MC
>
> On Thu, Aug 25, 2011 at 11:55 AM, Adam Kelloway <adam.kelloway at newpace.ca>
> wrote:
>>
>> Hi there,
>>
>> I have a freeswitch installation that I can make sip calls to to listen
>> to IVR menus. The sessions last as long as either side does not hang up.
>> The exception to this is when I use the 'record' dial plan tool. The sip
>> session ends unexpectedly after about 32+ seconds into the recording.
>> This happens every time I use the record tool. Note that I have set the
>> maximum message length to 120 seconds, so this shouldn't be coming into
>> play here (and shouldn't affect the session anyway).
>>
>> Has anyone ever experienced this, and do you have any suggestions as to
>> what might be the cause?
>>
>> Note that there is no NAT involved here. There are also no Expires or
>> Session-Expires header(s) in the sip INVITE or response that would
>> affect the length of the session. Indeed, the same type of session can
>> continue indefinitely until about 32+ seconds after I invoke the record
>> dial plan tool.
>>
>> Thanks,
>>
>> Adam
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From justlikeef at gmail.com  Tue Aug 30 00:43:08 2011
From: justlikeef at gmail.com (Rob Hutton)
Date: Mon, 29 Aug 2011 16:43:08 -0400
Subject: [Freeswitch-users] Dialed number identification service
In-Reply-To: <1314647560536-6739256.post@n2.nabble.com>
References: <1314628406963-6737827.post@n2.nabble.com>
	<201108291536.26499.justlikeef@gmail.com>
	<1314647560536-6739256.post@n2.nabble.com>
Message-ID: <201108291643.09154.justlikeef@gmail.com>

Most of the phones I have seen have some sort of header that you can add via the dialplan that will allow you to display whatever you want.  You probably want to set it to something like ${destination_number}.

See the Custom Channel Variables on http://wiki.freeswitch.org/wiki/Channel_Variables and I think it will lead you in the righ direction.

On Monday 29 August 2011 15:52:40 Aakash wrote:
> Yes Rob...You are right..Just i need to display the DNIS digits in the sip
> phone.
> 
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Dialed-number-identification-service-tp6737827p6739256.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


From anthony.minessale at gmail.com  Tue Aug 30 00:48:32 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 29 Aug 2011 15:48:32 -0500
Subject: [Freeswitch-users] Condition based on custom sip header
In-Reply-To: <F5B5EF8F78C0BC4BB417E1DBB9D7F7D0180A75B7CD@jjex01.jajah.dublin>
References: <F5B5EF8F78C0BC4BB417E1DBB9D7F7D0180A75B7CD@jjex01.jajah.dublin>
Message-ID: <CAKbxfG-pD89e2p+JtYi__kngCt1azbNkCvmnHDUsLGfjbuB-qA@mail.gmail.com>

if its custom anyway and all you need to do is add X- then what's the
problem?  Using X- is *nearly* a requirement in email/http/sip That's
the whole point of them having a prefix....

On Mon, Aug 29, 2011 at 10:36 AM, Alex Massover <alex at jajah.com> wrote:
> Hi,
>
>
>
> I?m trying to implement dialplan condition based on custom SIP header
> (?Header1: value1?). As far as I understood there?s no way to read custom
> SIP header which isn?t X- header with sip_h_. Is that correct?
>
>
>
> What are the other options, please? Is lua?s getHeader() suitable for this?
>
>
>
> --
>
> Best Regards,
>
> Alex Massover
>
>
>
> This mail was sent via Mail-SeCure System.
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From ijurado at econcept.es  Tue Aug 30 01:31:22 2011
From: ijurado at econcept.es (Isaac Jurado)
Date: Mon, 29 Aug 2011 23:31:22 +0200
Subject: [Freeswitch-users] How to start IVR script via dialplan by
 making an outbound call
In-Reply-To: <1314638671.27927.YahooMailClassic@web39701.mail.mud.yahoo.com>
References: <1314638671.27927.YahooMailClassic@web39701.mail.mud.yahoo.com>
Message-ID: <CABj_V-mFDfiQR+C=aK3wS_VvPF0KugM9wMA0-7sOU2JuW=ZSYw@mail.gmail.com>

On Mon, Aug 29, 2011 at 7:24 PM, king2kin <xing2kin at yahoo.com> wrote:
> Hi folks,
>
> It's easy to start to run IVR script by an inbound call on FreeSwitch.
> However, I don't know to start to run IVR script in the following
> case:
>
> I'd like to run an external client application which will ask
> FreeSwitch to make an outbound call to a mobile phone number, when the
> phone rings, someone picks it up and then she/he starts to interact
> with FreeSwitch normally like an inbount-call initiates to run IVR
> script.
>
> Could anyone please give me some advice?

I think that, from the FreeSWITCH console you can do something like:

originate path/to/mobile_num &bridge(path/to/special_extension)

And configure "path/to/special_extension" to execute the "ivr" dialplan
application.

Cheers.

-- 
Isaac Jurado
Internet Busines Solutions eConcept


From ijurado at econcept.es  Tue Aug 30 01:35:32 2011
From: ijurado at econcept.es (Isaac Jurado)
Date: Mon, 29 Aug 2011 23:35:32 +0200
Subject: [Freeswitch-users] How to save voicemail msgs to a db
In-Reply-To: <E826C94A3C6BE44198FAF4D66B5AB4D81BDD817201@VA3DIAXVS351.RED001.local>
References: <33095823FD21DF429B481B5163264B79511864FA82@VMBX102.ihostexchange.net>
	<E826C94A3C6BE44198FAF4D66B5AB4D81BDC435DF9@VA3DIAXVS351.RED001.local>
	<33095823FD21DF429B481B5163264B79511864FB4B@VMBX102.ihostexchange.net>
	<E826C94A3C6BE44198FAF4D66B5AB4D81BDD817201@VA3DIAXVS351.RED001.local>
Message-ID: <CABj_V-kNRbtyJZM_frOgxxG2_w58WPX3W+RijSMycHAEkEwsmQ@mail.gmail.com>

On Mon, Aug 29, 2011 at 7:36 PM, Robert Hadley
<robert.hadley at teotech.com> wrote:
>
>> Can the voicemail msgs (*.wav) be saved to the db as well?
>
> No, the path to the file is saved in the DB, but the file is stored in
> a folder.  There is a default storage folder or you can customize it
> via this parameter:
> ? ? ?<!--<param name="storage-dir" value="/tmp"/>-->

For the record.  We intercept the storage-dir with a small FUSE driver
so we can store voicemails in a MongoDB/GridFS collection.

Cheers.

-- 
Isaac Jurado
Internet Busines Solutions eConcept


From jmesquita at freeswitch.org  Tue Aug 30 02:07:04 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 29 Aug 2011 19:07:04 -0300
Subject: [Freeswitch-users] How to save voicemail msgs to a db
In-Reply-To: <CABj_V-kNRbtyJZM_frOgxxG2_w58WPX3W+RijSMycHAEkEwsmQ@mail.gmail.com>
References: <33095823FD21DF429B481B5163264B79511864FA82@VMBX102.ihostexchange.net>
	<E826C94A3C6BE44198FAF4D66B5AB4D81BDC435DF9@VA3DIAXVS351.RED001.local>
	<33095823FD21DF429B481B5163264B79511864FB4B@VMBX102.ihostexchange.net>
	<E826C94A3C6BE44198FAF4D66B5AB4D81BDD817201@VA3DIAXVS351.RED001.local>
	<CABj_V-kNRbtyJZM_frOgxxG2_w58WPX3W+RijSMycHAEkEwsmQ@mail.gmail.com>
Message-ID: <CABH8QX2s=18Yfd81rCOjEMRSNGVoWqAmwViXj69pWry4_WetaQ@mail.gmail.com>

I would love to see a mongodb/gridfs driver for FS instead.. :-) Maybe
someday I will make it happen

Regards,
Jo?o Mesquita



On Mon, Aug 29, 2011 at 6:35 PM, Isaac Jurado <ijurado at econcept.es> wrote:

> On Mon, Aug 29, 2011 at 7:36 PM, Robert Hadley
> <robert.hadley at teotech.com> wrote:
> >
> >> Can the voicemail msgs (*.wav) be saved to the db as well?
> >
> > No, the path to the file is saved in the DB, but the file is stored in
> > a folder.  There is a default storage folder or you can customize it
> > via this parameter:
> >      <!--<param name="storage-dir" value="/tmp"/>-->
>
> For the record.  We intercept the storage-dir with a small FUSE driver
> so we can store voicemails in a MongoDB/GridFS collection.
>
> Cheers.
>
> --
> Isaac Jurado
> Internet Busines Solutions eConcept
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110829/2f5da0b4/attachment-0001.html 

From xing2kin at yahoo.com  Tue Aug 30 03:32:03 2011
From: xing2kin at yahoo.com (king2kin)
Date: Mon, 29 Aug 2011 16:32:03 -0700 (PDT)
Subject: [Freeswitch-users] How to start IVR script via dialplan by
	making an outbound call
In-Reply-To: <CAGRr059W+oh8nkAK+p+gim8a32SW=1pN-VYLwtRGD5bH9HKj5w@mail.gmail.com>
Message-ID: <1314660723.81838.YahooMailClassic@web39705.mail.mud.yahoo.com>

yes, your understanding is right.

--- On Mon, 8/29/11, Brad Mina <brad at tech21.com> wrote:


From: Brad Mina <brad at tech21.com>
Subject: Re: [Freeswitch-users] How to start IVR script via dialplan by making an outbound call
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Monday, August 29, 2011, 10:32 AM


If I understand this correctly, you want an automated dialing script which will spawn an IVR application upon the callee answering?


On Mon, Aug 29, 2011 at 10:24 AM, king2kin <xing2kin at yahoo.com> wrote:

Hi folks,

It's easy to start to run IVR script by an inbound call on FreeSwitch. However, I don't know to start to run IVR script in the following case:

I'd like to run an external client application which will ask FreeSwitch to make an outbound call to a mobile phone number, when the phone rings, someone picks it up and then she/he starts to interact with FreeSwitch normally like an inbount-call initiates to run IVR script.

Could anyone please give me some advice?

Thanks

x.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


-----Inline Attachment Follows-----



FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110829/22cf61a2/attachment.html 

From gabe at gundy.org  Tue Aug 30 07:06:48 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Mon, 29 Aug 2011 21:06:48 -0600
Subject: [Freeswitch-users] Error compling jsxml.c
In-Reply-To: <4E5AE0B9.7040307@ihug.co.nz>
References: <4E5AE0B9.7040307@ihug.co.nz>
Message-ID: <CAMwZ-tLcjBoApF9AT5Vf2v4OpLK0=89rZE=ViCmnLfQ1nVHS-w@mail.gmail.com>

On Sun, Aug 28, 2011 at 6:43 PM, D Foulkes <dfoulkes at ihug.co.nz> wrote:
> Having problems with a setup on a Ubuntu 10.10, so I decided to do a
> rebuild from scratch.

If you decide to build debs (makes like pretty easy), this is what I
used for 10.10:
http://gundy.org/post/2547859794/building-freeswitch-on-ubuntu-10-10

It's been a while since I've tested it on that version, but as far as
I remember, the build works.

Best,
Gabe


From mitja.thomas1 at ewetel.de  Tue Aug 30 10:44:36 2011
From: mitja.thomas1 at ewetel.de (Mitja Thomas)
Date: Tue, 30 Aug 2011 08:44:36 +0200
Subject: [Freeswitch-users] Start Conference with first user
In-Reply-To: <CAKzWOxUj0zQUBGVdcJ2cC2JMwN_FAQbP6+7qzJRv-aCf7wqBUw@mail.gmail.com>
References: <4E57A27C.7070803@ewetel.de>
	<CAKzWOxUj0zQUBGVdcJ2cC2JMwN_FAQbP6+7qzJRv-aCf7wqBUw@mail.gmail.com>
Message-ID: <4E5C86D4.30509@ewetel.de>

Hi Michael,

what you suggest is exactly what I was trying to do. Actually it does 
work: I can join with a couple of non_moderators and get the enter_sound 
as soon as the second one joins. when the moderator joins the 
enter_sound is turned off.

But my initial Problem remains. When just one participant joins there is 
no enter_sound because the conference did not start yet. So if someone 
joins he waits and waits and all of a sudden (when the second one joins) 
the sound gets played. So what I was looking for was a way to start the 
conference, when the first participant joins.

I managed to get the sound working the way I want. Ignoring the 
enter_sound and just playing the sound from dialplan after checking if a 
moderator is present in the given conference room.

Still I think that the other way might be interessting not just for me.

Thanks for your time MC, answering every stupid question the list spits 
out :)

Mitja


> I believe the mintwo flag is not what you want. Instead you probably 
> want the controls that let you change the enter sound:
>
> conference <conf> enter_sound on|off|none|file <filename>
>
> So when the conference is first created it will use whatever is in the 
> conference profile for the enter sound and then when the conference 
> moderator joins you can use the api trick disable or change the enter 
> sound. Let's say the name of the conference is "conference-3000" and 
> you just want to turn off the enter sound.  You could do something 
> like this in the dialplan right before sending the moderator into the 
> conference:
>
> <action application="set" data="api=${conference conference-3000 
> enter_sound off}"/>
>
> Hope that helps!
>
> -MC
>
> On Fri, Aug 26, 2011 at 6:41 AM, Mitja Thomas <mitja.thomas1 at ewetel.de 
> <mailto:mitja.thomas1 at ewetel.de>> wrote:
>
>     Hello List,
>
>     I try to start a conference in mod_conference when the first member
>     attempts to join. This may sound weird, but the conference is set to
>     wait for the moderator anyhow and I want to use the enter-sound to
>     play
>     something like "There is currently no Moderator in this room. Please
>     hold the line". If the moderator joins, the enter-sound setting
>     will be
>     set to none from dialplan. I thought I just have to NOT set the mintwo
>     member flag, but that didnt do the trick.
>     When the first member (no moderator) join there is no enter-sound,
>     just
>     the moh file. When the second member joins (no moderator) both get the
>     enter-sound.
>     I thought there was a bug with the mintwo flag and I was close to
>     submit
>     to JIRA because of that, but looking further into the code Im not sure
>     if the mintwo flag is even supposed to behave the way I want to.
>     It just
>     kills the conference when less then two users are in it.
>
>     So what I want to know, am I expecting the wrong behaviour from the
>     mintwo member flag? If so, is there a different approach to tell the
>     conference users right at the beginning if there is a moderator
>     present
>     or not.
>
>     Regards,
>
>     Mitja
>
>
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>


-- 
Mitja Thomas
Vertrieb Gesch?ftskunden

Branchenl?sungen / Service Entwicklung

Telefon: 	+49 (0) 441 -  8000-4916
E-Mail: 	mitja.thomas at ewe.de


EWE TEL GmbH
Cloppenburger Stra?e 310
26133 Oldenburg
E-Mail: info at ewe.de
Internet: www.ewe.de

Handelsregister Amtsgericht Oldenburg HRB 3723
Vorsitzender des Aufsichtsrates: Dr. Werner Brinker
Gesch?ftsf?hrung: Konrad Meier (Vorsitzender), Dirk Brameier, Ulf Heggenberger, Norbert Westfal

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110830/75d29e62/attachment.html 

From msc at freeswitch.org  Tue Aug 30 12:39:08 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 30 Aug 2011 01:39:08 -0700
Subject: [Freeswitch-users] Session ends unexpectedly during record dial
 plan usage
In-Reply-To: <CAKbxfG-Krk-MKXZZdTcTJEyf628siaMvGHTy1PQ_nC+1DnYoKQ@mail.gmail.com>
References: <4E569AA6.2010008@newpace.ca>
	<CAKzWOxWk_9Janh4ZEjpx2J1NKFcQGf0oy97zZC8w14xku130Wg@mail.gmail.com>
	<CAKbxfG-Krk-MKXZZdTcTJEyf628siaMvGHTy1PQ_nC+1DnYoKQ@mail.gmail.com>
Message-ID: <CAKzWOxX0LHJVZqeHn8z2QfypvXifLUMVuTYnT0Tgp3OGBd0q3w@mail.gmail.com>

On Monday, August 29, 2011, Anthony Minessale <anthony.minessale at gmail.com>
wrote:
> /me punches MSC in the arm.....

Haha, I deserved that one. Might wanna smack me with the ClueBat (tm) as
well.

Adam,
If you simply want to record the call between two parties then look on the
wiki for the record_session app. Look at the diff between it and the record
app. They are *totally* different concepts.

Let us know if you have any other questions.

-MC
>
> On Mon, Aug 29, 2011 at 12:29 PM, Michael Collins <msc at freeswitch.org>
wrote:
>> Go ahead and get a console debug log on this along with a SIP trace. Drop
it
>> in pastebin. Hopefully it contains some clues as to what is happening.
>> -MC
>>
>> On Thu, Aug 25, 2011 at 11:55 AM, Adam Kelloway <adam.kelloway at newpace.ca
>
>> wrote:
>>>
>>> Hi there,
>>>
>>> I have a freeswitch installation that I can make sip calls to to listen
>>> to IVR menus. The sessions last as long as either side does not hang up.
>>> The exception to this is when I use the 'record' dial plan tool. The sip
>>> session ends unexpectedly after about 32+ seconds into the recording.
>>> This happens every time I use the record tool. Note that I have set the
>>> maximum message length to 120 seconds, so this shouldn't be coming into
>>> play here (and shouldn't affect the session anyway).
>>>
>>> Has anyone ever experienced this, and do you have any suggestions as to
>>> what might be the cause?
>>>
>>> Note that there is no NAT involved here. There are also no Expires or
>>> Session-Expires header(s) in the sip INVITE or response that would
>>> affect the length of the session. Indeed, the same type of session can
>>> continue indefinitely until about 32+ seconds after I invoke the record
>>> dial plan tool.
>>>
>>> Thanks,
>>>
>>> Adam
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110830/ecf761d0/attachment-0001.html 

From alex at jajah.com  Tue Aug 30 13:26:06 2011
From: alex at jajah.com (Alex Massover)
Date: Tue, 30 Aug 2011 12:26:06 +0300
Subject: [Freeswitch-users] Condition based on custom sip header
In-Reply-To: <CAKbxfG-pD89e2p+JtYi__kngCt1azbNkCvmnHDUsLGfjbuB-qA@mail.gmail.com>
References: <F5B5EF8F78C0BC4BB417E1DBB9D7F7D0180A75B7CD@jjex01.jajah.dublin>
	<CAKbxfG-pD89e2p+JtYi__kngCt1azbNkCvmnHDUsLGfjbuB-qA@mail.gmail.com>
Message-ID: <F5B5EF8F78C0BC4BB417E1DBB9D7F7D0180A75B951@jjex01.jajah.dublin>

Hi Anthony,

I agree with you, but unfortunately the format of the header is out of my control, our partner sends the INVITE with custom headers. And I need to be able to make condition case in dialplan according to that header.

So there's no way, not even with lua/perl? Maybe there's access to whole SIP message and I can parse it by myself?

I can put for example OpenSIPS before FS and manipulate the header there, but it's kind of strange actually that in SIP softswitch there's no way access to arbitrary SIP header.

Another idea is to make a dirty hack in mod_sofia and otherwrite some existing SIP channel variable, which I don't use :)

> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
> users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale
> Sent: ????? 29 ?????? 2011 23:49
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Condition based on custom sip header
> 
> if its custom anyway and all you need to do is add X- then what's the
> problem?  Using X- is *nearly* a requirement in email/http/sip That's
> the whole point of them having a prefix....
> 
> On Mon, Aug 29, 2011 at 10:36 AM, Alex Massover <alex at jajah.com> wrote:
> > Hi,
> >
> >
> >
> > I?m trying to implement dialplan condition based on custom SIP header
> > (?Header1: value1?). As far as I understood there?s no way to read
> custom
> > SIP header which isn?t X- header with sip_h_. Is that correct?
> >
> >
> >
> > What are the other options, please? Is lua?s getHeader() suitable for
> this?
> >
> >
> >
> > --
> >
> > Best Regards,
> >
> > Alex Massover
> >
> >
> >
> > This mail was sent via Mail-SeCure System.
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> users
> > http://www.freeswitch.org
> >
> >
> 
> 
> 
> --
> Anthony Minessale II
> 
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> users
> http://www.freeswitch.org
> 
> This mail was received via Mail-SeCure System.
> 


This mail was sent via Mail-SeCure System.




From xing2kin at yahoo.com  Tue Aug 30 13:26:50 2011
From: xing2kin at yahoo.com (king2kin)
Date: Tue, 30 Aug 2011 02:26:50 -0700 (PDT)
Subject: [Freeswitch-users] How to start IVR script via dialplan by
	making an outbound call
In-Reply-To: <CABj_V-mFDfiQR+C=aK3wS_VvPF0KugM9wMA0-7sOU2JuW=ZSYw@mail.gmail.com>
Message-ID: <1314696410.9812.YahooMailClassic@web39703.mail.mud.yahoo.com>

Thank for your reply. Something seems wrong with 'record' for out dial IVR.

I made a simple dialplan and Lua script for testing IVR started by outbound-call instead of inbound call as follows:

- xml dialplan extension "8887":
{
<include>
	<extension name="Simple IVR Lua Test">
	<condition field="destination_number" expression="^(8887)$">
		<action application="lua" data="test1.lua"/>
	</condition>
	</extension> 
</include>
}

- Lua script "test1.lua"
{

-- test1.lua
-- Answer call, play a prompt, record a msg, play back msg, hangup

-- Set the path separator
pathsep = '/'

-- Windows users do this instead:
-- pathsep = '\'

--Answer the call
session:answer()

--Create a string with path and filename of a sound file
prompt = "ivr" .. pathsep .. "ivr-welcome_to_freeswitch.wav"

-- Print a log message
freeswitch.consoleLog("INFO","Prompt file is '" .. prompt .. "'\n")

--Play the prompt
session:streamFile(prompt)

-- Record record file
session:streamFile("phrase:voicemail_record_message")
			
-- Play a ""bong"" tone prior to recording
session:streamFile("tone_stream://v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1000, 0, 640)")

-- record a message
filename = session:getVariable('sounds_dir') .. pathsep .. "123.wav"
session:recordFile(filename,300,100,10)

-- play back the recorded msg
session:streamFile(filename)

-- Hangup
session:hangup()
}

- Testing results:

1. call FS IVR normally by an inbound call via X-Lite with user/1005

{
dial 8887
}
then, everything runs as what I expect; it may answer call, play a prompt, record a msg, play back msg, hangup.

2. call FS IVR by an outbound call (FS call a user who picks up the call, then follow IVR to take actions)

from the command-line of Freewitch, I type the following command:
{
originate user/1005 &transfer(8887 xml default)
}
As soon as the above command is submitted, X-Lite of user/1005 rings, I hit button of X-Lite to answer the call, ok it plays prompt and pong tone,
I start to speak until I hit a dtmf button '#', it stops recording, however, nothing is recorded and the file is empty although a sound file "123.wav" (this filename was hardcoded) was indeeded generated on the disk.

I don't know why recordFile does NOT work for the case of outbound call IVR, but recordFile does work for inbound call IVR; they share the same dialplan extension and IVR Lua script. What's problem here? I also tried
"session:record(...)" and "session:record_session(...)", they all didn't work for outbound call IVR.

3. I also tried your testing command with "bridge", e.g.
{
originate user/1005 &bridge(user/8887)
or
originate user/1005 &bridge(user/8887 at default) 
}

"bridge" basically doesn't work for my case.

I read another post from someone else, which mentioned that sip "record" always failed after 32 seconds. Is it possible that sip "record" has a bug?


--- On Mon, 8/29/11, Isaac Jurado <ijurado at econcept.es> wrote:

> From: Isaac Jurado <ijurado at econcept.es>
> Subject: Re: [Freeswitch-users] How to start IVR script via dialplan by making an outbound call
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Date: Monday, August 29, 2011, 2:31 PM
> On Mon, Aug 29, 2011 at 7:24 PM,
> king2kin <xing2kin at yahoo.com>
> wrote:
> > Hi folks,
> >
> > It's easy to start to run IVR script by an inbound
> call on FreeSwitch.
> > However, I don't know to start to run IVR script in
> the following
> > case:
> >
> > I'd like to run an external client application which
> will ask
> > FreeSwitch to make an outbound call to a mobile phone
> number, when the
> > phone rings, someone picks it up and then she/he
> starts to interact
> > with FreeSwitch normally like an inbount-call
> initiates to run IVR
> > script.
> >
> > Could anyone please give me some advice?
> 
> I think that, from the FreeSWITCH console you can do
> something like:
> 
> originate path/to/mobile_num
> &bridge(path/to/special_extension)
> 
> And configure "path/to/special_extension" to execute the
> "ivr" dialplan
> application.
> 
> Cheers.
> 
> -- 
> Isaac Jurado
> Internet Busines Solutions eConcept
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


From xing2kin at yahoo.com  Tue Aug 30 13:39:25 2011
From: xing2kin at yahoo.com (king2kin)
Date: Tue, 30 Aug 2011 02:39:25 -0700 (PDT)
Subject: [Freeswitch-users] Session ends unexpectedly during record dial
	plan usage
In-Reply-To: <CAKzWOxX0LHJVZqeHn8z2QfypvXifLUMVuTYnT0Tgp3OGBd0q3w@mail.gmail.com>
Message-ID: <1314697165.26521.YahooMailClassic@web39701.mail.mud.yahoo.com>

I also encountered a problem in sip recording during outbound IVR call where Freeswitch makes an outbound call to some user, and he/she answered the call to take actions following FS IVR.

for the same dialplan extension and IVR lua script, IVR runs perfect for inbound call; however, the IVR always failed to record message for outbound call although it may still play audio and detect dtmf ...? For recording, a sound file is always generated on disk but it contains nothing.

For its details (dialplan and ivr lua script), please look at my post with subject [How to start IVR script via dialplan by making an outbound call].

Is it possible that sip record has some bug?

--- On Tue, 8/30/11, Michael Collins <msc at freeswitch.org> wrote:

From: Michael Collins <msc at freeswitch.org>
Subject: Re: [Freeswitch-users] Session ends unexpectedly during record dial plan usage
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Tuesday, August 30, 2011, 1:39 AM



On Monday, August 29, 2011, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> /me punches MSC in the arm.....

Haha, I deserved that one. Might wanna smack me with the ClueBat (tm) as well.


Adam,
 If you simply want to record the call between two parties then look on the wiki for the record_session app. Look at the diff between it and the record app. They are *totally* different concepts. 

Let us know if you have any other questions.


-MC
>
> On Mon, Aug 29, 2011 at 12:29 PM, Michael Collins <msc at freeswitch.org> wrote:
>> Go ahead and get a console debug log on this along with a SIP trace. Drop it

>> in pastebin. Hopefully it contains some clues as to what is happening.
>> -MC
>>
>> On Thu, Aug 25, 2011 at 11:55 AM, Adam Kelloway <adam.kelloway at newpace.ca>

>> wrote:
>>>
>>> Hi there,
>>>
>>> I have a freeswitch installation that I can make sip calls to to listen
>>> to IVR menus. The sessions last as long as either side does not hang up.

>>> The exception to this is when I use the 'record' dial plan tool. The sip
>>> session ends unexpectedly after about 32+ seconds into the recording.
>>> This happens every time I use the record tool. Note that I have set the

>>> maximum message length to 120 seconds, so this shouldn't be coming into
>>> play here (and shouldn't affect the session anyway).
>>>
>>> Has anyone ever experienced this, and do you have any suggestions as to

>>> what might be the cause?
>>>
>>> Note that there is no NAT involved here. There are also no Expires or
>>> Session-Expires header(s) in the sip INVITE or response that would

>>> affect the length of the session. Indeed, the same type of session can
>>> continue indefinitely until about 32+ seconds after I invoke the record
>>> dial plan tool.
>>>

>>> Thanks,
>>>
>>> Adam
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org

>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

>>> http://www.freeswitch.org
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org

>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org

> pstn:+19193869900
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

>

-----Inline Attachment Follows-----


FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110830/449fc005/attachment.html 

From mrene_lists at avgs.ca  Tue Aug 30 13:46:44 2011
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Tue, 30 Aug 2011 11:46:44 +0200
Subject: [Freeswitch-users] ringing status and mod_rtmp
In-Reply-To: <909E89D1B3434E64B75BFC53726CE0A9@e1705>
References: <909E89D1B3434E64B75BFC53726CE0A9@e1705>
Message-ID: <D8C0BFEB-00F9-4F40-A341-FB7398050B9F@avgs.ca>

There is a callback for the call's state, it should be set to RINGING whenever the other side rings. 

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 2011-08-25, at 4:38 PM, Madovsky wrote:

> Hi folks,
>  
> is there any callback event for ringing status of leg B with mod_rtmp ?
>  
> Thanks
>  
> Franck
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110830/7bb0ed3c/attachment-0001.html 

From covici at ccs.covici.com  Tue Aug 30 14:39:56 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Tue, 30 Aug 2011 06:39:56 -0400
Subject: [Freeswitch-users] problem using mod_managed under linux
Message-ID: <30572.1314700796@ccs.covici.com>

Hi.  I am trying to use mod_managed on my linux box -- gentoo
distribution -- and its driving me ...


I wrote a very simple program for a test.  The program just sets a dtmf
callback and then streams a file and writes a log entry when it sees the
dtmf.

Under Windows 7 net framework 4, it works just fine.  Under Linux, using
mono versions 2.8.2 qand 2.10.4, I get the following exception which I
will put in a pastebin.

http://pastebin.freeswitch.org/17232

And the app is in this one

http://pastebin.freeswitch.org/17233

I have tried using gmcs and dmcs thinking it may be a .net framework
issue, and if I do it wrong, I can bring down fs itself, but the best I
can get under Linux is the exception shown above.

Thanks in advance for anything you can come up with on this.

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From kbdfck at gmail.com  Tue Aug 30 15:50:02 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Tue, 30 Aug 2011 15:50:02 +0400
Subject: [Freeswitch-users] No sound in intercepted call leg
Message-ID: <CA+GmxbaAacSHoWCcgh1MRHTp8sLFodqCKWP=sALmAXT=y+2FNA@mail.gmail.com>

Hi All

<action application="set" data="intercept_unanswered_only=true"/>
<action application="intercept"
data="${hash(select/domain-last_dial_ext/${pbx_subscriber_customer_id}-${pbx_subscriber_callgroup})}"/>
<action application="sleep" data="2000"/>

Uuid of inbound call is placed into hash.

After intercept app run, there is some periodical beeps in interceptor's
phone, and intercepted channel is muted after a period of half-second and
then disconnected by gateway after RTP timeout. If I place interceptor's
phone on hold, intercepted channel hears MOH from FS, but no sound.

If I hangup interceptor's device right after intercept , call returns to
interceptor's device, and when I take it I can speak to intercepted channel.
What am I doing wrong?

-- 
Best regards,

Dmitry Sytchev,
IT Engineer
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110830/5324c5b3/attachment.html 

From kbdfck at gmail.com  Tue Aug 30 16:06:26 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Tue, 30 Aug 2011 16:06:26 +0400
Subject: [Freeswitch-users] Inband DTMF and bind_meta_app problem again
In-Reply-To: <CAKzWOxXZeDRui4n2A4THoX8dE7ochZHS0QM0hizRH1n_A-md1Q@mail.gmail.com>
References: <CA+GmxbYZ1PHwf60O6+bqaypfOyMuXMkLTm7=BFC4=nmuW_Ng=g@mail.gmail.com>
	<CAKzWOxXZeDRui4n2A4THoX8dE7ochZHS0QM0hizRH1n_A-md1Q@mail.gmail.com>
Message-ID: <CA+Gmxbae+wjZ3_XrSj3V5qU2y_V3FN8Vs4xnb=ueJr_OM24Bqw@mail.gmail.com>

It will be hard to track down latest working commit since I update FS
constantly :(
I opened ticket http://jira.freeswitch.org/browse/FS-3529 about att_xfer
problems. I managed DTMF trigger to work.

2011/8/29 Michael Collins <msc at freeswitch.org>

> Best thing to do would be to open a jira and do your best to narrow down
> the commit where it went from working to non-working. Also, do a test with
> latest git to confirm that it hasn't been fixed in the last 10 minutes. Tony
> has been known to do that. ;)
>
> -MC
>
> On Mon, Aug 29, 2011 at 12:36 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
>
>> Hi All
>> After recent updates of DTMF negotiation or some associated stuff we can't
>> use bind_meta_app with inband DTMF. bind_meta_app ignores bindings, although
>> start_dtmf and spandsp_start_dtmf detect DTMF. We use git-fb6e979 2011-08-26
>> 04-48-33 +0000.
>>
>>
>> <anti-action application="set" data="dtmf_type=${pbx_call_dtmf_type}"/>
>> <anti-action application="spandsp_start_dtmf"/>
>> <anti-action application="bind_meta_app" data="7 a s
>> execute_extension::att_xfer XML features"/>
>>
>> EXECUTE sofia/local/test1 at 85.114.2.200 bind_meta_app(7 a s
>> execute_extension::att_xfer XML features)
>>
>> 2011-08-29 11:39:09.371414 [INFO] switch_ivr_async.c:3066 Bound A-Leg: *7
>> execute_extension::att_xfer XML features
>> ...cut...
>>  2011-08-29 11:40:15.751415 [DEBUG] mod_spandsp_dsp.c:56 DTMF BEGIN
>> DETECTED: [*]
>> 2011-08-29 11:40:15.751415 [DEBUG] switch_ivr_bridge.c:391 Send signal
>> sofia/external/89215572714 [BREAK]
>> 2011-08-29 11:40:15.831571 [DEBUG] mod_spandsp_dsp.c:68 DTMF END DETECTED:
>> [*], duration = 80 ms
>> 2011-08-29 11:40:15.951420 [DEBUG] mod_spandsp_dsp.c:56 DTMF BEGIN
>> DETECTED: [7]
>> 2011-08-29 11:40:15.951420 [DEBUG] switch_ivr_bridge.c:391 Send signal
>> sofia/external/89215572714 [BREAK]
>> 2011-08-29 11:40:16.051411 [DEBUG] mod_spandsp_dsp.c:68 DTMF END DETECTED:
>> [7], duration = 100 ms
>> 2011-08-29 11:40:17.391410 [DEBUG] mod_spandsp_dsp.c:56 DTMF BEGIN
>> DETECTED: [*]
>> 2011-08-29 11:40:17.391410 [DEBUG] switch_ivr_bridge.c:391 Send signal
>> sofia/external/89215572714 [BREAK]
>> 2011-08-29 11:40:17.471409 [DEBUG] mod_spandsp_dsp.c:68 DTMF END DETECTED:
>> [*], duration = 80 ms
>> 2011-08-29 11:40:17.571412 [DEBUG] mod_spandsp_dsp.c:56 DTMF BEGIN
>> DETECTED: [7]
>> 2011-08-29 11:40:17.571412 [DEBUG] switch_ivr_bridge.c:391 Send signal
>> sofia/external/89215572714 [BREAK]
>> 2011-08-29 11:40:17.631411 [DEBUG] mod_spandsp_dsp.c:68 DTMF END DETECTED:
>> [7], duration = 60 ms
>>
>> But no defined application is launched.
>>
>>
>> --
>> Best regards,
>>
>> Dmitry Sytchev,
>> IT Engineer
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Best regards,

Dmitry Sytchev,
IT Engineer
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110830/2455b9c2/attachment.html 

From adam.kelloway at newpace.ca  Tue Aug 30 16:10:27 2011
From: adam.kelloway at newpace.ca (Adam Kelloway)
Date: Tue, 30 Aug 2011 09:10:27 -0300
Subject: [Freeswitch-users] Session ends unexpectedly during record
 dial	plan usage
In-Reply-To: <CAKzWOxX0LHJVZqeHn8z2QfypvXifLUMVuTYnT0Tgp3OGBd0q3w@mail.gmail.com>
References: <4E569AA6.2010008@newpace.ca>	<CAKzWOxWk_9Janh4ZEjpx2J1NKFcQGf0oy97zZC8w14xku130Wg@mail.gmail.com>	<CAKbxfG-Krk-MKXZZdTcTJEyf628siaMvGHTy1PQ_nC+1DnYoKQ@mail.gmail.com>
	<CAKzWOxX0LHJVZqeHn8z2QfypvXifLUMVuTYnT0Tgp3OGBd0q3w@mail.gmail.com>
Message-ID: <4E5CD333.50401@newpace.ca>

This was reproduced when making a call to FS using a Linphone SIP 
client. The SIP client displays a message saying that the session was 
ended by the remote peer (FS) unexpectedly. A SIP trace shows, however, 
that the Linphone client ended it unexpectedly (sent the BYE). I haven't 
seen this when calling from other user agents. The FS logs didn't show 
anything out of the ordinary.

I only use that client for testing anyway. If anyone has it installed, 
they can try out this scenario for their own curiosity and see if you 
see the same behavior.

In any case, thanks for the reply,

Adam

On 3:59 PM, Michael Collins wrote:
>
>
> On Monday, August 29, 2011, Anthony Minessale 
> <anthony.minessale at gmail.com <mailto:anthony.minessale at gmail.com>> wrote:
> > /me punches MSC in the arm.....
>
> Haha, I deserved that one. Might wanna smack me with the ClueBat (tm) 
> as well.
>
> Adam,
> If you simply want to record the call between two parties then look on 
> the wiki for the record_session app. Look at the diff between it and 
> the record app. They are *totally* different concepts.
>
> Let us know if you have any other questions.
>
> -MC
> >
> > On Mon, Aug 29, 2011 at 12:29 PM, Michael Collins 
> <msc at freeswitch.org <mailto:msc at freeswitch.org>> wrote:
> >> Go ahead and get a console debug log on this along with a SIP 
> trace. Drop it
> >> in pastebin. Hopefully it contains some clues as to what is happening.
> >> -MC
> >>
> >> On Thu, Aug 25, 2011 at 11:55 AM, Adam Kelloway 
> <adam.kelloway at newpace.ca <mailto:adam.kelloway at newpace.ca>>
> >> wrote:
> >>>
> >>> Hi there,
> >>>
> >>> I have a freeswitch installation that I can make sip calls to to 
> listen
> >>> to IVR menus. The sessions last as long as either side does not 
> hang up.
> >>> The exception to this is when I use the 'record' dial plan tool. 
> The sip
> >>> session ends unexpectedly after about 32+ seconds into the recording.
> >>> This happens every time I use the record tool. Note that I have 
> set the
> >>> maximum message length to 120 seconds, so this shouldn't be coming 
> into
> >>> play here (and shouldn't affect the session anyway).
> >>>
> >>> Has anyone ever experienced this, and do you have any suggestions 
> as to
> >>> what might be the cause?
> >>>
> >>> Note that there is no NAT involved here. There are also no Expires or
> >>> Session-Expires header(s) in the sip INVITE or response that would
> >>> affect the length of the session. Indeed, the same type of session can
> >>> continue indefinitely until about 32+ seconds after I invoke the 
> record
> >>> dial plan tool.
> >>>
> >>> Thanks,
> >>>
> >>> Adam
> >>>
> >>>
> >>>
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org 
> <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> 
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >>
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org 
> <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> 
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> > googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:+19193869900
> >
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org 
> <mailto:FreeSWITCH-users at lists.freeswitch.org>
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > 

-- 
Adam
-- 
NewPace Logo 	
	
	
	Adam Kelloway

	Software Engineer, NewPace
phone 	+1 (902) 406--8375  x1031
email 	Adam.Kelloway at NewPace.com <mailto:Adam.Kelloway at newpace.com>
aim <aim:GoIm?screenname=Adam.Kelloway at newpace.com>/msn 
<msnim:chat?contact=Adam.Kelloway at newpace.com> 	Adam.Kelloway 
<aim:GoIm?screenname=Adam.Kelloway at newpace.ca>@NewPace.ca

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110830/2bc6bbf4/attachment-0001.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: Newpace_50x50.png
Type: image/png
Size: 4620 bytes
Desc: not available
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110830/2bc6bbf4/attachment-0001.png 

From peter.olsson at visionutveckling.se  Tue Aug 30 16:22:40 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 30 Aug 2011 14:22:40 +0200
Subject: [Freeswitch-users] Session ends unexpectedly during record
	dial	plan usage
In-Reply-To: <4E5CD333.50401@newpace.ca>
References: <4E569AA6.2010008@newpace.ca>
	<CAKzWOxWk_9Janh4ZEjpx2J1NKFcQGf0oy97zZC8w14xku130Wg@mail.gmail.com>
	<CAKbxfG-Krk-MKXZZdTcTJEyf628siaMvGHTy1PQ_nC+1DnYoKQ@mail.gmail.com>
	<CAKzWOxX0LHJVZqeHn8z2QfypvXifLUMVuTYnT0Tgp3OGBd0q3w@mail.gmail.com>
	<4E5CD333.50401@newpace.ca>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59EBE872DE@cooper>

If using record, try setting this in the dialplan first;

<action application="set" data="record_waste_resources=true"/>

This will force FS to send RTP - which might be the cause of the problem.

/Peter


Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Adam Kelloway
Skickat: den 30 augusti 2011 14:10
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Session ends unexpectedly during record dial plan usage

This was reproduced when making a call to FS using a Linphone SIP client. The SIP client displays a message saying that the session was ended by the remote peer (FS) unexpectedly. A SIP trace shows, however, that the Linphone client ended it unexpectedly (sent the BYE). I haven't seen this when calling from other user agents. The FS logs didn't show anything out of the ordinary.

I only use that client for testing anyway. If anyone has it installed, they can try out this scenario for their own curiosity and see if you see the same behavior.

In any case, thanks for the reply,

Adam

On 3:59 PM, Michael Collins wrote:


On Monday, August 29, 2011, Anthony Minessale <anthony.minessale at gmail.com<mailto:anthony.minessale at gmail.com>> wrote:
> /me punches MSC in the arm.....

Haha, I deserved that one. Might wanna smack me with the ClueBat (tm) as well.

Adam,
If you simply want to record the call between two parties then look on the wiki for the record_session app. Look at the diff between it and the record app. They are *totally* different concepts.

Let us know if you have any other questions.

-MC
>
> On Mon, Aug 29, 2011 at 12:29 PM, Michael Collins <msc at freeswitch.org<mailto:msc at freeswitch.org>> wrote:
>> Go ahead and get a console debug log on this along with a SIP trace. Drop it
>> in pastebin. Hopefully it contains some clues as to what is happening.
>> -MC
>>
>> On Thu, Aug 25, 2011 at 11:55 AM, Adam Kelloway <adam.kelloway at newpace.ca<mailto:adam.kelloway at newpace.ca>>
>> wrote:
>>>
>>> Hi there,
>>>
>>> I have a freeswitch installation that I can make sip calls to to listen
>>> to IVR menus. The sessions last as long as either side does not hang up.
>>> The exception to this is when I use the 'record' dial plan tool. The sip
>>> session ends unexpectedly after about 32+ seconds into the recording.
>>> This happens every time I use the record tool. Note that I have set the
>>> maximum message length to 120 seconds, so this shouldn't be coming into
>>> play here (and shouldn't affect the session anyway).
>>>
>>> Has anyone ever experienced this, and do you have any suggestions as to
>>> what might be the cause?
>>>
>>> Note that there is no NAT involved here. There are also no Expires or
>>> Session-Expires header(s) in the sip INVITE or response that would
>>> affect the length of the session. Indeed, the same type of session can
>>> continue indefinitely until about 32+ seconds after I invoke the record
>>> dial plan tool.
>>>
>>> Thanks,
>>>
>>> Adam
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com<mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

--
Adam
--
[cid:image001.png at 01CC6720.4676D8B0]

Adam Kelloway


Software Engineer, NewPace

phone

+1 (902) 406-8375  x1031

email

Adam.Kelloway at NewPace.com<mailto:Adam.Kelloway at newpace.com>

aim<aim:GoIm?screenname=Adam.Kelloway at newpace.com>/msn<msnim:chat?contact=Adam.Kelloway at newpace.com>

Adam.Kelloway<aim:GoIm?screenname=Adam.Kelloway at newpace.ca>@NewPace.ca

!DSPAM:4e5cd35d32761866311061!
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110830/e4a905ff/attachment.html 
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image001.png
Type: image/png
Size: 4620 bytes
Desc: image001.png
Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110830/e4a905ff/attachment.png 

From kbdfck at gmail.com  Tue Aug 30 17:55:01 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Tue, 30 Aug 2011 17:55:01 +0400
Subject: [Freeswitch-users] No sound in intercepted call leg
In-Reply-To: <CA+GmxbaAacSHoWCcgh1MRHTp8sLFodqCKWP=sALmAXT=y+2FNA@mail.gmail.com>
References: <CA+GmxbaAacSHoWCcgh1MRHTp8sLFodqCKWP=sALmAXT=y+2FNA@mail.gmail.com>
Message-ID: <CA+GmxbaB=VUYRNV7yc5MGAMfXnnLd5i1ZsOveyDLhr5MaH5LfQ@mail.gmail.com>

Seems to be solved, problem was in unconfigured Linksys PAP2T.
Interception works.

2011/8/30 Dmitry Sytchev <kbdfck at gmail.com>

> Hi All
>
> <action application="set" data="intercept_unanswered_only=true"/>
> <action application="intercept"
> data="${hash(select/domain-last_dial_ext/${pbx_subscriber_customer_id}-${pbx_subscriber_callgroup})}"/>
> <action application="sleep" data="2000"/>
>
> Uuid of inbound call is placed into hash.
>
> After intercept app run, there is some periodical beeps in interceptor's
> phone, and intercepted channel is muted after a period of half-second and
> then disconnected by gateway after RTP timeout. If I place interceptor's
> phone on hold, intercepted channel hears MOH from FS, but no sound.
>
> If I hangup interceptor's device right after intercept , call returns to
> interceptor's device, and when I take it I can speak to intercepted channel.
> What am I doing wrong?
>
> --
> Best regards,
>
> Dmitry Sytchev,
> IT Engineer
>



-- 
Best regards,

Dmitry Sytchev,
IT Engineer
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110830/7fb53ebc/attachment-0001.html 

From pray at theprossergroup.com  Tue Aug 30 18:11:59 2011
From: pray at theprossergroup.com (Praveen Ray)
Date: Tue, 30 Aug 2011 10:11:59 -0400
Subject: [Freeswitch-users] FS behind an IPCOP firewall
Message-ID: <CAK2JmwpAHY-wR5rWFZ-YRKWj4aO9J4PO8oa3bxNsPLiFL3U8Cg@mail.gmail.com>

Hi
Here's is my setup. It's not that complicated.

            172.168.128.0/24 subnet
FS <-------------------------------------------> IPCOP Firewall
                                                                   |
|                 10.10.0.0/24 subnet
Extension 1000 <--------------------------------->|
|<-------------------------------------------- Extension 1001
(172.168.128.188)
(10.10.0.170)

Firewall is set to allow all UDP traffic in the port rane 1024-65000 between
these two subnets.

Both 1000 and 1001 register successfully with FS. I can make outgoing calls
(using vitelity) using both. I can call 1000 from 1001 ok. However,
calling 1001 from 1000 doesn't work. It goes right to 1001's VM. It's as if
FS doesn't know 1001 is registered. My config is almost same as what
comes with vanilla FS install (except I defined outgoing vitelity and
1000.xml and 1001.xml extensions).

Any obvious config I'm missing? Thanks in advance.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110830/8038a9cf/attachment.html 

From chad at apartmentlines.com  Tue Aug 30 19:17:51 2011
From: chad at apartmentlines.com (Chad Phillips -- Apartment Lines)
Date: Tue, 30 Aug 2011 08:17:51 -0700
Subject: [Freeswitch-users] How to save voicemail msgs to a db
In-Reply-To: <CABj_V-kNRbtyJZM_frOgxxG2_w58WPX3W+RijSMycHAEkEwsmQ@mail.gmail.com>
References: <33095823FD21DF429B481B5163264B79511864FA82@VMBX102.ihostexchange.net>
	<E826C94A3C6BE44198FAF4D66B5AB4D81BDC435DF9@VA3DIAXVS351.RED001.local>
	<33095823FD21DF429B481B5163264B79511864FB4B@VMBX102.ihostexchange.net>
	<E826C94A3C6BE44198FAF4D66B5AB4D81BDD817201@VA3DIAXVS351.RED001.local>
	<CABj_V-kNRbtyJZM_frOgxxG2_w58WPX3W+RijSMycHAEkEwsmQ@mail.gmail.com>
Message-ID: <2DF1946B-FDD4-4577-A8DE-F54AC1D3B4CC@apartmentlines.com>


On Aug 29, 2011, at 2:35 PM, Isaac Jurado wrote:

> For the record.  We intercept the storage-dir with a small FUSE driver
> so we can store voicemails in a MongoDB/GridFS collection.

how reliable have you found FUSE to be?  i'm considering doing a Lua-based FUSE binding to CouchDB for the same purpose, but i've had concerns that the approach wouldn't be as stable a a 'normal' filesystem; what's your experience been?

chad



From avi at avimarcus.net  Tue Aug 30 19:27:22 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Tue, 30 Aug 2011 18:27:22 +0300
Subject: [Freeswitch-users] How to save voicemail msgs to a db
In-Reply-To: <2DF1946B-FDD4-4577-A8DE-F54AC1D3B4CC@apartmentlines.com>
References: <33095823FD21DF429B481B5163264B79511864FA82@VMBX102.ihostexchange.net>
	<E826C94A3C6BE44198FAF4D66B5AB4D81BDC435DF9@VA3DIAXVS351.RED001.local>
	<33095823FD21DF429B481B5163264B79511864FB4B@VMBX102.ihostexchange.net>
	<E826C94A3C6BE44198FAF4D66B5AB4D81BDD817201@VA3DIAXVS351.RED001.local>
	<CABj_V-kNRbtyJZM_frOgxxG2_w58WPX3W+RijSMycHAEkEwsmQ@mail.gmail.com>
	<2DF1946B-FDD4-4577-A8DE-F54AC1D3B4CC@apartmentlines.com>
Message-ID: <CALNZuwm6TExwogNJ4jV9ThwGDjDvdsTXa5OKU6Bj6H-oo5qO0w@mail.gmail.com>

If you really want to do this, the most stable would be incorporating the DB
/ odbc writing directly into the voicemail mod and add it as an option. The
fuse setup seems to be a hack.

-Avi


On Tue, Aug 30, 2011 at 6:17 PM, Chad Phillips -- Apartment Lines <
chad at apartmentlines.com> wrote:

>
> On Aug 29, 2011, at 2:35 PM, Isaac Jurado wrote:
>
> > For the record.  We intercept the storage-dir with a small FUSE driver
> > so we can store voicemails in a MongoDB/GridFS collection.
>
> how reliable have you found FUSE to be?  i'm considering doing a Lua-based
> FUSE binding to CouchDB for the same purpose, but i've had concerns that the
> approach wouldn't be as stable a a 'normal' filesystem; what's your
> experience been?
>
> chad
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110830/b97e1707/attachment.html 

From ijurado at econcept.es  Tue Aug 30 19:43:02 2011
From: ijurado at econcept.es (Isaac Jurado)
Date: Tue, 30 Aug 2011 17:43:02 +0200
Subject: [Freeswitch-users] How to save voicemail msgs to a db
In-Reply-To: <2DF1946B-FDD4-4577-A8DE-F54AC1D3B4CC@apartmentlines.com>
References: <33095823FD21DF429B481B5163264B79511864FA82@VMBX102.ihostexchange.net>
	<E826C94A3C6BE44198FAF4D66B5AB4D81BDC435DF9@VA3DIAXVS351.RED001.local>
	<33095823FD21DF429B481B5163264B79511864FB4B@VMBX102.ihostexchange.net>
	<E826C94A3C6BE44198FAF4D66B5AB4D81BDD817201@VA3DIAXVS351.RED001.local>
	<CABj_V-kNRbtyJZM_frOgxxG2_w58WPX3W+RijSMycHAEkEwsmQ@mail.gmail.com>
	<2DF1946B-FDD4-4577-A8DE-F54AC1D3B4CC@apartmentlines.com>
Message-ID: <CABj_V-nXgZuwPErccy5stN9+y3gJmVEg4LtvkMv3EKDdq4EWwQ@mail.gmail.com>

On Tue, Aug 30, 2011 at 5:17 PM, Chad Phillips -- Apartment Lines
<chad at apartmentlines.com> wrote:
>
> On Aug 29, 2011, at 2:35 PM, Isaac Jurado wrote:
>
>> For the record. ?We intercept the storage-dir with a small FUSE
>> driver so we can store voicemails in a MongoDB/GridFS collection.
>
> how reliable have you found FUSE to be? ?i'm considering doing a
> Lua-based FUSE binding to CouchDB for the same purpose, but i've had
> concerns that the approach wouldn't be as stable a a 'normal'
> filesystem; what's your experience been?

Well, we don't have load enough to find bottlenecks.  Neither we have
time to prepare a stress test.

Our implementation is a single python script that uses python-fuse and
pymongo.  Altogether, they seem to work concurrently (with threads) and
cooperate nicely.  Although everything is mostly IO bound, eventually we
may hit the GIL wall, in such case we will have to reconsider our
solution.

You may have better luck with Lua, in case you do threading, by using a
different interpreter instance per-thread (just like mod_wsgi does with
Python).

Cheers.

-- 
Isaac Jurado
Internet Busines Solutions eConcept


From kevin at networxonline.com  Tue Aug 30 01:54:13 2011
From: kevin at networxonline.com (Kevin Reeves)
Date: Mon, 29 Aug 2011 14:54:13 -0700
Subject: [Freeswitch-users] Calling FMS with mod_rtmp
Message-ID: <B2C62E9A-3C98-41AF-9500-C6EAA45AF849@networxonline.com>

I'm interested in making a call from inside Freeswitch out to Flash Media Server using mod_rtmp.  I'd like to do this so I can attach audio from a phone call into FMS.  If I'm reading the documentation correctly, the only way to attach the 2 is to have FMS register with Freeswitch.

Any suggestions you have would be greatly appreciated.

Thanks,
Kevin

From danlanweb at gmail.com  Tue Aug 30 03:15:44 2011
From: danlanweb at gmail.com (Dan Lan)
Date: Mon, 29 Aug 2011 16:15:44 -0700
Subject: [Freeswitch-users] Question -- DTMF from RFC2833 to SIP INFO
Message-ID: <CAAuJ8MD3wTijLNTzP10bofPiLuEm8yVNVBGdygkRX985HF6u-w@mail.gmail.com>

Hi,
I am new to the Freeswitch community, just compiled my first FS window
version one month ago.
I have registered two SNOM 300 IP-Phone with extension 1001, 1002 into FS.
I also setup an external gateway as SIP trunk to service provider. (For
certain reason, the service provider only accept INFO as the DTMF signal.)

I add the following line into the external gateway profile
<param name="dtmf-type" value="info"/>

I also add the following line into the dail-plan
<action application = "set" data="dtmf_type=info"/>

When I dail the phone number from extension 1001, the channel bridge through
the SIP trunk with both-way audio no problem.
However, when I enter the DTMF, the signal did not send to the provider SIP
trunk with SIP INFO.

from the log file, I can see the DTMF was recorgnized by FreeSwitch from
extension 1001 (it was sending RFC2933), and I can see FS try to send signal
to the external trunk. (but no SIP INFO or RFC2833 in Wireshark capture)

[DEBUG] switch_rtp.c:3303 RTP RECV DTMF 1:1280
2011-08-29 15:44:10.081713 [DEBUG] switch_ivr_bridge.c:391 Send signal
sofia/external/2131234567 at xxx.xxx.xxx.xxx:5060 [BREAK]
2011-08-29 15:44:11.033081 [DEBUG] switch_rtp.c:3303 RTP RECV DTMF 2:1120
2011-08-29 15:44:11.033081 [DEBUG] switch_ivr_bridge.c:391 Send signal
sofia/external/2131234567 at xxx.xxx.xxx.xxx:5060 [BREAK]
2011-08-29

Did I do something wrong here? Is FS capable of convert RFC2833 signal from
one leg and send SIP INFO to another leg?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110829/26a37886/attachment.html 

From kolmisoft.marketing at gmail.com  Tue Aug 30 13:24:01 2011
From: kolmisoft.marketing at gmail.com (Kolmisoft Marketing)
Date: Tue, 30 Aug 2011 12:24:01 +0300
Subject: [Freeswitch-users] FREE webinar video about Auto-Dialer Business
	Model (Telemarketing)
In-Reply-To: <CAGhGO7hS1TuPYyV1ouPLTGn9kEVe+gkCjyB5ti+tX=1vAokPsA@mail.gmail.com>
References: <CAGhGO7hS1TuPYyV1ouPLTGn9kEVe+gkCjyB5ti+tX=1vAokPsA@mail.gmail.com>
Message-ID: <CAGhGO7hoWrisBfkdzbrMPX=50LHpOwiX5C31KN1TxzMTQ2Pdjw@mail.gmail.com>

Hello,

We would like to share our webinar about Auto-Dialer Business Model
(Telemarketing).

It is educational video which we made for our clients and now we are sharing
it with you.

http://www.kolmisoft.com/how-to-start-a-VoIP-business/webinars/

Enjoy.

NOTE: This is not attempt to sell you anything. No product or service is
sold/marketed in the video.

Regards,
Kolmisoft Team
www.kolmisoft.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110830/38289507/attachment.html 

From infos at madovsky.org  Tue Aug 30 20:36:22 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 30 Aug 2011 12:36:22 -0400
Subject: [Freeswitch-users] Calling FMS with mod_rtmp
References: <B2C62E9A-3C98-41AF-9500-C6EAA45AF849@networxonline.com>
Message-ID: <2C6A32EE4B454BC9897CA8FCB17B5547@e1705>

you can't .
to register directly from flash client
only RTMFP protocol is allowed

----- Original Message ----- 
From: "Kevin Reeves" <kevin at networxonline.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Monday, August 29, 2011 5:54 PM
Subject: [Freeswitch-users] Calling FMS with mod_rtmp


> I'm interested in making a call from inside Freeswitch out to Flash Media 
> Server using mod_rtmp.  I'd like to do this so I can attach audio from a 
> phone call into FMS.  If I'm reading the documentation correctly, the only 
> way to attach the 2 is to have FMS register with Freeswitch.
>
> Any suggestions you have would be greatly appreciated.
>
> Thanks,
> Kevin
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From anthony.minessale at gmail.com  Tue Aug 30 20:51:03 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 30 Aug 2011 11:51:03 -0500
Subject: [Freeswitch-users] Condition based on custom sip header
In-Reply-To: <F5B5EF8F78C0BC4BB417E1DBB9D7F7D0180A75B951@jjex01.jajah.dublin>
References: <F5B5EF8F78C0BC4BB417E1DBB9D7F7D0180A75B7CD@jjex01.jajah.dublin>
	<CAKbxfG-pD89e2p+JtYi__kngCt1azbNkCvmnHDUsLGfjbuB-qA@mail.gmail.com>
	<F5B5EF8F78C0BC4BB417E1DBB9D7F7D0180A75B951@jjex01.jajah.dublin>
Message-ID: <CAKbxfG_dCxQ9qfmtPJMPfDJYr6ZFLyttmxDJmyyRFczhvCN1-g@mail.gmail.com>

> I can put for example OpenSIPS before FS and manipulate the header there, but it's kind of strange actually that in SIP softswitch there's no way access to arbitrary SIP header.
>
> Another idea is to make a dirty hack in mod_sofia and otherwrite some existing SIP channel variable, which I don't use :)

There is a way... It's called using the well-known practice of using
X-headers which is properly name-spaced and reserved for custom
headers.

When people make comments like this it actually annoys me... "Well
your free software and open community are sort of cool but really, if
it was worth a damn at all, it would certainly be able to make me
toast....."

You are the one with the non-standard requirement that you "can't
change" so there is a saying here in open source land ... "Patches
Welcome!"


-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From ijurado at econcept.es  Tue Aug 30 21:01:01 2011
From: ijurado at econcept.es (Isaac Jurado)
Date: Tue, 30 Aug 2011 19:01:01 +0200
Subject: [Freeswitch-users] How to save voicemail msgs to a db
In-Reply-To: <CALNZuwm6TExwogNJ4jV9ThwGDjDvdsTXa5OKU6Bj6H-oo5qO0w@mail.gmail.com>
References: <33095823FD21DF429B481B5163264B79511864FA82@VMBX102.ihostexchange.net>
	<E826C94A3C6BE44198FAF4D66B5AB4D81BDC435DF9@VA3DIAXVS351.RED001.local>
	<33095823FD21DF429B481B5163264B79511864FB4B@VMBX102.ihostexchange.net>
	<E826C94A3C6BE44198FAF4D66B5AB4D81BDD817201@VA3DIAXVS351.RED001.local>
	<CABj_V-kNRbtyJZM_frOgxxG2_w58WPX3W+RijSMycHAEkEwsmQ@mail.gmail.com>
	<2DF1946B-FDD4-4577-A8DE-F54AC1D3B4CC@apartmentlines.com>
	<CALNZuwm6TExwogNJ4jV9ThwGDjDvdsTXa5OKU6Bj6H-oo5qO0w@mail.gmail.com>
Message-ID: <CABj_V-kVCq8t8jRU_Hnu4jSG8qJueTBh+C3gSSn7jCMFLFhvXA@mail.gmail.com>

On Tue, Aug 30, 2011 at 5:27 PM, Avi Marcus <avi at avimarcus.net> wrote:
>
> If you really want to do this, the most stable would be incorporating
> the DB / odbc writing directly into the voicemail mod and add it as an
> option. The fuse setup seems to be a hack.

No doubt it is a hackish solution.  But in the GridFS case, the
filesystem semantics are a bit different.  File truncation to a >0 size,
for example, is an extremely expensive operation.  The mod_voicemail
makes a fairly extensive use of such operation.

Sounds like a lot of work.

Cheers.

-- 
Isaac Jurado
Internet Busines Solutions eConcept


From danlanweb at gmail.com  Wed Aug 31 02:57:05 2011
From: danlanweb at gmail.com (Dan Lan)
Date: Tue, 30 Aug 2011 15:57:05 -0700
Subject: [Freeswitch-users] Connecting Call Between Two SIP Trunks
Message-ID: <CAAuJ8MBmErPiahLHAWOFJb4jLCQX-ePc14HFT07hTw5wr7EOjQ@mail.gmail.com>

Hi,

I want to use FS to accept call from SIP_TrunkA and terminate to SIP_TrunkB
Both SIP trunks are using IP authentication, no need for username and
password.

for incoming call (SIP_TrunkA), I have add the IP address of SIP_TrunkA in
to acl.conf.xml
 <node type="allow" cidr="xxx.xxx.xxx.xxx/32"/>

I understand the incoming call will go to the public context, so I think I
need to do something here.

I dont know what to do next.
1. I try to establish a gateway for SIP_TrunkB for my outgoing call, but
sofia require me to have the username and password for the trunk. I dont
know where to add the SIP_TrunkB in freeswitch, since the provider of
SIP_TrunkB only need to recoginize my FS IP address.
2. After I establish SIP_TrunkB, how should I do on public dialplan to route
the call from SIP_TrunkA to SIP_TrunkB? should I use "transfer" or "bridge",
could I make a dialplan that can route all the call from IP address of A to
IP address of B?

Sorry for the newbie question, I try to look up on wiki but only got partial
information for me.

Any help and any directions or hints are appreciated

Dan Lan
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110830/50b54115/attachment.html 

From wayne at hamilton.net  Wed Aug 31 00:25:26 2011
From: wayne at hamilton.net (Wayne)
Date: Tue, 30 Aug 2011 15:25:26 -0500
Subject: [Freeswitch-users] Setup with Bandwidth.com
Message-ID: <0E3F1549A4FB4287A7F6B78A8EC2761F@ccs.local>

Does anyone have a setup with bandwidth.com? 

The example on the wiki has a username and password. I didn't receive a
username and password form bandwidth.com. Just setup a sip peer and have an
IP address. If I send calls to freeswitch from the peer I get Rejected by
acl "domains". Falling back to Digest auth. Looks like they are sending it
on port 5060. 
I'm sure the rejected by acl is the most asked question but I have been
unable to find the answer.

Thanks Wayne



From rpang88 at gmail.com  Wed Aug 31 00:47:02 2011
From: rpang88 at gmail.com (Ray Pang)
Date: Tue, 30 Aug 2011 13:47:02 -0700
Subject: [Freeswitch-users] DTMF from Nortel BCM 400 to Freeswitch
Message-ID: <CANDXbUyv9Q9i9FuajPs8rG7KaQagC3JG2Ji7zeJS9uZHYsxv+w@mail.gmail.com>

I?ve been unable to get DTMF to work from BCM 400 to Freeswitch





PSTN -> BCM -> sip forward -> Freeswitch Autoattendant.



DTMF never gets recognized:



2011-08-30 13:13:46.164796 [DEBUG] switch_ivr_menu.c:428 Executing IVR menu
auto_attendant_1

2011-08-30 13:13:46.164796 [DEBUG] switch_ivr_play_say.c:1278 Codec
Activated L16 at 8000hz 1 channels 30ms

2011-08-30 13:13:46.174819 [DEBUG] sofia.c:4761 Channel
sofia/sipinterface_1/3335 entering state [ready][200]

2011-08-30 13:13:46.412814 [DEBUG] switch_rtp.c:3083 Correct ip/port
confirmed.

2011-08-30 13:13:46.940791 [DEBUG] sofia.c:6308 dispatched freeswitch event
for INFO

2011-08-30 13:13:47.360757 [DEBUG] sofia.c:6308 dispatched freeswitch event
for INFO

2011-08-30 13:13:48.350702 [DEBUG] sofia.c:6308 dispatched freeswitch event
for INFO

2011-08-30 13:13:48.390701 [DEBUG] sofia.c:6308 dispatched freeswitch event
for INFO

2011-08-30 13:13:48.397689 [DEBUG] sofia.c:6308 dispatched freeswitch event
for INFO

2011-08-30 13:13:48.872658 [DEBUG] switch_ivr_play_say.c:1648 done playing
file

2011-08-30 13:13:48.872658 [DEBUG] switch_ivr_menu.c:343 waiting for 4/4
digits t/o 2000

2011-08-30 13:13:58.893140 [DEBUG] switch_ivr_menu.c:390 digits ''



Packet capture shows this tidbit:



p=Digit-Collection

y=Digits

d=7



I?m very new to this, any help or advice will be appreiciated.



TIA,



RP
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110830/7519de49/attachment.html 

From sescher_ml at omeco.de  Wed Aug 31 01:06:05 2011
From: sescher_ml at omeco.de (Silvio Escher)
Date: Tue, 30 Aug 2011 23:06:05 +0200
Subject: [Freeswitch-users] Variables in Dialplan - Problem with getting
 Variables from User Directory
Message-ID: <4E5D50BD.8010208@omeco.de>

Hi there,

iam really getting crazy these days ;-)

I've changed our Freeswitch Version some Weeks ago ( cannot remember exactly but proably from 1.0.6 
) to the git Head. ).
Since this Change ( or better since the followed Config File Adaptions ;-) ) i noticed that i cannot 
use User Variables/Params inside the XML-Dialplan anymore.

Some (relevant?! ;) ) Copy&Paste ..

my Userentry

<include>
<user id="99">
<params>
<param name="password" value="something"/>
<param name="vm-mailto" value="maybe_me at omeco.de"/>
<param name="vm-mailfrom" value="noreply at omeco.de"/>
<param name="vm-enabled" value="true"/>
<param name="vm-password" value="user-choose"/>
<param name="vm-email-all-messages" value="false"/>
<param name="vm-notify-email-all-messages" value="true"/>
<param name="vm-keep-local-after-email" value="true"/>
</params>
<variables>
<variable name="outbound_caller_id_name" value="Omeco GmbH"/>
<variable name="outbound_caller_id_number" value="06316803540"/>
<variable name="effective_caller_id_name" value="Silvio Escher"/>
<variable name="effective_caller_id_number" value="40"/>
<variable name="callgroup" value="1"/>
</variables>
</user>
</include>

some dp snippet..

before the change something like

<action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/>
<action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/>

was working fine ..

Actually iam using

<action application="set" data="effective_caller_id_name=${user_data(${username}@${domain_name} var 
outbound_caller_id_name)}"/>
<action application="set" data="effective_caller_id_number=${user_data(${username}@${domain_name} 
var outbound_caller_id_number)}"/>

or an

<action application="set_user" data="$1@${domain_name}"/>

to get some or all Variables.

But i Noticed that the "set_user" Thingie just gets me the Variables - not the Params.
So i've still Issues with mod_voicemail ( Mailto is undef )

2011-08-30 22:02:32.904211 [DEBUG] switch_utils.c:709 Emailed data to [(null)]
2011-08-30 22:02:32.904211 [DEBUG] mod_voicemail.c:2809 Sending notify message to (null)

Maybe i has something todo with an missing auth or wrong Domain or whatever .. but i have no clue 
where to look further.

Any Help ( from little Hints till complete Miracle Solve ;-) ) is actually very welcome.

Thanks in Advance,
Silvio


-- 
Silvio Escher
omeco GmbH

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110830/8e8ee7cc/attachment.html 

From gerardo.barajas at gmail.com  Wed Aug 31 02:54:26 2011
From: gerardo.barajas at gmail.com (Gerardo Barajas)
Date: Tue, 30 Aug 2011 17:54:26 -0500
Subject: [Freeswitch-users] =?windows-1252?q?FreeSWITCH=99_in_Mexico?=
In-Reply-To: <CABH8QX3icB5P-OiEzkqMkwcVtBCmcBNGr=P-iLyD_w2nAOmjpg@mail.gmail.com>
References: <CABH8QX3icB5P-OiEzkqMkwcVtBCmcBNGr=P-iLyD_w2nAOmjpg@mail.gmail.com>
Message-ID: <CAOQhXxEOcRVU7wEYimw=1pj_L+CzrR8jiAxcEMmY50RRtS6Xig@mail.gmail.com>

Nice having you in Mexico.
Really nice talk.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110830/cbd4d515/attachment.html 

From brad at tech21.com  Wed Aug 31 03:10:44 2011
From: brad at tech21.com (Brad Mina)
Date: Tue, 30 Aug 2011 16:10:44 -0700
Subject: [Freeswitch-users] Connecting Call Between Two SIP Trunks
In-Reply-To: <CAAuJ8MBmErPiahLHAWOFJb4jLCQX-ePc14HFT07hTw5wr7EOjQ@mail.gmail.com>
References: <CAAuJ8MBmErPiahLHAWOFJb4jLCQX-ePc14HFT07hTw5wr7EOjQ@mail.gmail.com>
Message-ID: <CAGRr05-WAih_JvibyS6km2ON2gnYDaZbgbVebioQvzxaHgjfZg@mail.gmail.com>

1. Username and password are manditory in the XML - you don't have to put
anything that makes sense, just fill it in with your DID or something and
keep register=false and those details will never do anything.

2. bridge would be the proper tool, you might have to mess around with proxy
media or make sure proxy_media is off to ensure the data is coming from you
directly and not negotiated between providers.

On Tue, Aug 30, 2011 at 3:57 PM, Dan Lan <danlanweb at gmail.com> wrote:

> Hi,
>
> I want to use FS to accept call from SIP_TrunkA and terminate to SIP_TrunkB
> Both SIP trunks are using IP authentication, no need for username and
> password.
>
> for incoming call (SIP_TrunkA), I have add the IP address of SIP_TrunkA in
> to acl.conf.xml
>  <node type="allow" cidr="xxx.xxx.xxx.xxx/32"/>
>
> I understand the incoming call will go to the public context, so I think I
> need to do something here.
>
> I dont know what to do next.
> 1. I try to establish a gateway for SIP_TrunkB for my outgoing call, but
> sofia require me to have the username and password for the trunk. I dont
> know where to add the SIP_TrunkB in freeswitch, since the provider of
> SIP_TrunkB only need to recoginize my FS IP address.
> 2. After I establish SIP_TrunkB, how should I do on public dialplan to
> route the call from SIP_TrunkA to SIP_TrunkB? should I use "transfer" or
> "bridge", could I make a dialplan that can route all the call from IP
> address of A to IP address of B?
>
> Sorry for the newbie question, I try to look up on wiki but only got
> partial information for me.
>
> Any help and any directions or hints are appreciated
>
> Dan Lan
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110830/188e9089/attachment-0001.html 

From rhuddleston at gmail.com  Wed Aug 31 03:18:50 2011
From: rhuddleston at gmail.com (Robert-iPhone)
Date: Tue, 30 Aug 2011 19:18:50 -0400
Subject: [Freeswitch-users] Setup with Bandwidth.com
In-Reply-To: <0E3F1549A4FB4287A7F6B78A8EC2761F@ccs.local>
References: <0E3F1549A4FB4287A7F6B78A8EC2761F@ccs.local>
Message-ID: <C360A16E-0214-4DC4-8CF2-959BFF660349@gmail.com>

Read up on ACL thats what I had to do.
Bandwidth.com only offers IP Auth sip trunks.
Contact me offlist if you need further help

Sent from my iPhone

On Aug 30, 2011, at 4:25 PM, "Wayne" <wayne at hamilton.net> wrote:

> Does anyone have a setup with bandwidth.com? 
> 
> The example on the wiki has a username and password. I didn't receive a
> username and password form bandwidth.com. Just setup a sip peer and have an
> IP address. If I send calls to freeswitch from the peer I get Rejected by
> acl "domains". Falling back to Digest auth. Looks like they are sending it
> on port 5060. 
> I'm sure the rejected by acl is the most asked question but I have been
> unable to find the answer.
> 
> Thanks Wayne
> 
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From rpang88 at gmail.com  Wed Aug 31 07:47:04 2011
From: rpang88 at gmail.com (Ray Pang)
Date: Tue, 30 Aug 2011 20:47:04 -0700
Subject: [Freeswitch-users] DTMF from Nortel BCM 400 to Freeswitch
In-Reply-To: <CANDXbUyv9Q9i9FuajPs8rG7KaQagC3JG2Ji7zeJS9uZHYsxv+w@mail.gmail.com>
References: <CANDXbUyv9Q9i9FuajPs8rG7KaQagC3JG2Ji7zeJS9uZHYsxv+w@mail.gmail.com>
Message-ID: <CANDXbUz8S-i21HcDDyuHJ5oe1HAgzi-8pc-MhNzE_nmkA+JSmA@mail.gmail.com>

>
> I?ve been unable to get DTMF to work from BCM 400 to Freeswitch.  I've
> tried all DTMF settings with no luck.
>
>
>
>
>
> PSTN -> BCM -> sip forward -> Freeswitch Autoattendant.
>
>
>
> DTMF never gets recognized:
>
>
>
> 2011-08-30 13:13:46.164796 [DEBUG] switch_ivr_menu.c:428 Executing IVR menu
> auto_attendant_1
>
> 2011-08-30 13:13:46.164796 [DEBUG] switch_ivr_play_say.c:1278 Codec
> Activated L16 at 8000hz 1 channels 30ms
>
> 2011-08-30 13:13:46.174819 [DEBUG] sofia.c:4761 Channel
> sofia/sipinterface_1/3335 entering state [ready][200]
>
> 2011-08-30 13:13:46.412814 [DEBUG] switch_rtp.c:3083 Correct ip/port
> confirmed.
>
> 2011-08-30 13:13:46.940791 [DEBUG] sofia.c:6308 dispatched freeswitch event
> for INFO
>
> 2011-08-30 13:13:47.360757 [DEBUG] sofia.c:6308 dispatched freeswitch event
> for INFO
>
> 2011-08-30 13:13:48.350702 [DEBUG] sofia.c:6308 dispatched freeswitch event
> for INFO
>
> 2011-08-30 13:13:48.390701 [DEBUG] sofia.c:6308 dispatched freeswitch event
> for INFO
>
> 2011-08-30 13:13:48.397689 [DEBUG] sofia.c:6308 dispatched freeswitch event
> for INFO
>
> 2011-08-30 13:13:48.872658 [DEBUG] switch_ivr_play_say.c:1648 done playing
> file
>
> 2011-08-30 13:13:48.872658 [DEBUG] switch_ivr_menu.c:343 waiting for 4/4
> digits t/o 2000
>
> 2011-08-30 13:13:58.893140 [DEBUG] switch_ivr_menu.c:390 digits ''
>
>
>
> Packet capture shows this tidbit:
>
>
>
> p=Digit-Collection
>
> y=Digits
>
> d=7
>
>
>
> I?m very new to this, any help or advice will be appreiciated.
>
>
>
> TIA,
>
>
>
> RP
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110830/07c292cb/attachment.html 

From norstar at bigmir.net  Wed Aug 31 10:23:28 2011
From: norstar at bigmir.net (sucsession)
Date: Wed, 31 Aug 2011 09:23:28 +0300
Subject: [Freeswitch-users]  Incoming DID
Message-ID: <04592272.20110831092328@bigmir.net>

Hello Freeswitch-users,

I have problem with incoming DID routing.
Internal  phones  normally  registered  and can make outgoing call and
call to each other.
But incoming DID call routed to voice mail or "goodbye" prompt.
Calls to other services (IVR, conference) are routed normally.
I'm connected to Nortel CS1000 via sip trunk.
Call log below:
Thanks.

2011-08-22 04:44:54.970599 [WARNING] sofia_reg.c:1241 SIP auth challenge (REGISTER) on sofia profile 'internal' for [8100 at 10.160.0.3] from ip 10.160.1.212
2011-08-22 04:45:00.718687 [NOTICE] switch_channel.c:816 New Channel sofia/external/7777 [39a3c5ec-834a-4c80-9b2e-b95e1b0f6fbd]
2011-08-22 04:45:00.719679 [DEBUG] sofia.c:4761 Channel sofia/external/7777 entering state [received][100]
2011-08-22 04:45:00.719679 [DEBUG] sofia.c:4772 Remote SDP:
v=0

o=- 663 1 IN IP4 10.1.1.22

s=-

t=0 0

m=audio 5200 RTP/AVP 8 0 18 101 111

c=IN IP4 10.160.0.205

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=rtpmap:111 X-nt-inforeq/8000

a=ptime:20


2011-08-22 04:45:00.719679 [DEBUG] sofia_glue.c:4656 Audio Codec Compare [PCMA:8:8000:20:64000]/[G722:9:8000:20:64000]
2011-08-22 04:45:00.719679 [DEBUG] sofia_glue.c:4656 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000]
2011-08-22 04:45:00.719679 [DEBUG] sofia_glue.c:4656 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000]
2011-08-22 04:45:00.719679 [DEBUG] sofia_glue.c:2788 Set Codec sofia/external/7777 PCMA/8000 20 ms 160 samples 64000 bits
2011-08-22 04:45:00.719679 [DEBUG] sofia_glue.c:4770 Set 2833 dtmf send/recv payload to 101
2011-08-22 04:45:00.719679 [DEBUG] sofia.c:4943 (sofia/external/7777) State Change CS_NEW -> CS_INIT
2011-08-22 04:45:00.719679 [DEBUG] switch_core_session.c:1114 Send signal sofia/external/7777 [BREAK]
2011-08-22 04:45:00.719679 [DEBUG] switch_core_state_machine.c:325 (sofia/external/7777) Running State Change CS_INIT
2011-08-22 04:45:00.719679 [DEBUG] switch_core_state_machine.c:361 (sofia/external/7777) State INIT
2011-08-22 04:45:00.719679 [DEBUG] mod_sofia.c:84 sofia/external/7777 SOFIA INIT
2011-08-22 04:45:00.719679 [DEBUG] mod_sofia.c:124 (sofia/external/7777) State Change CS_INIT -> CS_ROUTING
2011-08-22 04:45:00.719679 [DEBUG] switch_core_session.c:1114 Send signal sofia/external/7777 [BREAK]
2011-08-22 04:45:00.719679 [DEBUG] switch_core_state_machine.c:361 (sofia/external/7777) State INIT going to sleep
2011-08-22 04:45:00.719679 [DEBUG] switch_core_state_machine.c:325 (sofia/external/7777) Running State Change CS_ROUTING
2011-08-22 04:45:00.719679 [DEBUG] switch_channel.c:1667 (sofia/external/7777) Callstate Change DOWN -> RINGING
2011-08-22 04:45:00.719679 [DEBUG] switch_core_state_machine.c:364 (sofia/external/7777) State ROUTING
2011-08-22 04:45:00.719679 [DEBUG] mod_sofia.c:147 sofia/external/7777 SOFIA ROUTING
2011-08-22 04:45:00.719679 [DEBUG] switch_core_state_machine.c:77 sofia/external/7777 Standard ROUTING
2011-08-22 04:45:00.719679 [INFO] mod_dialplan_xml.c:331 Processing Vasya <7777;phone-context=cdp.udp>->8100;phone-context=cdp.udp in context public
Dialplan: sofia/external/7777 parsing [public->unloop] continue=false
Dialplan: sofia/external/7777 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/7777 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/external/7777 parsing [public->outside_call] continue=true
Dialplan: sofia/external/7777 Absolute Condition [outside_call]
Dialplan: sofia/external/7777 Action set(outside_call=true) 
Dialplan: sofia/external/7777 Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) 
Dialplan: sofia/external/7777 parsing [public->call_debug] continue=true
Dialplan: sofia/external/7777 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/external/7777 parsing [public->public_extensions] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [public_extensions] destination_number(8100;phone-context=cdp.udp) =~ /^(10[01][0-9])$/ break=on-false
Dialplan: sofia/external/7777 parsing [public->8000] continue=true
Dialplan: sofia/external/7777 Regex (PASS) [8000] context(public) =~ /public/ break=on-false
Dialplan: sofia/external/7777 Regex (FAIL) [8000] destination_number(8100;phone-context=cdp.udp) =~ /8000/ break=on-false
Dialplan: sofia/external/7777 parsing [public->8001] continue=false
Dialplan: sofia/external/7777 Regex (PASS) [8001] context(public) =~ /public/ break=on-false
Dialplan: sofia/external/7777 Regex (FAIL) [8001] destination_number(8100;phone-context=cdp.udp) =~ /8001/ break=on-false
Dialplan: sofia/external/7777 parsing [public->8100] continue=false
Dialplan: sofia/external/7777 Regex (PASS) [8100] context(public) =~ /public/ break=on-false
Dialplan: sofia/external/7777 Regex (PASS) [8100] destination_number(8100;phone-context=cdp.udp) =~ /8100/ break=on-false
Dialplan: sofia/external/7777 Action transfer(8100 XML Default) 
2011-08-22 04:45:00.720681 [DEBUG] switch_core_state_machine.c:119 (sofia/external/7777) State Change CS_ROUTING -> CS_EXECUTE
2011-08-22 04:45:00.720681 [DEBUG] switch_core_session.c:1114 Send signal sofia/external/7777 [BREAK]
2011-08-22 04:45:00.720681 [DEBUG] switch_core_state_machine.c:364 (sofia/external/7777) State ROUTING going to sleep
2011-08-22 04:45:00.720681 [DEBUG] switch_core_state_machine.c:325 (sofia/external/7777) Running State Change CS_EXECUTE
2011-08-22 04:45:00.720681 [DEBUG] switch_core_state_machine.c:371 (sofia/external/7777) State EXECUTE
2011-08-22 04:45:00.721680 [DEBUG] mod_sofia.c:240 sofia/external/7777 SOFIA EXECUTE
2011-08-22 04:45:00.721680 [DEBUG] switch_core_state_machine.c:157 sofia/external/7777 Standard EXECUTE
EXECUTE sofia/external/7777 set(outside_call=true)
2011-08-22 04:45:00.721680 [DEBUG] mod_dptools.c:1060 sofia/external/7777 SET [outside_call]=[true]
EXECUTE sofia/external/7777 set(RFC2822_DATE=Mon, 22 Aug 2011 04:45:00 +0300)
2011-08-22 04:45:00.721680 [DEBUG] mod_dptools.c:1060 sofia/external/7777 SET [RFC2822_DATE]=[Mon, 22 Aug 2011 04:45:00 +0300]
EXECUTE sofia/external/7777 transfer(8100 XML Default)
2011-08-22 04:45:00.721680 [DEBUG] switch_ivr.c:1597 (sofia/external/7777) State Change CS_EXECUTE -> CS_ROUTING
2011-08-22 04:45:00.721680 [DEBUG] switch_core_session.c:1114 Send signal sofia/external/7777 [BREAK]
2011-08-22 04:45:00.721680 [DEBUG] switch_core_session.c:707 Send signal sofia/external/7777 [BREAK]
2011-08-22 04:45:00.721680 [NOTICE] switch_ivr.c:1603 Transfer sofia/external/7777 to XML[8100 at Default]
2011-08-22 04:45:00.721680 [DEBUG] switch_core_state_machine.c:371 (sofia/external/7777) State EXECUTE going to sleep
2011-08-22 04:45:00.721680 [DEBUG] switch_core_state_machine.c:325 (sofia/external/7777) Running State Change CS_ROUTING
2011-08-22 04:45:00.722681 [DEBUG] switch_core_state_machine.c:364 (sofia/external/7777) State ROUTING
2011-08-22 04:45:00.722681 [DEBUG] mod_sofia.c:147 sofia/external/7777 SOFIA ROUTING
2011-08-22 04:45:00.722681 [DEBUG] switch_core_state_machine.c:77 sofia/external/7777 Standard ROUTING
2011-08-22 04:45:00.722681 [INFO] mod_dialplan_xml.c:331 Processing Vasya <7777;phone-context=cdp.udp>->8100 in context Default
Dialplan: sofia/external/7777 parsing [Default->unloop] continue=false
Dialplan: sofia/external/7777 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/7777 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->tod_example] continue=true
Dialplan: sofia/external/7777 Date/Time Match (FAIL) [tod_example] break=on-false
Dialplan: sofia/external/7777 parsing [Default->holiday_example] continue=true
Dialplan: sofia/external/7777 Date/Time Match (FAIL) [holiday_example] break=on-false
Dialplan: sofia/external/7777 parsing [Default->global-intercept] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [global-intercept] destination_number(8100) =~ /^\*886$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->group-intercept] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [group-intercept] destination_number(8100) =~ /^\*8$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->intercept-ext] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [intercept-ext] destination_number(8100) =~ /^\*\*(\d+)$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->redial] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [redial] destination_number(8100) =~ /^(redial|\*870)$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->global] continue=true
Dialplan: sofia/external/7777 Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/external/7777 Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never
Dialplan: sofia/external/7777 Absolute Condition [global]
Dialplan: sofia/external/7777 Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) 
Dialplan: sofia/external/7777 Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) 
Dialplan: sofia/external/7777 Action hash(insert/${domain_name}-last_dial/global/${uuid}) 
Dialplan: sofia/external/7777 Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) 
Dialplan: sofia/external/7777 parsing [Default->snom-demo-2] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [snom-demo-2] destination_number(8100) =~ /^\*9001$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->snom-demo-1] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [snom-demo-1] destination_number(8100) =~ /^\*9000$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->eavesdrop] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [eavesdrop] destination_number(8100) =~ /^\*88(\d{2,7})$|^\*0(.*)$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->eavesdrop] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [eavesdrop] destination_number(8100) =~ /^\*779$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->call_privacy] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [call_privacy] destination_number(8100) =~ /^\*67(\d+)$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->call_return] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [call_return] destination_number(8100) =~ /^\*69$|^869$|^lcr$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->del-group] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [del-group] destination_number(8100) =~ /^\*\*80(\d{2})$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->add-group] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [add-group] destination_number(8100) =~ /^\*\*81(\d{2})$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->call-group-simo] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [call-group-simo] destination_number(8100) =~ /^\*\*82(\d{2})$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->call-group-order] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [call-group-order] destination_number(8100) =~ /^\*83(\d{2})$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->extension-intercom] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [extension-intercom] destination_number(8100) =~ /^\*8(\d{2,7})$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->Local_Extension_Skinny] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [Local_Extension_Skinny] destination_number(8100) =~ /^(11[01][0-9])$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->send_to_voicemail] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [send_to_voicemail] destination_number(8100) =~ /^\*99(\d{2,7})$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->Conference] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [Conference] destination_number(8100) =~ /^8888$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->sc1000.10d] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [sc1000.10d] destination_number(8100) =~ /^(\d{10})$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->sc1000.7d] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [sc1000.7d] destination_number(8100) =~ /^(\d{7})$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->sc1000.d4] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [sc1000.d4] destination_number(8100) =~ /^(7\d{3})$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->Conference_Equation] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [Conference_Equation] destination_number(8100) =~ /^8800$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->101] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [101] destination_number(8100) =~ /^101$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->pizza_demo] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [pizza_demo] destination_number(8100) =~ /^(pizza|74992)$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->Talking Clock Time] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [Talking Clock Time] destination_number(8100) =~ /9170/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->Talking Clock Date] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [Talking Clock Date] destination_number(8100) =~ /9171/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->Talking Clock Date and Time] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [Talking Clock Date and Time] destination_number(8100) =~ /9172/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->Recordings] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [Recordings] destination_number(8100) =~ /^\*732$|^\*732673$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->group_dial_sales] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [group_dial_sales] destination_number(8100) =~ /^\*2000$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->group_dial_support] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [group_dial_support] destination_number(8100) =~ /^\*2001$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->group_dial_billing] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [group_dial_billing] destination_number(8100) =~ /^\*2002$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->operator] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [operator] destination_number(8100) =~ /^operator$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->vmain] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [vmain] destination_number(8100) =~ /^vmain$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->vmain1] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [vmain1] destination_number(8100) =~ /^vmain1$|^\*97$|^\*4000$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->vmain2] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [vmain2] destination_number(8100) =~ /^vmain2$|^\*98$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->sip_uri] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [sip_uri] destination_number(8100) =~ /^sip:(.*)$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->nb_conferences] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [nb_conferences] destination_number(8100) =~ /^\*(30\d{2})$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->wb_conferences] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [wb_conferences] destination_number(8100) =~ /^\*(31\d{2})$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->uwb_conferences] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [uwb_conferences] destination_number(8100) =~ /^\*(32\d{2})$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->cdquality_conferences] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [cdquality_conferences] destination_number(8100) =~ /^\*(33\d{2})$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->freeswitch_public_conf_via_sip] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [freeswitch_public_conf_via_sip] destination_number(8100) =~ /^\*9(888|8888|1616|3232)$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->mad_boss_intercom] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [mad_boss_intercom] destination_number(8100) =~ /^\*0911$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->mad_boss_intercom] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [mad_boss_intercom] destination_number(8100) =~ /^\*0912$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->mad_boss] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [mad_boss] destination_number(8100) =~ /^\*0913$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->ivr_demo] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [ivr_demo] destination_number(8100) =~ /^\*5000$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->dynamic_conference] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [dynamic_conference] destination_number(8100) =~ /^\*5001$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->rtp_multicast_page] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [rtp_multicast_page] destination_number(8100) =~ /^pagegroup$|^\*7243$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->park] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [park] destination_number(8100) =~ /^\*5900$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->unpark] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [unpark] destination_number(8100) =~ /^\*5901$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->valet_park_in] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [valet_park_in] destination_number(8100) =~ /^\*(6000)$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->valet_park_out] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [valet_park_out] destination_number(8100) =~ /^\*(60\d\d)$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->park] continue=false
Dialplan: sofia/external/7777 Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/external/7777 Regex (FAIL) [park] destination_number(8100) =~ /park\+(\d+)/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->unpark] continue=false
Dialplan: sofia/external/7777 Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/external/7777 Regex (FAIL) [unpark] destination_number(8100) =~ /^parking$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->park] continue=false
Dialplan: sofia/external/7777 Regex (PASS) [park] source(mod_sofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/external/7777 Regex (FAIL) [park] destination_number(8100) =~ /callpark/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->unpark] continue=false
Dialplan: sofia/external/7777 Regex (PASS) [unpark] source(mod_sofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/external/7777 Regex (FAIL) [unpark] destination_number(8100) =~ /pickup/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->wait] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [wait] destination_number(8100) =~ /^wait$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->fax_receive] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [fax_receive] destination_number(8100) =~ /^\*9178$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->fax_transmit] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [fax_transmit] destination_number(8100) =~ /^\*9179$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->ringback_180] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [ringback_180] destination_number(8100) =~ /^\*9180$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->ringback_183_uk_ring] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [ringback_183_uk_ring] destination_number(8100) =~ /^\*9181$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->ringback_183_music_ring] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [ringback_183_music_ring] destination_number(8100) =~ /^\*9182$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->ringback_post_answer_uk_ring] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [ringback_post_answer_uk_ring] destination_number(8100) =~ /^\*9183$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->ringback_post_answer_music] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [ringback_post_answer_music] destination_number(8100) =~ /^\*9184$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->ClueCon] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [ClueCon] destination_number(8100) =~ /^\*9191$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->show_info] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [show_info] destination_number(8100) =~ /^\*9192$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->video_record] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [video_record] destination_number(8100) =~ /^\*9193$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->video_playback] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [video_playback] destination_number(8100) =~ /^\*9194$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->delay_echo] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [delay_echo] destination_number(8100) =~ /^\*9195$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->echo] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [echo] destination_number(8100) =~ /^\*9196$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->milliwatt] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [milliwatt] destination_number(8100) =~ /^\*9197$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->tone_stream] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [tone_stream] destination_number(8100) =~ /^\*9198$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->zrtp_enrollement] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [zrtp_enrollement] destination_number(8100) =~ /^\*9787$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->hold_music] continue=false
Dialplan: sofia/external/7777 Regex (FAIL) [hold_music] destination_number(8100) =~ /^\*9664$/ break=on-false
Dialplan: sofia/external/7777 parsing [Default->Local_Extension] continue=false
Dialplan: sofia/external/7777 Regex (PASS) [Local_Extension] destination_number(8100) =~ /(^\d{2,7}$)/ break=on-false
Dialplan: sofia/external/7777 Action set(dialed_extension=8100) 
Dialplan: sofia/external/7777 Action export(dialed_extension=8100) 
Dialplan: sofia/external/7777 Action bind_meta_app(1 b s execute_extension::dx XML features) 
Dialplan: sofia/external/7777 Action bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/archive/${strftime(%Y)}/${strftime(%b)}/${strftime(%d)}/${uuid}.wav) 
Dialplan: sofia/external/7777 Action bind_meta_app(3 b s execute_extension::cf XML features) 
Dialplan: sofia/external/7777 Action set(ringback=${us-ring}) 
Dialplan: sofia/external/7777 Action set(transfer_ringback=local_stream://moh) 
Dialplan: sofia/external/7777 Action set(call_timeout=30) 
Dialplan: sofia/external/7777 Action set(hangup_after_bridge=true) 
Dialplan: sofia/external/7777 Action set(continue_on_fail=true) 
Dialplan: sofia/external/7777 Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}) 
Dialplan: sofia/external/7777 Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) 
Dialplan: sofia/external/7777 Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}) 
Dialplan: sofia/external/7777 Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) 
Dialplan: sofia/external/7777 Action bridge(user/${dialed_extension}@${domain_name}) 
Dialplan: sofia/external/7777 Action answer() 
Dialplan: sofia/external/7777 Action sleep(1000) 
Dialplan: sofia/external/7777 Action voicemail(default ${domain_name} ${dialed_extension}) 
2011-08-22 04:45:00.724680 [DEBUG] switch_core_state_machine.c:119 (sofia/external/7777) State Change CS_ROUTING -> CS_EXECUTE
2011-08-22 04:45:00.724680 [DEBUG] switch_core_session.c:1114 Send signal sofia/external/7777 [BREAK]
2011-08-22 04:45:00.724680 [DEBUG] switch_core_state_machine.c:364 (sofia/external/7777) State ROUTING going to sleep
2011-08-22 04:45:00.724680 [DEBUG] switch_core_state_machine.c:325 (sofia/external/7777) Running State Change CS_EXECUTE
2011-08-22 04:45:00.724680 [DEBUG] switch_core_state_machine.c:371 (sofia/external/7777) State EXECUTE
2011-08-22 04:45:00.724680 [DEBUG] mod_sofia.c:240 sofia/external/7777 SOFIA EXECUTE
2011-08-22 04:45:00.724680 [DEBUG] switch_core_state_machine.c:157 sofia/external/7777 Standard EXECUTE
EXECUTE sofia/external/7777 hash(insert/10.160.0.3-spymap/7777;phone-context=cdp.udp/39a3c5ec-834a-4c80-9b2e-b95e1b0f6fbd)
EXECUTE sofia/external/7777 hash(insert/10.160.0.3-last_dial/7777;phone-context=cdp.udp/8100)
EXECUTE sofia/external/7777 hash(insert/10.160.0.3-last_dial/global/39a3c5ec-834a-4c80-9b2e-b95e1b0f6fbd)
EXECUTE sofia/external/7777 set(RFC2822_DATE=Mon, 22 Aug 2011 04:45:00 +0300)
2011-08-22 04:45:00.725680 [DEBUG] mod_dptools.c:1060 sofia/external/7777 SET [RFC2822_DATE]=[Mon, 22 Aug 2011 04:45:00 +0300]
EXECUTE sofia/external/7777 set(dialed_extension=8100)
2011-08-22 04:45:00.725680 [DEBUG] mod_dptools.c:1060 sofia/external/7777 SET [dialed_extension]=[8100]
EXECUTE sofia/external/7777 export(dialed_extension=8100)
2011-08-22 04:45:00.725680 [DEBUG] switch_channel.c:965 EXPORT (export_vars) [dialed_extension]=[8100]
EXECUTE sofia/external/7777 bind_meta_app(1 b s execute_extension::dx XML features)
2011-08-22 04:45:00.726681 [INFO] switch_ivr_async.c:3014 Bound B-Leg: *1 execute_extension::dx XML features
EXECUTE sofia/external/7777 bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/archive/2011/Aug/22/39a3c5ec-834a-4c80-9b2e-b95e1b0f6fbd.wav)
2011-08-22 04:45:00.726681 [INFO] switch_ivr_async.c:3014 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/archive/2011/Aug/22/39a3c5ec-834a-4c80-9b2e-b95e1b0f6fbd.wav
EXECUTE sofia/external/7777 bind_meta_app(3 b s execute_extension::cf XML features)
2011-08-22 04:45:00.726681 [INFO] switch_ivr_async.c:3014 Bound B-Leg: *3 execute_extension::cf XML features
EXECUTE sofia/external/7777 set(ringback=%(2000, 4000, 440.0, 480.0))
2011-08-22 04:45:00.726681 [DEBUG] mod_dptools.c:1060 sofia/external/7777 SET [ringback]=[%(2000, 4000, 440.0, 480.0)]
EXECUTE sofia/external/7777 set(transfer_ringback=local_stream://moh)
2011-08-22 04:45:00.727682 [DEBUG] mod_dptools.c:1060 sofia/external/7777 SET [transfer_ringback]=[local_stream://moh]
EXECUTE sofia/external/7777 set(call_timeout=30)
2011-08-22 04:45:00.727682 [DEBUG] mod_dptools.c:1060 sofia/external/7777 SET [call_timeout]=[30]
EXECUTE sofia/external/7777 set(hangup_after_bridge=true)
2011-08-22 04:45:00.727682 [DEBUG] mod_dptools.c:1060 sofia/external/7777 SET [hangup_after_bridge]=[true]
EXECUTE sofia/external/7777 set(continue_on_fail=true)
2011-08-22 04:45:00.727682 [DEBUG] mod_dptools.c:1060 sofia/external/7777 SET [continue_on_fail]=[true]
EXECUTE sofia/external/7777 hash(insert/10.160.0.3-call_return/8100/7777;phone-context=cdp.udp)
EXECUTE sofia/external/7777 hash(insert/10.160.0.3-last_dial_ext/8100/39a3c5ec-834a-4c80-9b2e-b95e1b0f6fbd)
EXECUTE sofia/external/7777 set(called_party_callgroup=)
2011-08-22 04:45:00.728678 [DEBUG] mod_dptools.c:1060 sofia/external/7777 SET [called_party_callgroup]=[UNDEF]
EXECUTE sofia/external/7777 hash(insert/10.160.0.3-last_dial//39a3c5ec-834a-4c80-9b2e-b95e1b0f6fbd)
EXECUTE sofia/external/7777 bridge(user/8100 at 10.160.0.3)
2011-08-22 04:45:00.729678 [DEBUG] switch_channel.c:922 sofia/external/7777 EXPORTING[export_vars] [dialed_extension]=[8100] to event
2011-08-22 04:45:00.729678 [DEBUG] switch_ivr_originate.c:1873 Parsing global variables
2011-08-22 04:45:00.729678 [DEBUG] switch_channel.c:922 sofia/external/7777 EXPORTING[export_vars] [dialed_extension]=[8100] to event
2011-08-22 04:45:00.729678 [DEBUG] switch_ivr_originate.c:1873 Parsing global variables
2011-08-22 04:45:00.729678 [DEBUG] switch_event.c:1170 Parsing variable [presence_id]=[8100 at 10.160.0.3]
2011-08-22 04:45:00.730678 [NOTICE] switch_channel.c:816 New Channel sofia/internal/sip:8100 at 10.160.1.212:5060 [cb0e41a2-c78f-492d-bf62-0253dc747736]
2011-08-22 04:45:00.730678 [DEBUG] mod_sofia.c:4305 (sofia/internal/sip:8100 at 10.160.1.212:5060) State Change CS_NEW -> CS_INIT
2011-08-22 04:45:00.730678 [DEBUG] switch_core_session.c:1114 Send signal sofia/internal/sip:8100 at 10.160.1.212:5060 [BREAK]
2011-08-22 04:45:00.730678 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/sip:8100 at 10.160.1.212:5060) Running State Change CS_INIT
2011-08-22 04:45:00.730678 [DEBUG] switch_core_state_machine.c:361 (sofia/internal/sip:8100 at 10.160.1.212:5060) State INIT
2011-08-22 04:45:00.730678 [DEBUG] mod_sofia.c:84 sofia/internal/sip:8100 at 10.160.1.212:5060 SOFIA INIT
2011-08-22 04:45:00.731677 [DEBUG] mod_sofia.c:124 (sofia/internal/sip:8100 at 10.160.1.212:5060) State Change CS_INIT -> CS_ROUTING
2011-08-22 04:45:00.731677 [DEBUG] switch_core_session.c:1114 Send signal sofia/internal/sip:8100 at 10.160.1.212:5060 [BREAK]
2011-08-22 04:45:00.731677 [DEBUG] switch_core_state_machine.c:361 (sofia/internal/sip:8100 at 10.160.1.212:5060) State INIT going to sleep
2011-08-22 04:45:00.731677 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/sip:8100 at 10.160.1.212:5060) Running State Change CS_ROUTING
2011-08-22 04:45:00.731677 [DEBUG] switch_channel.c:1667 (sofia/internal/sip:8100 at 10.160.1.212:5060) Callstate Change DOWN -> RINGING
2011-08-22 04:45:00.731677 [DEBUG] sofia.c:4761 Channel sofia/internal/sip:8100 at 10.160.1.212:5060 entering state [calling][0]
2011-08-22 04:45:00.731677 [DEBUG] switch_core_state_machine.c:364 (sofia/internal/sip:8100 at 10.160.1.212:5060) State ROUTING
2011-08-22 04:45:00.731677 [DEBUG] mod_sofia.c:147 sofia/internal/sip:8100 at 10.160.1.212:5060 SOFIA ROUTING
2011-08-22 04:45:00.731677 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/sip:8100 at 10.160.1.212:5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2011-08-22 04:45:00.731677 [DEBUG] switch_core_session.c:1114 Send signal sofia/internal/sip:8100 at 10.160.1.212:5060 [BREAK]
2011-08-22 04:45:00.731677 [DEBUG] switch_core_state_machine.c:364 (sofia/internal/sip:8100 at 10.160.1.212:5060) State ROUTING going to sleep
2011-08-22 04:45:00.731677 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/sip:8100 at 10.160.1.212:5060) Running State Change CS_CONSUME_MEDIA
2011-08-22 04:45:00.731677 [DEBUG] switch_core_state_machine.c:383 (sofia/internal/sip:8100 at 10.160.1.212:5060) State CONSUME_MEDIA
2011-08-22 04:45:00.731677 [DEBUG] switch_core_state_machine.c:383 (sofia/internal/sip:8100 at 10.160.1.212:5060) State CONSUME_MEDIA going to sleep
2011-08-22 04:45:00.741675 [DEBUG] sofia.c:4761 Channel sofia/internal/sip:8100 at 10.160.1.212:5060 entering state [terminated][415]
2011-08-22 04:45:00.741675 [DEBUG] switch_channel.c:2562 (sofia/internal/sip:8100 at 10.160.1.212:5060) Callstate Change RINGING -> HANGUP
2011-08-22 04:45:00.741675 [NOTICE] sofia.c:5407 Hangup sofia/internal/sip:8100 at 10.160.1.212:5060 [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
2011-08-22 04:45:00.741675 [DEBUG] switch_channel.c:2578 Send signal sofia/internal/sip:8100 at 10.160.1.212:5060 [KILL]
2011-08-22 04:45:00.741675 [DEBUG] switch_core_session.c:1114 Send signal sofia/internal/sip:8100 at 10.160.1.212:5060 [BREAK]
2011-08-22 04:45:00.741675 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/sip:8100 at 10.160.1.212:5060) Running State Change CS_HANGUP
2011-08-22 04:45:00.741675 [DEBUG] switch_ivr_originate.c:3299 Originate Resulted in Error Cause: 79 [SERVICE_NOT_IMPLEMENTED]
2011-08-22 04:45:00.741675 [ERR] switch_ivr_originate.c:2447 Cannot create outgoing channel of type [user] cause: [SERVICE_NOT_IMPLEMENTED]
2011-08-22 04:45:00.741675 [DEBUG] switch_ivr_originate.c:3299 Originate Resulted in Error Cause: 79 [SERVICE_NOT_IMPLEMENTED]
2011-08-22 04:45:00.741675 [INFO] mod_dptools.c:2647 Originate Failed.  Cause: SERVICE_NOT_IMPLEMENTED
EXECUTE sofia/external/7777 answer()
2011-08-22 04:45:00.742676 [DEBUG] sofia_glue.c:3022 AUDIO RTP [sofia/external/7777] 10.160.0.3 port 23546 -> 10.160.0.205 port 5200 codec: 8 ms: 20
2011-08-22 04:45:00.742676 [DEBUG] switch_rtp.c:1623 Starting timer [soft] 160 bytes per 20ms
2011-08-22 04:45:00.742676 [DEBUG] switch_core_state_machine.c:565 (sofia/internal/sip:8100 at 10.160.1.212:5060) State HANGUP
2011-08-22 04:45:00.742676 [DEBUG] mod_sofia.c:451 sofia/internal/sip:8100 at 10.160.1.212:5060 Overriding SIP cause 501 with 415 from the other leg
2011-08-22 04:45:00.742676 [DEBUG] mod_sofia.c:457 Channel sofia/internal/sip:8100 at 10.160.1.212:5060 hanging up, cause: SERVICE_NOT_IMPLEMENTED
2011-08-22 04:45:00.743676 [DEBUG] switch_core_state_machine.c:46 sofia/internal/sip:8100 at 10.160.1.212:5060 Standard HANGUP, cause: SERVICE_NOT_IMPLEMENTED
2011-08-22 04:45:00.743676 [DEBUG] switch_core_state_machine.c:565 (sofia/internal/sip:8100 at 10.160.1.212:5060) State HANGUP going to sleep
2011-08-22 04:45:00.743676 [DEBUG] switch_core_state_machine.c:356 (sofia/internal/sip:8100 at 10.160.1.212:5060) State Change CS_HANGUP -> CS_REPORTING
2011-08-22 04:45:00.743676 [DEBUG] switch_core_session.c:1114 Send signal sofia/internal/sip:8100 at 10.160.1.212:5060 [BREAK]
2011-08-22 04:45:00.743676 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/sip:8100 at 10.160.1.212:5060) Running State Change CS_REPORTING
2011-08-22 04:45:00.743676 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/sip:8100 at 10.160.1.212:5060) State REPORTING
2011-08-22 04:45:00.743676 [DEBUG] switch_core_state_machine.c:53 sofia/internal/sip:8100 at 10.160.1.212:5060 Standard REPORTING, cause: SERVICE_NOT_IMPLEMENTED
2011-08-22 04:45:00.743676 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/sip:8100 at 10.160.1.212:5060) State REPORTING going to sleep
2011-08-22 04:45:00.743676 [DEBUG] switch_core_state_machine.c:350 (sofia/internal/sip:8100 at 10.160.1.212:5060) State Change CS_REPORTING -> CS_DESTROY
2011-08-22 04:45:00.743676 [DEBUG] switch_core_session.c:1114 Send signal sofia/internal/sip:8100 at 10.160.1.212:5060 [BREAK]
2011-08-22 04:45:00.743676 [DEBUG] switch_core_session.c:1286 Session 530 (sofia/internal/sip:8100 at 10.160.1.212:5060) Locked, Waiting on external entities
2011-08-22 04:45:00.743676 [NOTICE] switch_core_session.c:1304 Session 530 (sofia/internal/sip:8100 at 10.160.1.212:5060) Ended
2011-08-22 04:45:00.743676 [NOTICE] switch_core_session.c:1306 Close Channel sofia/internal/sip:8100 at 10.160.1.212:5060 [CS_DESTROY]
2011-08-22 04:45:00.743676 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/sip:8100 at 10.160.1.212:5060) Callstate Change HANGUP -> DOWN
2011-08-22 04:45:00.743676 [DEBUG] switch_core_state_machine.c:457 (sofia/internal/sip:8100 at 10.160.1.212:5060) Running State Change CS_DESTROY
2011-08-22 04:45:00.743676 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/sip:8100 at 10.160.1.212:5060) State DESTROY
2011-08-22 04:45:00.743676 [DEBUG] mod_sofia.c:362 sofia/internal/sip:8100 at 10.160.1.212:5060 SOFIA DESTROY
2011-08-22 04:45:00.743676 [DEBUG] switch_core_state_machine.c:60 sofia/internal/sip:8100 at 10.160.1.212:5060 Standard DESTROY
2011-08-22 04:45:00.743676 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/sip:8100 at 10.160.1.212:5060) State DESTROY going to sleep
2011-08-22 04:45:00.743676 [DEBUG] sofia_glue.c:3284 Set 2833 dtmf send payload to 101
2011-08-22 04:45:00.744676 [DEBUG] sofia_glue.c:3289 Set 2833 dtmf receive payload to 101
2011-08-22 04:45:00.744676 [DEBUG] mod_sofia.c:681 Local SDP sofia/external/7777:
v=0
o=FreeSWITCH 1313953954 1313953955 IN IP4 10.160.0.3
s=FreeSWITCH
c=IN IP4 10.160.0.3
t=0 0
m=audio 23546 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2011-08-22 04:45:00.744676 [DEBUG] switch_core_session.c:707 Send signal sofia/external/7777 [BREAK]
2011-08-22 04:45:00.744676 [DEBUG] switch_channel.c:2829 (sofia/external/7777) Callstate Change RINGING -> ACTIVE
2011-08-22 04:45:00.744676 [NOTICE] mod_dptools.c:930 Channel [sofia/external/7777] has been answered
2011-08-22 04:45:00.744676 [DEBUG] sofia.c:4761 Channel sofia/external/7777 entering state [completed][200]
EXECUTE sofia/external/7777 sleep(1000)
2011-08-22 04:45:00.746677 [DEBUG] sofia.c:4761 Channel sofia/external/7777 entering state [ready][200]
2011-08-22 04:45:00.945642 [DEBUG] switch_rtp.c:3082 Correct ip/port confirmed.
EXECUTE sofia/external/7777 voicemail(default 10.160.0.3 8100)
2011-08-22 04:45:01.866492 [DEBUG] switch_ivr_play_say.c:67 No language specified - Using [ru]
2011-08-22 04:45:01.877489 [DEBUG] switch_ivr_play_say.c:244 Handle play-file:[voicemail/vm-person.wav] (ru:ru)
2011-08-22 04:45:01.877489 [DEBUG] switch_ivr_play_say.c:1279 Codec Activated L16 at 8000hz 1 channels 20ms
2011-08-22 04:45:02.220434 [DEBUG] switch_channel.c:2562 (sofia/external/7777) Callstate Change ACTIVE -> HANGUP
2011-08-22 04:45:02.220434 [NOTICE] sofia.c:538 Hangup sofia/external/7777 [CS_EXECUTE] [NORMAL_CLEARING]
2011-08-22 04:45:02.220434 [DEBUG] switch_channel.c:2578 Send signal sofia/external/7777 [KILL]
2011-08-22 04:45:02.221602 [DEBUG] switch_core_session.c:1114 Send signal sofia/external/7777 [BREAK]
2011-08-22 04:45:02.226433 [DEBUG] switch_ivr_play_say.c:1649 done playing file
2011-08-22 04:45:02.326421 [DEBUG] switch_ivr_play_say.c:244 Handle say:[8100] (ru:ru)
2011-08-22 04:45:02.428408 [DEBUG] switch_core_session.c:2057 sofia/external/7777 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
2011-08-22 04:45:02.428408 [DEBUG] switch_core_state_machine.c:371 (sofia/external/7777) State EXECUTE going to sleep
2011-08-22 04:45:02.428408 [DEBUG] switch_core_state_machine.c:325 (sofia/external/7777) Running State Change CS_HANGUP
2011-08-22 04:45:02.429464 [DEBUG] switch_core_state_machine.c:565 (sofia/external/7777) State HANGUP
2011-08-22 04:45:02.429464 [DEBUG] mod_sofia.c:451 sofia/external/7777 Overriding SIP cause 480 with 200 from the other leg
2011-08-22 04:45:02.429464 [DEBUG] mod_sofia.c:457 Channel sofia/external/7777 hanging up, cause: NORMAL_CLEARING
2011-08-22 04:45:02.429464 [DEBUG] switch_core_state_machine.c:46 sofia/external/7777 Standard HANGUP, cause: NORMAL_CLEARING
2011-08-22 04:45:02.429464 [DEBUG] switch_core_state_machine.c:565 (sofia/external/7777) State HANGUP going to sleep
2011-08-22 04:45:02.429464 [DEBUG] switch_core_state_machine.c:356 (sofia/external/7777) State Change CS_HANGUP -> CS_REPORTING
2011-08-22 04:45:02.429464 [DEBUG] switch_core_session.c:1114 Send signal sofia/external/7777 [BREAK]
2011-08-22 04:45:02.429464 [DEBUG] switch_core_state_machine.c:325 (sofia/external/7777) Running State Change CS_REPORTING
2011-08-22 04:45:02.429464 [DEBUG] switch_core_state_machine.c:625 (sofia/external/7777) State REPORTING
2011-08-22 04:45:02.473396 [DEBUG] switch_core_state_machine.c:53 sofia/external/7777 Standard REPORTING, cause: NORMAL_CLEARING
2011-08-22 04:45:02.473396 [DEBUG] switch_core_state_machine.c:625 (sofia/external/7777) State REPORTING going to sleep
2011-08-22 04:45:02.473396 [DEBUG] switch_core_state_machine.c:350 (sofia/external/7777) State Change CS_REPORTING -> CS_DESTROY
2011-08-22 04:45:02.473396 [DEBUG] switch_core_session.c:1114 Send signal sofia/external/7777 [BREAK]
2011-08-22 04:45:02.473396 [DEBUG] switch_core_session.c:1286 Session 529 (sofia/external/7777) Locked, Waiting on external entities
2011-08-22 04:45:02.473396 [NOTICE] switch_core_session.c:1304 Session 529 (sofia/external/7777) Ended
2011-08-22 04:45:02.473396 [NOTICE] switch_core_session.c:1306 Close Channel sofia/external/7777 [CS_DESTROY]
2011-08-22 04:45:02.473396 [DEBUG] switch_core_state_machine.c:454 (sofia/external/7777) Callstate Change HANGUP -> DOWN


-- 
Best regards,
 sucsession                          mailto:norstar at bigmir.net



From jcgpoza at gmail.com  Wed Aug 31 14:48:18 2011
From: jcgpoza at gmail.com (Dissident)
Date: Wed, 31 Aug 2011 03:48:18 -0700 (PDT)
Subject: [Freeswitch-users] Condition based on custom sip header
In-Reply-To: <CAKbxfG_dCxQ9qfmtPJMPfDJYr6ZFLyttmxDJmyyRFczhvCN1-g@mail.gmail.com>
References: <F5B5EF8F78C0BC4BB417E1DBB9D7F7D0180A75B7CD@jjex01.jajah.dublin>
	<CAKbxfG-pD89e2p+JtYi__kngCt1azbNkCvmnHDUsLGfjbuB-qA@mail.gmail.com>
	<F5B5EF8F78C0BC4BB417E1DBB9D7F7D0180A75B951@jjex01.jajah.dublin>
	<CAKbxfG_dCxQ9qfmtPJMPfDJYr6ZFLyttmxDJmyyRFczhvCN1-g@mail.gmail.com>
Message-ID: <1314787698150-6745987.post@n2.nabble.com>

Hello Alex, 

I had to face the same situation weeks ago...
Yes, It's shame that many hardware/software makers are ignoring the RFCs and
doing things their own way but what can you do...
Here is how I sorted it out.

http://freeswitch-users.2379917.n2.nabble.com/Sip-Headers-advice-Not-parsing-properly-td6606461.html

good luck


--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Condition-based-on-custom-sip-header-tp6738108p6745987.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From mrene_lists at avgs.ca  Wed Aug 31 14:54:00 2011
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Wed, 31 Aug 2011 12:54:00 +0200
Subject: [Freeswitch-users] Calling FMS with mod_rtmp
In-Reply-To: <2C6A32EE4B454BC9897CA8FCB17B5547@e1705>
References: <B2C62E9A-3C98-41AF-9500-C6EAA45AF849@networxonline.com>
	<2C6A32EE4B454BC9897CA8FCB17B5547@e1705>
Message-ID: <E3DB0966-705A-458A-A455-9F3437553686@avgs.ca>

Hi,

RTMP is the protocol we support. We do NOT support RTMFP (which is the udp-based heavily crypted and obfuscated peer to peer protocol that goes with the recent flash players).

Right now we only have a server mode RTMP handler so you would need to connect to FS from FMS using NetConnection/NetStream like you would do on the client. You also need to call a few functions to authenticate and make a call so that you actually play something, those are the same as the sample flex client included in the freeswitch tree (everything in the sample is done in javascript and the function calls are proxied to freeswitch through the flash player).

You essentially need login and makeCall, then as soon as you playback something you'll get the audio from the call (we only support one active call per RTMP connection: you can have calls on hold and do three-ways, but only one audio stream will be active).

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 2011-08-30, at 6:36 PM, Madovsky wrote:

> you can't .
> to register directly from flash client
> only RTMFP protocol is allowed
> 
> ----- Original Message ----- 
> From: "Kevin Reeves" <kevin at networxonline.com>
> To: <freeswitch-users at lists.freeswitch.org>
> Sent: Monday, August 29, 2011 5:54 PM
> Subject: [Freeswitch-users] Calling FMS with mod_rtmp
> 
> 
>> I'm interested in making a call from inside Freeswitch out to Flash Media 
>> Server using mod_rtmp.  I'd like to do this so I can attach audio from a 
>> phone call into FMS.  If I'm reading the documentation correctly, the only 
>> way to attach the 2 is to have FMS register with Freeswitch.
>> 
>> Any suggestions you have would be greatly appreciated.
>> 
>> Thanks,
>> Kevin
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org 
> 
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From kemen04 at gmail.com  Wed Aug 31 15:09:28 2011
From: kemen04 at gmail.com (Travis Kemen)
Date: Wed, 31 Aug 2011 06:09:28 -0500
Subject: [Freeswitch-users] Setup with Bandwidth.com
In-Reply-To: <0E3F1549A4FB4287A7F6B78A8EC2761F@ccs.local>
References: <0E3F1549A4FB4287A7F6B78A8EC2761F@ccs.local>
Message-ID: <CAG_VuqYe_muhL8ZFH+=Wty46591MWfMWNa_QS5ss_vMjtK-gjQ@mail.gmail.com>

Here is the profile I use to connect to them, you could just put a dummy
password in being you don't have one. They do use port 5060 so you will need
to change your external port to be 5060 and your internal to something else
if you want to do it this way.

<include>
  <gateway name="bandwidth.com">
    <param name="username" value="PHONENUMBER"/>
    <param name="password" value="PASSWORD"/>
    <param name="proxy" value="IP"/>
    <param name="expire-seconds" value="600"/>
    <param name="register" value="false"/>
    <param name="retry_seconds" value="30"/>
  </gateway>
</include>
<include>
  <gateway name="bandwidth.com-backup">
    <param name="username" value="PHONENUMBER"/>
    <param name="password" value="PASSWORD"/>
    <param name="proxy" value="IP"/>
    <param name="expire-seconds" value="600"/>
    <param name="register" value="false"/>
    <param name="retry_seconds" value="30"/>
  </gateway>
</include>

On Tue, Aug 30, 2011 at 3:25 PM, Wayne <wayne at hamilton.net> wrote:

> Does anyone have a setup with bandwidth.com?
>
> The example on the wiki has a username and password. I didn't receive a
> username and password form bandwidth.com. Just setup a sip peer and have
> an
> IP address. If I send calls to freeswitch from the peer I get Rejected by
> acl "domains". Falling back to Digest auth. Looks like they are sending it
> on port 5060.
> I'm sure the rejected by acl is the most asked question but I have been
> unable to find the answer.
>
> Thanks Wayne
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110831/eac8da60/attachment.html 

From mickkemp at gmail.com  Wed Aug 31 15:21:06 2011
From: mickkemp at gmail.com (Michael Kemp)
Date: Wed, 31 Aug 2011 12:21:06 +0100
Subject: [Freeswitch-users] mod_conference - 2-digit DTMF and events
Message-ID: <CAOFe-q46f7Dh31fO=maDDcE-xoFscsv3xPq_UTB1YuBiu9NvNA@mail.gmail.com>

Hi Moises

Thank you for implementing this feature request so quickly!  I will
endeavour to install and test it and let you know the results this
week.

Best Wishes

Michael Kemp


From michal.bielicki at seventhsignal.de  Wed Aug 31 17:20:30 2011
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Wed, 31 Aug 2011 15:20:30 +0200
Subject: [Freeswitch-users] Variables in Dialplan - Problem with getting
	Variables from User Directory
In-Reply-To: <4E5D50BD.8010208@omeco.de>
References: <4E5D50BD.8010208@omeco.de>
Message-ID: <A1E3DB7E-64B5-4E74-89FF-7F87E7F8D501@seventhsignal.de>


Am 30.08.2011 um 23:06 schrieb Silvio Escher:

> Hi there,
> 
> iam really getting crazy these days ;-)
> 
> I've changed our Freeswitch Version some Weeks ago ( cannot remember exactly but proably from 1.0.6 ) to the git Head. ).
> Since this Change ( or better since the followed Config File Adaptions ;-) ) i noticed that i cannot use User Variables/Params inside the XML-Dialplan anymore.
> 
> Some (relevant?! ;) ) Copy&Paste ..
> 
> my Userentry
> 
> <include>
>   <user id="99">
>     <params>
>       <param name="password" value="something"/>
>       <param name="vm-mailto" value="maybe_me at omeco.de"/>
>       <param name="vm-mailfrom" value="noreply at omeco.de"/>
>       <param name="vm-enabled" value="true"/> 
>       <param name="vm-password" value="user-choose"/>
>       <param name="vm-email-all-messages" value="false"/>
>       <param name="vm-notify-email-all-messages" value="true"/>
>       <param name="vm-keep-local-after-email" value="true"/>
>     </params>
>     <variables>
>       <variable name="outbound_caller_id_name" value="Omeco GmbH"/>                         
>       <variable name="outbound_caller_id_number" value="06316803540"/>
>       <variable name="effective_caller_id_name" value="Silvio Escher"/>                                                                           
>       <variable name="effective_caller_id_number" value="40"/>                                                                                      
>       <variable name="callgroup" value="1"/>
>     </variables>
>   </user>
> </include>
> 
> some dp snippet..
> 
> before the change something like
> 
> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/>
> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/>
> 
> was working fine .. 
> 
> Actually iam using 
> 
> <action application="set" data="effective_caller_id_name=${user_data(${username}@${domain_name} var outbound_caller_id_name)}"/>
> <action application="set" data="effective_caller_id_number=${user_data(${username}@${domain_name} var outbound_caller_id_number)}"/> 

if you set the user with set_user there is no requirement to use user_data since it should get the params for the respective user automatically.

> 
> or an 
> 
> <action application="set_user" data="$1@${domain_name}"/>
> 
> to get some or all Variables.
> 
> But i Noticed that the "set_user" Thingie just gets me the Variables - not the Params.
> So i've still Issues with mod_voicemail ( Mailto is undef )

params are checked on call, so only there when required and checked each time, vars are fixed.
> 
> 2011-08-30 22:02:32.904211 [DEBUG] switch_utils.c:709 Emailed data to [(null)]
> 2011-08-30 22:02:32.904211 [DEBUG] mod_voicemail.c:2809 Sending notify message to (null)
> 
> Maybe i has something todo with an missing auth or wrong Domain or whatever .. but i have no clue where to look further.
> 
> Any Help ( from little Hints till complete Miracle Solve ;-) ) is actually very welcome.
> 
> Thanks in Advance,
> Silvio
> 
> 
> -- 
> Silvio Escher
> omeco GmbH
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Michal Bielicki
Gesch?ftsf?hrer / CEO

Seventh Signal Ltd. & Co. KG
Weigandufer 45, B?ro 115, D-12059 Berlin
Voice: +49 30 60988730

Amtsgericht Charlottenburg HRA 44413 B
Ust.-ID: DE266981999
Gesch?ftsf?hrer: Michal Bielicki
Pers?nlich Haftende Gesellschafterin:
Seventh Signal Ltd, 69 Great Hampton St. Birmingham, 
B18 6EW, GB, Company Nr.: 06889439
WWW.: http://www.seventhsignal.de



----

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110831/eb8618d1/attachment-0001.html 

From alex at jajah.com  Wed Aug 31 18:29:56 2011
From: alex at jajah.com (Alex Massover)
Date: Wed, 31 Aug 2011 17:29:56 +0300
Subject: [Freeswitch-users] Condition based on custom sip header
In-Reply-To: <1314787698150-6745987.post@n2.nabble.com>
References: <F5B5EF8F78C0BC4BB417E1DBB9D7F7D0180A75B7CD@jjex01.jajah.dublin>
	<CAKbxfG-pD89e2p+JtYi__kngCt1azbNkCvmnHDUsLGfjbuB-qA@mail.gmail.com>
	<F5B5EF8F78C0BC4BB417E1DBB9D7F7D0180A75B951@jjex01.jajah.dublin>
	<CAKbxfG_dCxQ9qfmtPJMPfDJYr6ZFLyttmxDJmyyRFczhvCN1-g@mail.gmail.com>
	<1314787698150-6745987.post@n2.nabble.com>
Message-ID: <F5B5EF8F78C0BC4BB417E1DBB9D7F7D0180A8B6E34@jjex01.jajah.dublin>

Hi,

That's great! Exactly what I need.

Thanks!

> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
> users-bounces at lists.freeswitch.org] On Behalf Of Dissident
> Sent: ????? 31 ?????? 2011 13:48
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Condition based on custom sip header
> 
> Hello Alex,
> 
> I had to face the same situation weeks ago...
> Yes, It's shame that many hardware/software makers are ignoring the
> RFCs and
> doing things their own way but what can you do...
> Here is how I sorted it out.
> 
> http://freeswitch-users.2379917.n2.nabble.com/Sip-Headers-advice-Not-
> parsing-properly-td6606461.html
> 
> good luck
> 
> 
> --
> View this message in context: http://freeswitch-
> users.2379917.n2.nabble.com/Condition-based-on-custom-sip-header-
> tp6738108p6745987.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
> 
> This mail was received via Mail-SeCure System.
> 


This mail was sent via Mail-SeCure System.




From infos at madovsky.org  Wed Aug 31 18:37:09 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 31 Aug 2011 10:37:09 -0400
Subject: [Freeswitch-users] ringing status and mod_rtmp
References: <909E89D1B3434E64B75BFC53726CE0A9@e1705>
	<D8C0BFEB-00F9-4F40-A341-FB7398050B9F@avgs.ca>
Message-ID: <5ADC1262106946BF80D040E408D66E0F@e1705>

Hi Mathieu,

so it means that the leg A must generate a fake ringing even
if leg B is not reached ?

  ----- Original Message ----- 
  From: Mathieu Rene 
  To: FreeSWITCH Users Help 
  Sent: Tuesday, August 30, 2011 5:46 AM
  Subject: Re: [Freeswitch-users] ringing status and mod_rtmp


  There is a callback for the call's state, it should be set to RINGING whenever the other side rings. 


  Mathieu Rene
  Avant-Garde Solutions Inc
  Office: + 1 (514) 664-1044 x100
  Cell: +1 (514) 664-1044 x200
  mrene at avgs.ca








  On 2011-08-25, at 4:38 PM, Madovsky wrote:


    Hi folks,

    is there any callback event for ringing status of leg B with mod_rtmp ?

    Thanks

    Franck

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





------------------------------------------------------------------------------



  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110831/8888d855/attachment.html 

From mrene_lists at avgs.ca  Wed Aug 31 18:54:18 2011
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Wed, 31 Aug 2011 16:54:18 +0200
Subject: [Freeswitch-users] ringing status and mod_rtmp
In-Reply-To: <5ADC1262106946BF80D040E408D66E0F@e1705>
References: <909E89D1B3434E64B75BFC53726CE0A9@e1705>
	<D8C0BFEB-00F9-4F40-A341-FB7398050B9F@avgs.ca>
	<5ADC1262106946BF80D040E408D66E0F@e1705>
Message-ID: <A69AC365-F2AE-41F2-BE46-8C143726E6A1@avgs.ca>

Yeah, like on this page: http://wiki.freeswitch.org/wiki/Custom_Ring_Back_Tones

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 2011-08-31, at 4:37 PM, Madovsky wrote:

> Hi Mathieu,
>  
> so it means that the leg A must generate a fake ringing even
> if leg B is not reached ?
>  
> ----- Original Message -----
> From: Mathieu Rene
> To: FreeSWITCH Users Help
> Sent: Tuesday, August 30, 2011 5:46 AM
> Subject: Re: [Freeswitch-users] ringing status and mod_rtmp
> 
> There is a callback for the call's state, it should be set to RINGING whenever the other side rings. 
> 
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
> 
> 
> 
> 
> On 2011-08-25, at 4:38 PM, Madovsky wrote:
> 
>> Hi folks,
>>  
>> is there any callback event for ringing status of leg B with mod_rtmp ?
>>  
>> Thanks
>>  
>> Franck
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110831/597acf66/attachment.html 

From mrene_lists at avgs.ca  Wed Aug 31 18:55:35 2011
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Wed, 31 Aug 2011 16:55:35 +0200
Subject: [Freeswitch-users] FreeSwitch Utilizes 100% of CPU Sometimes
	When User Quits Conference
In-Reply-To: <007701cc6307$6e569270$4b03b750$@Merioles.net>
References: <002601cc6259$b8b8ec40$2a2ac4c0$@Merioles.net>
	<CAKbxfG8SAzOYxvi==cCEcxzq5a0QUiYV3MztxizMC71y3RbtLQ@mail.gmail.com>
	<007701cc6307$6e569270$4b03b750$@Merioles.net>
Message-ID: <EC1BACFC-8B9F-42F5-963C-F3BD4CCD9169@avgs.ca>

Hi,

Can you try the latest version to see if this is fixed?

Thanks,

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 2011-08-25, at 11:14 AM, Erwin Merioles wrote:

> Thanks for the reply!
> 
> I think I've found the problem. When our "host" ends the meeting, it sends a
> disconnect event to all participants. The SWITCH_POLLOUT event happens when
> the participants tries to end the meeting simultaneously with the host.
> 
> I think it is the server's way of doing a "cleanup" and tries to send any
> remaining data to whoever is left in the conference -- which in our case is
> none. This results to an infinite loop of tries.
> 
> The problem was fixed on our end by delaying the disconnection of the
> participants a bit.
> 
> Regards,
> 
> Erwin D. Merioles
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
> Minessale
> Sent: Thursday, August 25, 2011 12:01 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] FreeSwitch Utilizes 100% of CPU Sometimes
> When User Quits Conference
> 
> Really you should be reporting bugs to http://jira.freeswitch.org Are you
> only having this problem with mod_rtmp (its only 2 months old) Most likely
> you have created a new condition that the author has not taken into account.
> Ideally you should file it on jira under mod_rtmp and attach a back trace
> from a core dump produced by gcore.
> 
> 
> 
> On Wed, Aug 24, 2011 at 7:31 AM, Erwin Merioles <erwin at merioles.net> wrote:
>> Hey guys,
>> 
>> 
>> 
>> I?ve been having trouble with FreeSwitch for quite some time. We?re 
>> trying to use mod_rtmp to add sound to one of our applications,
> www.321meet.com.
>> Unfortunately, FreeSwitch?s CPU usage spikes when the host ( the first 
>> one to join the conference ), quits or closes the browser window. I?ve 
>> checked and this always happen when the following line is called ?
>> 
>> 
>> 
>> 2011-08-24 12:24:28.057200 [NOTICE] rtmp_tcp.c:73 Pollout: true
>> 
>> 
>> 
>> FS Console Log follows :
>> 
>> 
>> 
>> 2011-08-24 12:22:59.177294 [NOTICE] mod_rtmp.c:743 New RTMP session 
>> [4a97320c-e50d-4ed3-a59c-4aef799d379d]
>> 
>> 2011-08-24 12:22:59.477278 [NOTICE] rtmp_sig.c:121 Sent connect reply
>> 
>> 2011-08-24 12:23:15.597196 [INFO] rtmp_sig.c:136 Replied to 
>> createStream (1)
>> 
>> 2011-08-24 12:23:16.237201 [INFO] rtmp_sig.c:274 Got publish on stream 1.
>> 
>> 2011-08-24 12:23:17.177293 [INFO] rtmp_sig.c:136 Replied to 
>> createStream (2)
>> 
>> 2011-08-24 12:23:17.177293 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled 
>> control packet (type=0x3)
>> 
>> 2011-08-24 12:23:17.457195 [WARNING] sofia.c:4403 Ping succeeded 
>> voip9.telsome.com with code 404 - count -1/1/1, state UP
>> 
>> 2011-08-24 12:23:17.457195 [INFO] rtmp_sig.c:136 Replied to 
>> createStream (3)
>> 
>> 2011-08-24 12:23:17.457195 [NOTICE] switch_channel.c:897 New Channel
>> rtmp/default/3213533 [8e7ccce8-172c-4b91-9183-1ad9d2f0e6dd]
>> 
>> 2011-08-24 12:23:17.457195 [ERR] rtmp_sig.c:305 Couldn't create call.
>> 
>> 2011-08-24 12:23:17.457195 [WARNING] sofia.c:4403 Ping succeeded 
>> testin with code 404 - count -1/1/1, state UP
>> 
>> 2011-08-24 12:23:17.497199 [INFO] mod_dialplan_xml.c:336 Processing
>> <0000000000>->3213533 in context default
>> 
>> 2011-08-24 12:23:17.497199 [NOTICE] mod_rtmp.c:497 Channel 
>> [rtmp/default/3213533] has been answered
>> 
>> 2011-08-24 12:23:17.497199 [INFO] mod_conference.c:6644 using channel 
>> sound
>> prefix: /usr/local/freeswitch/sounds/en/us/callie
>> 
>> 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 
>> rtmp/default/3213533 binding '0' to 'mute'
>> 
>> 2011-08-24 12:23:17.597289 [INFO] switch_ivr_async.c:164 Digit parser
>> mod_conference: Setting realm to conf
>> 
>> 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 
>> rtmp/default/3213533 binding '*' to 'deaf mute'
>> 
>> 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 
>> rtmp/default/3213533 binding '9' to 'energy up'
>> 
>> 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 
>> rtmp/default/3213533 binding '8' to 'energy equ'
>> 
>> 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 
>> rtmp/default/3213533 binding '7' to 'energy dn'
>> 
>> 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 
>> rtmp/default/3213533 binding '3' to 'vol talk up'
>> 
>> 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 
>> rtmp/default/3213533 binding '2' to 'vol talk zero'
>> 
>> 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 
>> rtmp/default/3213533 binding '1' to 'vol talk dn'
>> 
>> 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 
>> rtmp/default/3213533 binding '6' to 'vol listen up'
>> 
>> 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 
>> rtmp/default/3213533 binding '5' to 'vol listen zero'
>> 
>> 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 
>> rtmp/default/3213533 binding '4' to 'vol listen dn'
>> 
>> 2011-08-24 12:23:17.597289 [INFO] mod_conference.c:7007 
>> rtmp/default/3213533 binding '#' to 'hangup'
>> 
>> 2011-08-24 12:23:17.797195 [INFO] rtmp_sig.c:159 Sending audio
>> 
>> 2011-08-24 12:23:17.797195 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled 
>> control packet (type=0x3)
>> 
>> 2011-08-24 12:23:18.117290 [INFO] rtmp_sig.c:274 Got publish on stream 3.
>> 
>> 2011-08-24 12:23:36.697195 [ERR] rtmp.c:678 Read error
>> 
>> 2011-08-24 12:23:36.697195 [NOTICE] mod_rtmp.c:788 RTMP session ended 
>> [4a97320c-e50d-4ed3-a59c-4aef799d379d]
>> 
>> 2011-08-24 12:23:36.697195 [NOTICE] mod_rtmp.c:803 Hangup
>> rtmp/default/3213533 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]
>> 
>> 2011-08-24 12:23:36.777294 [NOTICE] switch_core_session.c:1347 Session 
>> 1
>> (rtmp/default/3213533) Ended
>> 
>> 2011-08-24 12:23:36.777294 [NOTICE] switch_core_session.c:1349 Close 
>> Channel
>> rtmp/default/3213533 [CS_DESTROY]
>> 
>> 2011-08-24 12:23:48.237202 [NOTICE] mod_rtmp.c:743 New RTMP session 
>> [3ff4786f-7cae-4653-879a-ae95a9d50742]
>> 
>> 2011-08-24 12:23:48.557204 [NOTICE] rtmp_sig.c:121 Sent connect reply
>> 
>> 2011-08-24 12:23:55.357293 [NOTICE] mod_rtmp.c:743 New RTMP session 
>> [af763930-9054-4c76-a0eb-9c351e75949d]
>> 
>> 2011-08-24 12:23:55.757290 [NOTICE] rtmp_sig.c:121 Sent connect reply
>> 
>> 2011-08-24 12:23:58.957195 [INFO] rtmp_sig.c:136 Replied to 
>> createStream (1)
>> 
>> 2011-08-24 12:23:59.517204 [INFO] rtmp_sig.c:274 Got publish on stream 1.
>> 
>> 2011-08-24 12:24:00.557201 [INFO] rtmp_sig.c:136 Replied to 
>> createStream (2)
>> 
>> 2011-08-24 12:24:00.557201 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled 
>> control packet (type=0x3)
>> 
>> 2011-08-24 12:24:00.557201 [INFO] rtmp_sig.c:136 Replied to 
>> createStream (3)
>> 
>> 2011-08-24 12:24:00.857222 [NOTICE] switch_channel.c:897 New Channel
>> rtmp/default/3213533 [5de55413-5cb9-4c18-8b65-9c438833449f]
>> 
>> 2011-08-24 12:24:00.857222 [ERR] rtmp_sig.c:305 Couldn't create call.
>> 
>> 2011-08-24 12:24:00.857222 [INFO] mod_dialplan_xml.c:336 Processing
>> <0000000000>->3213533 in context default
>> 
>> 2011-08-24 12:24:00.857222 [NOTICE] mod_rtmp.c:497 Channel 
>> [rtmp/default/3213533] has been answered
>> 
>> 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:6644 using channel 
>> sound
>> prefix: /usr/local/freeswitch/sounds/en/us/callie
>> 
>> 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 
>> rtmp/default/3213533 binding '0' to 'mute'
>> 
>> 2011-08-24 12:24:00.857222 [INFO] switch_ivr_async.c:164 Digit parser
>> mod_conference: Setting realm to conf
>> 
>> 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 
>> rtmp/default/3213533 binding '*' to 'deaf mute'
>> 
>> 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 
>> rtmp/default/3213533 binding '9' to 'energy up'
>> 
>> 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 
>> rtmp/default/3213533 binding '8' to 'energy equ'
>> 
>> 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 
>> rtmp/default/3213533 binding '7' to 'energy dn'
>> 
>> 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 
>> rtmp/default/3213533 binding '3' to 'vol talk up'
>> 
>> 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 
>> rtmp/default/3213533 binding '2' to 'vol talk zero'
>> 
>> 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 
>> rtmp/default/3213533 binding '1' to 'vol talk dn'
>> 
>> 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 
>> rtmp/default/3213533 binding '6' to 'vol listen up'
>> 
>> 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 
>> rtmp/default/3213533 binding '5' to 'vol listen zero'
>> 
>> 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 
>> rtmp/default/3213533 binding '4' to 'vol listen dn'
>> 
>> 2011-08-24 12:24:00.857222 [INFO] mod_conference.c:7007 
>> rtmp/default/3213533 binding '#' to 'hangup'
>> 
>> 2011-08-24 12:24:01.197247 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled 
>> control packet (type=0x3)
>> 
>> 2011-08-24 12:24:01.197247 [INFO] rtmp_sig.c:274 Got publish on stream 3.
>> 
>> 2011-08-24 12:24:01.197247 [INFO] rtmp_sig.c:159 Sending audio
>> 
>> 2011-08-24 12:24:13.637291 [INFO] rtmp_sig.c:136 Replied to 
>> createStream (1)
>> 
>> 2011-08-24 12:24:14.157289 [INFO] rtmp_sig.c:274 Got publish on stream 1.
>> 
>> 2011-08-24 12:24:17.057195 [INFO] rtmp_sig.c:136 Replied to 
>> createStream (2)
>> 
>> 2011-08-24 12:24:17.057195 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled 
>> control packet (type=0x3)
>> 
>> 2011-08-24 12:24:17.057195 [INFO] rtmp_sig.c:136 Replied to 
>> createStream (3)
>> 
>> 2011-08-24 12:24:17.057195 [NOTICE] switch_channel.c:897 New Channel
>> rtmp/default/3213533 [4e6c35ce-ec56-4ef1-816a-b94704efbfbd]
>> 
>> 2011-08-24 12:24:17.057195 [ERR] rtmp_sig.c:305 Couldn't create call.
>> 
>> 2011-08-24 12:24:17.057195 [INFO] mod_dialplan_xml.c:336 Processing
>> <0000000000>->3213533 in context default
>> 
>> 2011-08-24 12:24:17.057195 [NOTICE] mod_rtmp.c:497 Channel 
>> [rtmp/default/3213533] has been answered
>> 
>> 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 
>> rtmp/default/3213533 binding '0' to 'mute'
>> 
>> 2011-08-24 12:24:17.057195 [INFO] switch_ivr_async.c:164 Digit parser
>> mod_conference: Setting realm to conf
>> 
>> 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 
>> rtmp/default/3213533 binding '*' to 'deaf mute'
>> 
>> 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 
>> rtmp/default/3213533 binding '9' to 'energy up'
>> 
>> 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 
>> rtmp/default/3213533 binding '8' to 'energy equ'
>> 
>> 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 
>> rtmp/default/3213533 binding '7' to 'energy dn'
>> 
>> 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 
>> rtmp/default/3213533 binding '3' to 'vol talk up'
>> 
>> 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 
>> rtmp/default/3213533 binding '2' to 'vol talk zero'
>> 
>> 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 
>> rtmp/default/3213533 binding '1' to 'vol talk dn'
>> 
>> 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 
>> rtmp/default/3213533 binding '6' to 'vol listen up'
>> 
>> 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 
>> rtmp/default/3213533 binding '5' to 'vol listen zero'
>> 
>> 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 
>> rtmp/default/3213533 binding '4' to 'vol listen dn'
>> 
>> 2011-08-24 12:24:17.057195 [INFO] mod_conference.c:7007 
>> rtmp/default/3213533 binding '#' to 'hangup'
>> 
>> 2011-08-24 12:24:17.497196 [WARNING] rtmp.c:99 [amfnumber=2] Unhandled 
>> control packet (type=0x3)
>> 
>> 2011-08-24 12:24:17.497196 [INFO] rtmp_sig.c:274 Got publish on stream 3.
>> 
>> 2011-08-24 12:24:17.497196 [INFO] rtmp_sig.c:159 Sending audio
>> 
>> 2011-08-24 12:24:27.657196 [ERR] rtmp.c:678 Read error
>> 
>> 2011-08-24 12:24:27.657196 [NOTICE] mod_rtmp.c:788 RTMP session ended 
>> [3ff4786f-7cae-4653-879a-ae95a9d50742]
>> 
>> 2011-08-24 12:24:27.657196 [NOTICE] mod_rtmp.c:803 Hangup
>> rtmp/default/3213533 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]
>> 
>> 2011-08-24 12:24:27.677201 [NOTICE] switch_core_session.c:1347 Session 
>> 2
>> (rtmp/default/3213533) Ended
>> 
>> 2011-08-24 12:24:27.677201 [NOTICE] switch_core_session.c:1349 Close 
>> Channel
>> rtmp/default/3213533 [CS_DESTROY]
>> 
>> 2011-08-24 12:24:28.057200 [NOTICE] rtmp_tcp.c:73 Pollout: true
>> 
>> 
>> 
>> Which results to :
>> 
>> 
>> 
>>   PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+  COMMAND
>> 
>> 1608 root      -2 -10  191m  20m 6588 R 94.3  3.5   5:39.50 fs
>> 
>> 
>> 
>> Any ideas? Help is VERY much appreciated. TIA!
>> 
>> 
>> 
>> Regards,
>> 
>> 
>> 
>> Erwin D. Merioles
>> 
>> 
>> 
>> merioles.net
>> 
>> +63 922 837 9466 | +63 917 501 1010 | +1 760 670 3241
>> 
>> aY!M : erwin_merioles | Skype : erwin.merioles
>> 
>> This message (including any attachments) contains information that may 
>> be confidential. Unless you are the intended recipient (or is 
>> authorized to receive for the intended recipient), you may not read, 
>> print, retain, use, copy, distribute, or disclose to anyone, any 
>> information contained here. If you have received this in error, please 
>> advise the sender by reply e-mail, and delete all copies of the original
> message (including attachments).
>> 
>> 
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>> rs
>> http://www.freeswitch.org
>> 
>> 
> 
> 
> 
> --
> Anthony Minessale II
> 
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From chris at ghosttelecom.com  Wed Aug 31 19:16:43 2011
From: chris at ghosttelecom.com (Chris Martineau)
Date: Wed, 31 Aug 2011 16:16:43 +0100
Subject: [Freeswitch-users] rtp natting
In-Reply-To: <CAKDfjgdhB_uMd4pMgUPzdKfsh8J3erE+yKc-Su_bzhcLpcTp6w@mail.gmail.com>
References: <1D10AB188D6CCA46BB4369E3268E36EF309C1A@SVR01.ghosttelecom.local>
	<CAKDfjgdhB_uMd4pMgUPzdKfsh8J3erE+yKc-Su_bzhcLpcTp6w@mail.gmail.com>
Message-ID: <1D10AB188D6CCA46BB4369E3268E36EF309D96@SVR01.ghosttelecom.local>

Hi,

Thanks for the quick response.

Okay that seems fair however for calls I wish to route through Freeswitch I would prefer not to add the additional step of the rtpproxy app for natting the rtp. Can you confirm whether Freeswitch will do the same natting function as rtpproxy by substituting the signalled rtp return port with the actual port from incoming rtp packets from the source. This would simplify things greatly.

Regards

Chris

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kristian Kielhofner
Sent: 26 August 2011 19:01
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] rtp natting

Keep doing what you are doing with OpenSIPS.  Add FreeSWITCH for
things it's good at:

- SBC functionality
- Hosted voice apps (like conferencing)

FreeSWITCH is good at those other things too but the SERs are still
king of scale.

On Fri, Aug 26, 2011 at 10:08 AM, Chris Martineau
<chris at ghosttelecom.com> wrote:
> Hi,
>
>
>
> I am new to freeswitch and wish to bolt it into our existing network to
> provide conferencing services.
>
>
>
> Currently we just use an opensips/rtpproxy configuration for simple proxy
> switching.
>
>
>
> However looking at the features of freeswitch it seems to me that I could
> replace my entire setup with just the freeswitch platform. Would that be a
> fair comment?
>
>
>
> If so then there a number of things that I am struggling to find clearly
> defined in the documentation.
>
>
>
> 1.?????? Currently our rtp proxy scenario issues a port to come back on but
> waits for incoming rtp packets to determine the exact port to return rtp on
> when the caller is behind a nat. Does freeswitch do this, does it need to be
> configured to work that way? Freeswitch would be on a public address.
>
> 2.?????? Currently opensips stores all user info in a mysql database and
> with 500000+ users is easy to manipulate. How do you deal with such a large
> user database in a pure xml environment such as freeswitch?
>
>
>
> Many thanks for any help you can offer.
>
>
>
> Regards
>
>
>
> Chris
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Kristian Kielhofner


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


From msc at freeswitch.org  Wed Aug 31 19:18:18 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 31 Aug 2011 08:18:18 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today
Message-ID: <CAKzWOxWx76E8XyBhDZ=cuJqvudUNHgi5d04XaD+H50xUw=RKGg@mail.gmail.com>

Hello all,

Today's agenda is here:
http://wiki.freeswitch.org/wiki/FS_weekly_2011_08_31

I'd like to talk about the concept of "giving back" which I touched upon in
this article:
http://www.freeswitch.org/node/339

Talk to you at 1PM EST/10AM Pacific/1700 UTC!

-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110831/1a31a9c0/attachment.html 

From peter.schrock at gmail.com  Wed Aug 31 04:25:42 2011
From: peter.schrock at gmail.com (Peter Schrock)
Date: Tue, 30 Aug 2011 17:25:42 -0700
Subject: [Freeswitch-users] OS X ppc leopard "make" error message
Message-ID: <CA+PZoDQf__ux2XeibP9Nx=Awk8G90W6Q3gJt5uTtxiki=aymxw@mail.gmail.com>

I am new to FreeSWITCH (sort of, tried it a while back and dropped it).
I have a powerpc Apple running Leopard on it. I followed the directions by
Mario G and managed to get all the way up to where he discusses error's with
FLITE. I managed to go without the error message because I followed the
instructions on how to fix it. However, I have come up with this error
message and can't seem to make any sense of it. Help Please.

make[8]: *** No rule to make target `tport/libtport.la', needed by `
libsofia-sip-ua.la'.  Stop.
make[7]: *** [all-recursive] Error 1
Making all in packages
make[6]: *** [all-recursive] Error 1
make[5]: *** [all] Error 2
make[4]: *** [/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/
libsofia-sip-ua.la] Error 2
make[3]: *** [mod_sofia-all] Error 1
make[2]: *** [all-recursive] Error 1
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2

Peter
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110830/adac43ec/attachment-0001.html 

From ales.zelenik at it-tim.si  Wed Aug 31 13:45:46 2011
From: ales.zelenik at it-tim.si (=?UTF-8?Q?Ale=C5=A1?= Zelenik)
Date: Wed, 31 Aug 2011 11:45:46 +0200
Subject: [Freeswitch-users] Start FreeSWITCH as a Windows service with
	-nonat option
Message-ID: <1314783946.2217.12.camel@ales-Latitude-D630>

Hello all,

I am having trouble starting fs as a windows service with -nonat option

If I run from command prompt "FreeSwitchConsole -nonat" it works ok, but
I am left with open window in logged-in session.

Command to run as a service is "...path...\FreeSwitchConsole.exe
-service FreeSWITCH", which can be also verified with "sc qc FreeSWITCH"

If I want to add an argument -nonat with command:
sc config FreeSWITCH binPath= "C:\Program Files (x86)\FreeSWITCH
\FreeSwitchConsole.exe -nonat -service FreeSWITCH"

service won't start anymore

also, fs cannot be installed as a service with optional arguments, eg
FreeSwitchConsole -install -nonat

Anyone managed to run fs like this?

Thanks,
-- 
Ales Zelenik,





From peter.schrock at gmail.com  Wed Aug 31 15:25:59 2011
From: peter.schrock at gmail.com (Peter Schrock)
Date: Wed, 31 Aug 2011 04:25:59 -0700
Subject: [Freeswitch-users] OS X PPC Leopard error on "make"
Message-ID: <8319344044664188306@unknownmsgid>

I am new to FreeSWITCH (sort of, tried it a while back and dropped it).
I have a powerpc Apple running Leopard on it. I followed the directions by
Mario G and managed to get all the way up to where he discusses error's with
FLITE. I managed to go without the error message because I followed the
instructions on how to fix it. However, I have come up with this error
message and can't seem to make any sense of it. Help Please.

Here is the error I get:
make[8]: *** No rule to make target `tport/libtport.la', needed by `
libsofia-sip-ua.la'.  Stop.
make[7]: *** [all-recursive] Error 1
Making all in packages
make[6]: *** [all-recursive] Error 1
make[5]: *** [all] Error 2
make[4]: *** [/usr/local/src/freeswitch/libs/sofia-sip/libsofia-sip-ua/
libsofia-sip-ua.la] Error 2
make[3]: *** [mod_sofia-all] Error 1
make[2]: *** [all-recursive] Error 1
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2

Peter
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110831/cc53e488/attachment-0001.html 

From dhairya.blogs at gmail.com  Wed Aug 31 17:26:06 2011
From: dhairya.blogs at gmail.com (Dhairya Vora)
Date: Wed, 31 Aug 2011 18:56:06 +0530
Subject: [Freeswitch-users] CDR
Message-ID: <CA+P_fE9Q=1t+RcO8GKCsjobusGhov+cykrFkW6ESAkaxqskPLw@mail.gmail.com>

I need to show detailed information of a particular calluuid. I tried
parsing .xml file generated by
freeswitch(/usr/local/freeswitch/log/xml_cdr/archive/) but it does not show
*the time at which a digit was pressed*.
I checked plivo log(/usr/src/plivo/tmp/plivo-outbound.log) also but it
stores the log for all the calls in one file only.

I want this kind of information:
TIMEEVENTDETAILS2011-08-31 10:51:57PLAY
please_leave_your_message_after_the_beep.wav2011-08-31
10:52:04RECORD1_1.wav2011-08-31
10:52:08PLAYyour_recorded_message_is.wav2011-08-31
10:52:10PLAY1_1.wav*2011-08-31
10:52:21**DIGIT PRESSED**0
*2011-08-31 10:52:22PLAYthank_you.wav2011-08-31 10:52:23PLAY
your_message_has_been_recorded_successfully.wav
using .xml file generated by
freeswitch(/usr/local/freeswitch/log/xml_cdr/archive/) I could parse all the
information *other than "DIGIT PRESSED"* information.

Can anyone help ?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110831/226328f1/attachment-0001.html 

From tim.mcqueen at gmail.com  Wed Aug 31 19:07:36 2011
From: tim.mcqueen at gmail.com (Tim McQueen)
Date: Wed, 31 Aug 2011 10:07:36 -0500
Subject: [Freeswitch-users] problem using mod_managed under linux
In-Reply-To: <30572.1314700796@ccs.covici.com>
References: <30572.1314700796@ccs.covici.com>
Message-ID: <CAKE0JBAEBLL=Mtp0PxMieZY5hAt2B56ZUKM-C+QwyxrV=RgC6Q@mail.gmail.com>

I'm having the same problem.  (http://pastebin.freeswitch.org/17216)

Have you tried the Demo.csx to see if it will even run? In my case it
won't.  It is my opinon that it's because the FreeSwitch.Managed.dll is
dependent specifically on C# 4.0, since the same code will run correctly on
Windows.

I sent a message out to the freeswitch-devs group but I don't think it was
approved by the moderator, because I haven't seen it come back through the
list.  Someone on IRC said that mod_managed isn't maintained anymore, but I
see activity in FishEye from last week.

On Tue, Aug 30, 2011 at 5:39 AM, <covici at ccs.covici.com> wrote:

> Hi.  I am trying to use mod_managed on my linux box -- gentoo
> distribution -- and its driving me ...
>
>
> I wrote a very simple program for a test.  The program just sets a dtmf
> callback and then streams a file and writes a log entry when it sees the
> dtmf.
>
> Under Windows 7 net framework 4, it works just fine.  Under Linux, using
> mono versions 2.8.2 qand 2.10.4, I get the following exception which I
> will put in a pastebin.
>
> http://pastebin.freeswitch.org/17232
>
> And the app is in this one
>
> http://pastebin.freeswitch.org/17233
>
> I have tried using gmcs and dmcs thinking it may be a .net framework
> issue, and if I do it wrong, I can bring down fs itself, but the best I
> can get under Linux is the exception shown above.
>
> Thanks in advance for anything you can come up with on this.
>
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>         John Covici
>         covici at ccs.covici.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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110831/8d5e38c1/attachment-0001.html 

From mgg at giagnocavo.net  Wed Aug 31 19:35:57 2011
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Wed, 31 Aug 2011 11:35:57 -0400
Subject: [Freeswitch-users] problem using mod_managed under linux
In-Reply-To: <CAKE0JBAEBLL=Mtp0PxMieZY5hAt2B56ZUKM-C+QwyxrV=RgC6Q@mail.gmail.com>
References: <30572.1314700796@ccs.covici.com>
	<CAKE0JBAEBLL=Mtp0PxMieZY5hAt2B56ZUKM-C+QwyxrV=RgC6Q@mail.gmail.com>
Message-ID: <03351FCC6082174C8534AB714B8258A5E43D8C24@mse17be1.mse17.exchange.ms>

It's probably more related to some cross-appdomain/serialization stuff that's specific to Mono. I wrote mod_managed against Mono 2.4 or so, and I think last time I ran it was 2.6, and only on CentOS 5. Probably something changed and is triggering a bug in mod_managed in the newer builds.


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tim McQueen
Sent: Wednesday, August 31, 2011 9:08 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] problem using mod_managed under linux

I'm having the same problem.  (http://pastebin.freeswitch.org/17216)

Have you tried the Demo.csx to see if it will even run? In my case it won't.  It is my opinon that it's because the FreeSwitch.Managed.dll is dependent specifically on C# 4.0, since the same code will run correctly on Windows.

I sent a message out to the freeswitch-devs group but I don't think it was approved by the moderator, because I haven't seen it come back through the list.  Someone on IRC said that mod_managed isn't maintained anymore, but I see activity in FishEye from last week.
On Tue, Aug 30, 2011 at 5:39 AM, <covici at ccs.covici.com<mailto:covici at ccs.covici.com>> wrote:
Hi.  I am trying to use mod_managed on my linux box -- gentoo
distribution -- and its driving me ...


I wrote a very simple program for a test.  The program just sets a dtmf
callback and then streams a file and writes a log entry when it sees the
dtmf.

Under Windows 7 net framework 4, it works just fine.  Under Linux, using
mono versions 2.8.2 qand 2.10.4, I get the following exception which I
will put in a pastebin.

http://pastebin.freeswitch.org/17232

And the app is in this one

http://pastebin.freeswitch.org/17233

I have tried using gmcs and dmcs thinking it may be a .net framework
issue, and if I do it wrong, I can bring down fs itself, but the best I
can get under Linux is the exception shown above.

Thanks in advance for anything you can come up with on this.

--
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

        John Covici
        covici at ccs.covici.com<mailto:covici at ccs.covici.com>


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

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110831/9d443bc2/attachment.html 

From jmesquita at freeswitch.org  Wed Aug 31 19:48:13 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 31 Aug 2011 12:48:13 -0300
Subject: [Freeswitch-users] =?windows-1252?q?FreeSWITCH=99_in_Mexico?=
In-Reply-To: <CAOQhXxEOcRVU7wEYimw=1pj_L+CzrR8jiAxcEMmY50RRtS6Xig@mail.gmail.com>
References: <CABH8QX3icB5P-OiEzkqMkwcVtBCmcBNGr=P-iLyD_w2nAOmjpg@mail.gmail.com>
	<CAOQhXxEOcRVU7wEYimw=1pj_L+CzrR8jiAxcEMmY50RRtS6Xig@mail.gmail.com>
Message-ID: <CABH8QX20Q4Pm9xQFiOfe2Uyg0tqPPk1euyPzcC64FEcw7pU_PQ@mail.gmail.com>

Thank you very much Gerardo.

I appreciate your attendance as well as your interaction. I hope you
enjoyed.

Regards,
Jo?o Mesquita



2011/8/30 Gerardo Barajas <gerardo.barajas at gmail.com>

> Nice having you in Mexico.
> Really nice talk.
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110831/f5c3838a/attachment.html 

From peter.olsson at visionutveckling.se  Wed Aug 31 20:09:52 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 31 Aug 2011 18:09:52 +0200
Subject: [Freeswitch-users] Start FreeSWITCH as a Windows service
	with	-nonat option
In-Reply-To: <1314783946.2217.12.camel@ales-Latitude-D630>
References: <1314783946.2217.12.camel@ales-Latitude-D630>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF0D8@cooper>

Change the command line for the service, from "-service FreeSWITCH" to "-service FreeSWITCH -nonat" - I believe that the extra options must come after the service argument.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Ale? Zelenik [ales.zelenik at it-tim.si]
Skickat: den 31 augusti 2011 11:45
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] Start FreeSWITCH as a Windows service with     -nonat option

Hello all,

I am having trouble starting fs as a windows service with -nonat option

If I run from command prompt "FreeSwitchConsole -nonat" it works ok, but
I am left with open window in logged-in session.

Command to run as a service is "...path...\FreeSwitchConsole.exe
-service FreeSWITCH", which can be also verified with "sc qc FreeSWITCH"

If I want to add an argument -nonat with command:
sc config FreeSWITCH binPath= "C:\Program Files (x86)\FreeSWITCH
\FreeSwitchConsole.exe -nonat -service FreeSWITCH"

service won't start anymore

also, fs cannot be installed as a service with optional arguments, eg
FreeSwitchConsole -install -nonat

Anyone managed to run fs like this?

Thanks,
--
Ales Zelenik,





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

!DSPAM:4e5e50df32763458912694!



From msc at freeswitch.org  Wed Aug 31 20:27:05 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 31 Aug 2011 09:27:05 -0700
Subject: [Freeswitch-users] FS behind an IPCOP firewall
In-Reply-To: <CAK2JmwpAHY-wR5rWFZ-YRKWj4aO9J4PO8oa3bxNsPLiFL3U8Cg@mail.gmail.com>
References: <CAK2JmwpAHY-wR5rWFZ-YRKWj4aO9J4PO8oa3bxNsPLiFL3U8Cg@mail.gmail.com>
Message-ID: <CAKzWOxWnapPvMj=FVQ=Dh4cskN6jfh4ULyjYuBRMfYtV+_zdJw@mail.gmail.com>

What do you see on the FS console? Make sure you are in debug loglevel.
Also, you may want to turn on SIP trace as well so that you can observe the
actual SIP messages being sent and received.

-MC

On Tue, Aug 30, 2011 at 7:11 AM, Praveen Ray <pray at theprossergroup.com>wrote:

> Hi
> Here's is my setup. It's not that complicated.
>
>             172.168.128.0/24 subnet
> FS <-------------------------------------------> IPCOP Firewall
>                                                                    |
> |                 10.10.0.0/24 subnet
> Extension 1000 <--------------------------------->|
> |<-------------------------------------------- Extension 1001
> (172.168.128.188)
> (10.10.0.170)
>
> Firewall is set to allow all UDP traffic in the port rane 1024-65000
> between these two subnets.
>
> Both 1000 and 1001 register successfully with FS. I can make outgoing calls
> (using vitelity) using both. I can call 1000 from 1001 ok. However,
> calling 1001 from 1000 doesn't work. It goes right to 1001's VM. It's as if
> FS doesn't know 1001 is registered. My config is almost same as what
> comes with vanilla FS install (except I defined outgoing vitelity and
> 1000.xml and 1001.xml extensions).
>
> Any obvious config I'm missing? Thanks in advance.
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110831/ea8667cb/attachment.html 

From msc at freeswitch.org  Wed Aug 31 20:29:50 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 31 Aug 2011 09:29:50 -0700
Subject: [Freeswitch-users] DTMF from Nortel BCM 400 to Freeswitch
In-Reply-To: <CANDXbUz8S-i21HcDDyuHJ5oe1HAgzi-8pc-MhNzE_nmkA+JSmA@mail.gmail.com>
References: <CANDXbUyv9Q9i9FuajPs8rG7KaQagC3JG2Ji7zeJS9uZHYsxv+w@mail.gmail.com>
	<CANDXbUz8S-i21HcDDyuHJ5oe1HAgzi-8pc-MhNzE_nmkA+JSmA@mail.gmail.com>
Message-ID: <CAKzWOxV2u9XoU596c=H67Z2RbF5oEw3_Ze6mRp1V5-MkUdT0FA@mail.gmail.com>

On Tue, Aug 30, 2011 at 8:47 PM, Ray Pang <rpang88 at gmail.com> wrote:

> I?ve been unable to get DTMF to work from BCM 400 to Freeswitch.  I've
>> tried all DTMF settings with no luck.
>>
>>
>>
>
When you say that you've tried all DTMF settings, what does that mean? Also,
is the BCM converting in-band DTMFs into SIP INFO messages?

-MC
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110831/e9ae9fa2/attachment.html 

From covici at ccs.covici.com  Wed Aug 31 21:05:04 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Wed, 31 Aug 2011 13:05:04 -0400
Subject: [Freeswitch-users] problem using mod_managed under linux
In-Reply-To: <CAKE0JBAEBLL=Mtp0PxMieZY5hAt2B56ZUKM-C+QwyxrV=RgC6Q@mail.gmail.com>
References: <30572.1314700796@ccs.covici.com>
	<CAKE0JBAEBLL=Mtp0PxMieZY5hAt2B56ZUKM-C+QwyxrV=RgC6Q@mail.gmail.com>
Message-ID: <19062.1314810304@ccs.covici.com>

I tried to remake the .dll using dmcs which is supposed to be for
framework v4, but it did not help.

Tim McQueen <tim.mcqueen at gmail.com> wrote:

> I'm having the same problem.  (http://pastebin.freeswitch.org/17216)
> 
> Have you tried the Demo.csx to see if it will even run? In my case it
> won't.  It is my opinon that it's because the FreeSwitch.Managed.dll is
> dependent specifically on C# 4.0, since the same code will run correctly on
> Windows.
> 
> I sent a message out to the freeswitch-devs group but I don't think it was
> approved by the moderator, because I haven't seen it come back through the
> list.  Someone on IRC said that mod_managed isn't maintained anymore, but I
> see activity in FishEye from last week.
> 
> On Tue, Aug 30, 2011 at 5:39 AM, <covici at ccs.covici.com> wrote:
> 
> > Hi.  I am trying to use mod_managed on my linux box -- gentoo
> > distribution -- and its driving me ...
> >
> >
> > I wrote a very simple program for a test.  The program just sets a dtmf
> > callback and then streams a file and writes a log entry when it sees the
> > dtmf.
> >
> > Under Windows 7 net framework 4, it works just fine.  Under Linux, using
> > mono versions 2.8.2 qand 2.10.4, I get the following exception which I
> > will put in a pastebin.
> >
> > http://pastebin.freeswitch.org/17232
> >
> > And the app is in this one
> >
> > http://pastebin.freeswitch.org/17233
> >
> > I have tried using gmcs and dmcs thinking it may be a .net framework
> > issue, and if I do it wrong, I can bring down fs itself, but the best I
> > can get under Linux is the exception shown above.
> >
> > Thanks in advance for anything you can come up with on this.
> >
> > --
> > Your life is like a penny.  You're going to lose it.  The question is:
> > How do
> > you spend it?
> >
> >         John Covici
> >         covici at ccs.covici.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
> >
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From rpang88 at gmail.com  Wed Aug 31 21:11:25 2011
From: rpang88 at gmail.com (Ray Pang)
Date: Wed, 31 Aug 2011 10:11:25 -0700
Subject: [Freeswitch-users] DTMF from Nortel BCM 400 to Freeswitch
In-Reply-To: <CAKzWOxV2u9XoU596c=H67Z2RbF5oEw3_Ze6mRp1V5-MkUdT0FA@mail.gmail.com>
References: <CANDXbUyv9Q9i9FuajPs8rG7KaQagC3JG2Ji7zeJS9uZHYsxv+w@mail.gmail.com>
	<CANDXbUz8S-i21HcDDyuHJ5oe1HAgzi-8pc-MhNzE_nmkA+JSmA@mail.gmail.com>
	<CAKzWOxV2u9XoU596c=H67Z2RbF5oEw3_Ze6mRp1V5-MkUdT0FA@mail.gmail.com>
Message-ID: <CANDXbUzevDVxbo2ZHzVr1b5vtiVyM38m++WeYKvs8+8MGSKdTA@mail.gmail.com>

Yes.  Freeswitch is receiving SIP INFO messages (not sure if BCM is
converting or sending it native).  I've tried on Freeswitch using suggested
options with no avail.

Example: <param name="dtmf-type" value="info"/> or <variable
direction="inbound|outbound|both" name="dtmf_type" value="info"> and any
other variations I that i was able to find.
Thanks.
RP


On Wed, Aug 31, 2011 at 9:29 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Tue, Aug 30, 2011 at 8:47 PM, Ray Pang <rpang88 at gmail.com> wrote:
>
>> I?ve been unable to get DTMF to work from BCM 400 to Freeswitch.  I've
>>> tried all DTMF settings with no luck.
>>>
>>>
>>>
>>
> When you say that you've tried all DTMF settings, what does that mean?
> Also, is the BCM converting in-band DTMFs into SIP INFO messages?
>
> -MC
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110831/c2dbd648/attachment.html 

From danlanweb at gmail.com  Wed Aug 31 21:46:45 2011
From: danlanweb at gmail.com (Dan Lan)
Date: Wed, 31 Aug 2011 10:46:45 -0700
Subject: [Freeswitch-users] Connecting Call Between Two SIP Trunks
In-Reply-To: <CAGRr05-WAih_JvibyS6km2ON2gnYDaZbgbVebioQvzxaHgjfZg@mail.gmail.com>
References: <CAAuJ8MBmErPiahLHAWOFJb4jLCQX-ePc14HFT07hTw5wr7EOjQ@mail.gmail.com>
	<CAGRr05-WAih_JvibyS6km2ON2gnYDaZbgbVebioQvzxaHgjfZg@mail.gmail.com>
Message-ID: <CAAuJ8MAWniFh-npGA2NOhQd-SsXzZSWaw=vOwP+XHkz4UA-pYg@mail.gmail.com>

Hi, Brad:
This is Dan Lan. Thanks I got it working. Yes, I have to use "bridge" to
connect to Sofia external trunk.
I will put my configuration here for mail list archive purpose, so other
people can refer to this in the future.
1. First, make an extension profile in the dialplan\public folder as
following.
2. This dialplan will recognize the incoming IP trunk and bridge the call to
second IP trunk
3. The reason I disable DTMF, is that I hear 2 DTMF tone (maybe one in-band,
one RFC2833) so I disable one, and it works for me.

<include>
   <extension name="trunkin_to_trunkout">
     <condition field="network_addr" expression="^xxx\.xxx\.xxx\.xxx$"/>
<--  The first Trunk's IP address -->
 <condition field="destination_number" expression="^(\d+)$">
 <action application = "set" data="dtmf_type=none"/>
 <action application="bridge" data="sofia/external/$1 at xxx.xxx.xxx.xxx"/>
<-- The second Trunk's IP address -->
     </condition>
  </extension>
</include>
On Tue, Aug 30, 2011 at 4:10 PM, Brad Mina <brad at tech21.com> wrote:

> 1. Username and password are manditory in the XML - you don't have to put
> anything that makes sense, just fill it in with your DID or something and
> keep register=false and those details will never do anything.
>
> 2. bridge would be the proper tool, you might have to mess around with
> proxy media or make sure proxy_media is off to ensure the data is coming
> from you directly and not negotiated between providers.
>
>   On Tue, Aug 30, 2011 at 3:57 PM, Dan Lan <danlanweb at gmail.com> wrote:
>
>>   Hi,
>>
>> I want to use FS to accept call from SIP_TrunkA and terminate to
>> SIP_TrunkB
>> Both SIP trunks are using IP authentication, no need for username and
>> password.
>>
>> for incoming call (SIP_TrunkA), I have add the IP address of SIP_TrunkA in
>> to acl.conf.xml
>>  <node type="allow" cidr="xxx.xxx.xxx.xxx/32"/>
>>
>> I understand the incoming call will go to the public context, so I think I
>> need to do something here.
>>
>> I dont know what to do next.
>> 1. I try to establish a gateway for SIP_TrunkB for my outgoing call, but
>> sofia require me to have the username and password for the trunk. I dont
>> know where to add the SIP_TrunkB in freeswitch, since the provider of
>> SIP_TrunkB only need to recoginize my FS IP address.
>> 2. After I establish SIP_TrunkB, how should I do on public dialplan to
>> route the call from SIP_TrunkA to SIP_TrunkB? should I use "transfer" or
>> "bridge", could I make a dialplan that can route all the call from IP
>> address of A to IP address of B?
>>
>> Sorry for the newbie question, I try to look up on wiki but only got
>> partial information for me.
>>
>> Any help and any directions or hints are appreciated
>>
>> Dan Lan
>>
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110831/447b87ab/attachment.html 

From xing2kin at yahoo.com  Wed Aug 31 21:52:19 2011
From: xing2kin at yahoo.com (king2kin)
Date: Wed, 31 Aug 2011 10:52:19 -0700 (PDT)
Subject: [Freeswitch-users] session:recordFile(-) always creates empty wav
	file during outbound IVR call
Message-ID: <1314813139.64591.YahooMailClassic@web39701.mail.mud.yahoo.com>

Hi folks,

With Lua script and/or originate command, I have tried recording a message file during outbound IVR call over and over, session:recordFile(-) inside Lua script does create a wav file during each of my testings but the recorded audio file is always empty. 

However, session:recordFile(-) works well for inbound IVR call.

I tried the session:recordFile(-) via Lua script in three ways:

1. run lua script "test_outcall_ivr.lua" at freeswitch command-line:

luarun test_outcall_ivr.lua 


-- [test_outcall_ivr.lua]
{
local sessionx = freeswitch.Session("sofia/gateway/sip.tpad.com/1726011", session)

-- Set the path separator
pathsep = '/'

-- Windows users do this instead:
-- pathsep = '\'

-- Answer the call
-- sessionx:answer()

--Create a string with path and filename of a sound file
prompt = "ivr" .. pathsep .. "ivr-welcome_to_freeswitch.wav"

-- Print a log message
freeswitch.consoleLog("INFO","Prompt file is '" .. prompt .. "'\n")

--Play the prompt
sessionx:streamFile(prompt)

-- Record record file
sessionx:streamFile("phrase:voicemail_record_message")
			
-- Play a ""bong"" tone prior to recording
sessionx:streamFile("tone_stream://v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1000, 0, 640)")

-- record a message
filename = sessionx:getVariable('sounds_dir') .. pathsep .. "123.wav"
sessionx:recordFile(filename,300,100,10)

-- play back the recorded msg
sessionx:streamFile(filename)

-- Hangup
sessionx:hangup()

}

2. I also tried it differently by submitting the following commands at the FreeSwitch command-line interface:

originate user/1005 &transfer(8887 xml default)

originate user/1005 &lua('test1.lua')

originate sofia/gateway/sip.tpad.com/1726011 &lua('test1.lua')


-- [test1.lua]
{
-- Set the path separator
pathsep = '/'

-- Windows users do this instead:
-- pathsep = '\'

--Answer the call
session:answer()

--Create a string with path and filename of a sound file
prompt = "ivr" .. pathsep .. "ivr-welcome_to_freeswitch.wav"

-- Print a log message
freeswitch.consoleLog("INFO","Prompt file is '" .. prompt .. "'\n")

--Play the prompt
session:streamFile(prompt)

-- Record record file
session:streamFile("phrase:voicemail_record_message")
			
-- Play a ""bong"" tone prior to recording
session:streamFile("tone_stream://v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1000, 0, 640)")

-- record a message
filename = session:getVariable('sounds_dir') .. pathsep .. "123.wav"
session:recordFile(filename,300,100,10)

-- play back the recorded msg
session:streamFile(filename)

-- Hangup
session:hangup()
}

-- [xml dialplan for extension 8887]:
{
	<extension name="Simple Lua Test">
		<condition field="destination_number" expression="^(8887)$">
			<action application="set" data="record_waste_resources=true"/>
			<action application="lua" data="test1.lua"/>
		</condition>
	</extension>  
}

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

For all the above testing cases, session:recordFile(-) always creates an empty wav file for each of outbound IVR calls, however, if I make an inbound IVR?call to run Lua script "test1.lua", session:recordFile(-) always works perfect to generate a normal wav file.

So, what's wrong with [session:recordFile(-)] during an outbound IVR call?

x.k.


From sescher_ml at omeco.de  Wed Aug 31 21:55:04 2011
From: sescher_ml at omeco.de (Silvio Escher)
Date: Wed, 31 Aug 2011 19:55:04 +0200
Subject: [Freeswitch-users] Variables in Dialplan - Problem with getting
 Variables from User Directory
In-Reply-To: <A1E3DB7E-64B5-4E74-89FF-7F87E7F8D501@seventhsignal.de>
References: <4E5D50BD.8010208@omeco.de>
	<A1E3DB7E-64B5-4E74-89FF-7F87E7F8D501@seventhsignal.de>
Message-ID: <4E5E7578.2000803@omeco.de>

Am 31.08.11 15:20, schrieb Michal Bielicki:
>
> Am 30.08.2011 um 23:06 schrieb Silvio Escher:
>
>
> if you set the user with set_user there is no requirement to use user_data since it should get the 
> params for the respective user automatically.
>
yes - i already got this - i just want to show 2 ways to solve my issue partially ( maybe helpfull 
for others after google indexing ;) )
>> params are checked on call, so only there when required and checked each time, vars are fixed.
this wont help me much at this point - the question is "why" are the vars and params unavailable 
without set_user or user_data thingies ..
or at this special point - how to get the params available during the voicemail event

Best Regards,
Silvio




From mgg at giagnocavo.net  Wed Aug 31 22:13:18 2011
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Wed, 31 Aug 2011 14:13:18 -0400
Subject: [Freeswitch-users] problem using mod_managed under linux
In-Reply-To: <19062.1314810304@ccs.covici.com>
References: <30572.1314700796@ccs.covici.com>
	<CAKE0JBAEBLL=Mtp0PxMieZY5hAt2B56ZUKM-C+QwyxrV=RgC6Q@mail.gmail.com>
	<19062.1314810304@ccs.covici.com>
Message-ID: <03351FCC6082174C8534AB714B8258A5E43D8D3F@mse17be1.mse17.exchange.ms>

Right, the managed DLL should be identical. I don't believe there are any C# 4.0 features (are there any besides dynamic?) being used. It's more of a "gotta debug what's going on with Mono 2.8/2.10 with appdomains and see what's up". 

-Michael

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of covici at ccs.covici.com
Sent: Wednesday, August 31, 2011 11:05 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] problem using mod_managed under linux

I tried to remake the .dll using dmcs which is supposed to be for framework v4, but it did not help.

Tim McQueen <tim.mcqueen at gmail.com> wrote:

> I'm having the same problem.  (http://pastebin.freeswitch.org/17216)
> 
> Have you tried the Demo.csx to see if it will even run? In my case it 
> won't.  It is my opinon that it's because the FreeSwitch.Managed.dll 
> is dependent specifically on C# 4.0, since the same code will run 
> correctly on Windows.
> 
> I sent a message out to the freeswitch-devs group but I don't think it 
> was approved by the moderator, because I haven't seen it come back 
> through the list.  Someone on IRC said that mod_managed isn't 
> maintained anymore, but I see activity in FishEye from last week.
> 
> On Tue, Aug 30, 2011 at 5:39 AM, <covici at ccs.covici.com> wrote:
> 
> > Hi.  I am trying to use mod_managed on my linux box -- gentoo 
> > distribution -- and its driving me ...
> >
> >
> > I wrote a very simple program for a test.  The program just sets a 
> > dtmf callback and then streams a file and writes a log entry when it 
> > sees the dtmf.
> >
> > Under Windows 7 net framework 4, it works just fine.  Under Linux, 
> > using mono versions 2.8.2 qand 2.10.4, I get the following exception 
> > which I will put in a pastebin.
> >
> > http://pastebin.freeswitch.org/17232
> >
> > And the app is in this one
> >
> > http://pastebin.freeswitch.org/17233
> >
> > I have tried using gmcs and dmcs thinking it may be a .net framework 
> > issue, and if I do it wrong, I can bring down fs itself, but the 
> > best I can get under Linux is the exception shown above.
> >
> > Thanks in advance for anything you can come up with on this.
> >
> > --
> > Your life is like a penny.  You're going to lose it.  The question is:
> > How do
> > you spend it?
> >
> >         John Covici
> >         covici at ccs.covici.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-u
> > sers
> > http://www.freeswitch.org
> >
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org

--
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


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


From brad at tech21.com  Wed Aug 31 22:21:13 2011
From: brad at tech21.com (Brad Mina)
Date: Wed, 31 Aug 2011 11:21:13 -0700
Subject: [Freeswitch-users] Setup with Bandwidth.com
In-Reply-To: <0E3F1549A4FB4287A7F6B78A8EC2761F@ccs.local>
References: <0E3F1549A4FB4287A7F6B78A8EC2761F@ccs.local>
Message-ID: <CAGRr058n46Xmr6cx3bWvh9tkTQGOPZkwEz9-3EYYF6B0vqbZxg@mail.gmail.com>

Wayne, I've had a bit of decent fun with them.

First off call in and ask that they change the port to which they send data
to 5080 - this will fall in line with FS's external profile setup in a
vanilla environment.

Second, the username and password fields are optional, however they are
required in order for FS to accept the XML as valid. Also,
bandwidth.comdoes not have any usernames/passwords, but will send back
a 200OK from an
auth request if necessary.

Third, take note of the e.164 with leading '+', that screwed with me for a
while. While you're on the phone with them you might have them also change
the format to not use the leading + to make things a little easier.

On Tue, Aug 30, 2011 at 1:25 PM, Wayne <wayne at hamilton.net> wrote:

> Does anyone have a setup with bandwidth.com?
>
> The example on the wiki has a username and password. I didn't receive a
> username and password form bandwidth.com. Just setup a sip peer and have
> an
> IP address. If I send calls to freeswitch from the peer I get Rejected by
> acl "domains". Falling back to Digest auth. Looks like they are sending it
> on port 5060.
> I'm sure the rejected by acl is the most asked question but I have been
> unable to find the answer.
>
> Thanks Wayne
>
>
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110831/3b8449e7/attachment.html 

From covici at ccs.covici.com  Wed Aug 31 22:42:10 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Wed, 31 Aug 2011 14:42:10 -0400
Subject: [Freeswitch-users] problem using mod_managed under linux
In-Reply-To: <03351FCC6082174C8534AB714B8258A5E43D8D3F@mse17be1.mse17.exchange.ms>
References: <30572.1314700796@ccs.covici.com>
	<CAKE0JBAEBLL=Mtp0PxMieZY5hAt2B56ZUKM-C+QwyxrV=RgC6Q@mail.gmail.com>
	<19062.1314810304@ccs.covici.com>
	<03351FCC6082174C8534AB714B8258A5E43D8D3F@mse17be1.mse17.exchange.ms>
Message-ID: <32249.1314816130@ccs.covici.com>

I even tried this on DEbian squeeze which is using a mono 2.6 something
and it still does not work.

Michael Giagnocavo <mgg at giagnocavo.net> wrote:

> Right, the managed DLL should be identical. I don't believe there are any C# 4.0 features (are there any besides dynamic?) being used. It's more of a "gotta debug what's going on with Mono 2.8/2.10 with appdomains and see what's up". 
> 
> -Michael
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of covici at ccs.covici.com
> Sent: Wednesday, August 31, 2011 11:05 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] problem using mod_managed under linux
> 
> I tried to remake the .dll using dmcs which is supposed to be for framework v4, but it did not help.
> 
> Tim McQueen <tim.mcqueen at gmail.com> wrote:
> 
> > I'm having the same problem.  (http://pastebin.freeswitch.org/17216)
> > 
> > Have you tried the Demo.csx to see if it will even run? In my case it 
> > won't.  It is my opinon that it's because the FreeSwitch.Managed.dll 
> > is dependent specifically on C# 4.0, since the same code will run 
> > correctly on Windows.
> > 
> > I sent a message out to the freeswitch-devs group but I don't think it 
> > was approved by the moderator, because I haven't seen it come back 
> > through the list.  Someone on IRC said that mod_managed isn't 
> > maintained anymore, but I see activity in FishEye from last week.
> > 
> > On Tue, Aug 30, 2011 at 5:39 AM, <covici at ccs.covici.com> wrote:
> > 
> > > Hi.  I am trying to use mod_managed on my linux box -- gentoo 
> > > distribution -- and its driving me ...
> > >
> > >
> > > I wrote a very simple program for a test.  The program just sets a 
> > > dtmf callback and then streams a file and writes a log entry when it 
> > > sees the dtmf.
> > >
> > > Under Windows 7 net framework 4, it works just fine.  Under Linux, 
> > > using mono versions 2.8.2 qand 2.10.4, I get the following exception 
> > > which I will put in a pastebin.
> > >
> > > http://pastebin.freeswitch.org/17232
> > >
> > > And the app is in this one
> > >
> > > http://pastebin.freeswitch.org/17233
> > >
> > > I have tried using gmcs and dmcs thinking it may be a .net framework 
> > > issue, and if I do it wrong, I can bring down fs itself, but the 
> > > best I can get under Linux is the exception shown above.
> > >
> > > Thanks in advance for anything you can come up with on this.
> > >
> > > --
> > > Your life is like a penny.  You're going to lose it.  The question is:
> > > How do
> > > you spend it?
> > >
> > >         John Covici
> > >         covici at ccs.covici.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-u
> > > sers
> > > http://www.freeswitch.org
> > >
> > 
> > ----------------------------------------------------
> > Alternatives:
> > 
> > ----------------------------------------------------
> > 
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> > rs
> > http://www.freeswitch.org
> 
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
> 
>          John Covici
>          covici at ccs.covici.com
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From rhuddleston at gmail.com  Wed Aug 31 22:49:09 2011
From: rhuddleston at gmail.com (Robert Huddleston)
Date: Wed, 31 Aug 2011 14:49:09 -0400
Subject: [Freeswitch-users] Setup with Bandwidth.com
In-Reply-To: <CAGRr058n46Xmr6cx3bWvh9tkTQGOPZkwEz9-3EYYF6B0vqbZxg@mail.gmail.com>
References: <0E3F1549A4FB4287A7F6B78A8EC2761F@ccs.local>
	<CAGRr058n46Xmr6cx3bWvh9tkTQGOPZkwEz9-3EYYF6B0vqbZxg@mail.gmail.com>
Message-ID: <025001cc680e$ac215c00$04641400$@com>

Ya the e.164 format was a joke.. And for a while I had a problem with caller ID because I was using an gateway and they *WOULD NOT* send caller id.. Demanded I send it?

 

Found that setting the username ?not one they provided? to the caller ID I wanted on the gateway ? I could get it to send out caller id.

 

For a flat rate trunk I think their pricing is good.. But you can get a lot of NO answers from them on tech support issues.

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brad Mina
Sent: Wednesday, August 31, 2011 2:21 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Setup with Bandwidth.com

 

Wayne, I've had a bit of decent fun with them.

First off call in and ask that they change the port to which they send data to 5080 - this will fall in line with FS's external profile setup in a vanilla environment.


Second, the username and password fields are optional, however they are required in order for FS to accept the XML as valid. Also, bandwidth.com does not have any usernames/passwords, but will send back a 200OK from an auth request if necessary.

Third, take note of the e.164 with leading '+', that screwed with me for a while. While you're on the phone with them you might have them also change the format to not use the leading + to make things a little easier.

On Tue, Aug 30, 2011 at 1:25 PM, Wayne <wayne at hamilton.net> wrote:

Does anyone have a setup with bandwidth.com?

The example on the wiki has a username and password. I didn't receive a
username and password form bandwidth.com. Just setup a sip peer and have an
IP address. If I send calls to freeswitch from the peer I get Rejected by
acl "domains". Falling back to Digest auth. Looks like they are sending it
on port 5060.
I'm sure the rejected by acl is the most asked question but I have been
unable to find the answer.

Thanks Wayne



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

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110831/380d6aee/attachment.html 

From anthony.minessale at gmail.com  Wed Aug 31 23:17:59 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 31 Aug 2011 14:17:59 -0500
Subject: [Freeswitch-users] session:recordFile(-) always creates empty
 wav file during outbound IVR call
In-Reply-To: <1314813139.64591.YahooMailClassic@web39701.mail.mud.yahoo.com>
References: <1314813139.64591.YahooMailClassic@web39701.mail.mud.yahoo.com>
Message-ID: <CAKbxfG89m2cDYkWCqJrHAwse_2Ei26d7CBbOTcuykc4h9Eqw8g@mail.gmail.com>

try this dial string instead

{ignore_early_media=true}sofia/gateway/sip.tpad.com/1726011

On Wed, Aug 31, 2011 at 12:52 PM, king2kin <xing2kin at yahoo.com> wrote:
> Hi folks,
>
> With Lua script and/or originate command, I have tried recording a message file during outbound IVR call over and over, session:recordFile(-) inside Lua script does create a wav file during each of my testings but the recorded audio file is always empty.
>
> However, session:recordFile(-) works well for inbound IVR call.
>
> I tried the session:recordFile(-) via Lua script in three ways:
>
> 1. run lua script "test_outcall_ivr.lua" at freeswitch command-line:
>
> luarun test_outcall_ivr.lua
>
>
> -- [test_outcall_ivr.lua]
> {
> local sessionx = freeswitch.Session("sofia/gateway/sip.tpad.com/1726011", session)
>
> -- Set the path separator
> pathsep = '/'
>
> -- Windows users do this instead:
> -- pathsep = '\'
>
> -- Answer the call
> -- sessionx:answer()
>
> --Create a string with path and filename of a sound file
> prompt = "ivr" .. pathsep .. "ivr-welcome_to_freeswitch.wav"
>
> -- Print a log message
> freeswitch.consoleLog("INFO","Prompt file is '" .. prompt .. "'\n")
>
> --Play the prompt
> sessionx:streamFile(prompt)
>
> -- Record record file
> sessionx:streamFile("phrase:voicemail_record_message")
>
> -- Play a ""bong"" tone prior to recording
> sessionx:streamFile("tone_stream://v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1000, 0, 640)")
>
> -- record a message
> filename = sessionx:getVariable('sounds_dir') .. pathsep .. "123.wav"
> sessionx:recordFile(filename,300,100,10)
>
> -- play back the recorded msg
> sessionx:streamFile(filename)
>
> -- Hangup
> sessionx:hangup()
>
> }
>
> 2. I also tried it differently by submitting the following commands at the FreeSwitch command-line interface:
>
> originate user/1005 &transfer(8887 xml default)
>
> originate user/1005 &lua('test1.lua')
>
> originate sofia/gateway/sip.tpad.com/1726011 &lua('test1.lua')
>
>
> -- [test1.lua]
> {
> -- Set the path separator
> pathsep = '/'
>
> -- Windows users do this instead:
> -- pathsep = '\'
>
> --Answer the call
> session:answer()
>
> --Create a string with path and filename of a sound file
> prompt = "ivr" .. pathsep .. "ivr-welcome_to_freeswitch.wav"
>
> -- Print a log message
> freeswitch.consoleLog("INFO","Prompt file is '" .. prompt .. "'\n")
>
> --Play the prompt
> session:streamFile(prompt)
>
> -- Record record file
> session:streamFile("phrase:voicemail_record_message")
>
> -- Play a ""bong"" tone prior to recording
> session:streamFile("tone_stream://v=-7;%(100,0,941.0,1477.0);v=-7;>=2;+=.1;%(1000, 0, 640)")
>
> -- record a message
> filename = session:getVariable('sounds_dir') .. pathsep .. "123.wav"
> session:recordFile(filename,300,100,10)
>
> -- play back the recorded msg
> session:streamFile(filename)
>
> -- Hangup
> session:hangup()
> }
>
> -- [xml dialplan for extension 8887]:
> {
> ? ? ? ?<extension name="Simple Lua Test">
> ? ? ? ? ? ? ? ?<condition field="destination_number" expression="^(8887)$">
> ? ? ? ? ? ? ? ? ? ? ? ?<action application="set" data="record_waste_resources=true"/>
> ? ? ? ? ? ? ? ? ? ? ? ?<action application="lua" data="test1.lua"/>
> ? ? ? ? ? ? ? ?</condition>
> ? ? ? ?</extension>
> }
>
> ======================================================
>
> For all the above testing cases, session:recordFile(-) always creates an empty wav file for each of outbound IVR calls, however, if I make an inbound IVR?call to run Lua script "test1.lua", session:recordFile(-) always works perfect to generate a normal wav file.
>
> So, what's wrong with [session:recordFile(-)] during an outbound IVR call?
>
> x.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
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From covici at ccs.covici.com  Wed Aug 31 23:38:51 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Wed, 31 Aug 2011 15:38:51 -0400
Subject: [Freeswitch-users] problem using mod_managed under linux
In-Reply-To: <03351FCC6082174C8534AB714B8258A5E43D8C24@mse17be1.mse17.exchange.ms>
References: <30572.1314700796@ccs.covici.com>
	<CAKE0JBAEBLL=Mtp0PxMieZY5hAt2B56ZUKM-C+QwyxrV=RgC6Q@mail.gmail.com>
	<03351FCC6082174C8534AB714B8258A5E43D8C24@mse17be1.mse17.exchange.ms>
Message-ID: <7399.1314819531@ccs.covici.com>

Any chance of you fixing the bug?  I think you must be right because in
the server stack trace I see a lot of things involving serialization.

Michael Giagnocavo <mgg at giagnocavo.net> wrote:

> It's probably more related to some cross-appdomain/serialization stuff that's specific to Mono. I wrote mod_managed against Mono 2.4 or so, and I think last time I ran it was 2.6, and only on CentOS 5. Probably something changed and is triggering a bug in mod_managed in the newer builds.
> 
> 
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tim McQueen
> Sent: Wednesday, August 31, 2011 9:08 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] problem using mod_managed under linux
> 
> I'm having the same problem.  (http://pastebin.freeswitch.org/17216)
> 
> Have you tried the Demo.csx to see if it will even run? In my case it won't.  It is my opinon that it's because the FreeSwitch.Managed.dll is dependent specifically on C# 4.0, since the same code will run correctly on Windows.
> 
> I sent a message out to the freeswitch-devs group but I don't think it was approved by the moderator, because I haven't seen it come back through the list.  Someone on IRC said that mod_managed isn't maintained anymore, but I see activity in FishEye from last week.
> On Tue, Aug 30, 2011 at 5:39 AM, <covici at ccs.covici.com<mailto:covici at ccs.covici.com>> wrote:
> Hi.  I am trying to use mod_managed on my linux box -- gentoo
> distribution -- and its driving me ...
> 
> 
> I wrote a very simple program for a test.  The program just sets a dtmf
> callback and then streams a file and writes a log entry when it sees the
> dtmf.
> 
> Under Windows 7 net framework 4, it works just fine.  Under Linux, using
> mono versions 2.8.2 qand 2.10.4, I get the following exception which I
> will put in a pastebin.
> 
> http://pastebin.freeswitch.org/17232
> 
> And the app is in this one
> 
> http://pastebin.freeswitch.org/17233
> 
> I have tried using gmcs and dmcs thinking it may be a .net framework
> issue, and if I do it wrong, I can bring down fs itself, but the best I
> can get under Linux is the exception shown above.
> 
> Thanks in advance for anything you can come up with on this.
> 
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
> 
>         John Covici
>         covici at ccs.covici.com<mailto:covici at ccs.covici.com>
> 
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com