- FreeSWITCH-dev - FreeSWITCH Mailing Lists

FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 07 Nov '14

07 Nov '14

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

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mod_conference remote recording
by miguelo 05 Nov '14

05 Nov '14

Hello, I try to setup recording with mod_conference and mod_oreka. I know it is possible to record to file with mod_conference, but i want to record a conference using mod_oreka. I tried dialing out to the loopback interface and answering that call in the dialplan to start recording but the recording will stop when the end of the extension is reached. I also tried to change the c code to start another session and execute the application oreka_record on that session, but i couldn't figure out how to correctly setup media on that session. Does anyone know s any module i could use as reference or another method to acomplish this. Thanks, Miguelo

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Freeswitch "crash" on switch_core_media_toggle_hold
by miguelo sana 03 Nov '14

03 Nov '14

Hello, Today i experienced a very weird crash on my production freeswitch server. I use freeswitch for a conferencing service and at the moment of the crash there where around 40 participants in 15 different conferences. In one of the conferences a participant put his call on hold (probably with his mobile phone). At the moment when freeswitch received the message to put the call on unhold the b-leg channel was gone and the freeswitch crashed. I have created a backtrace on the crash and put it on the pastebin https://pastebin.freeswitch.org/23467 I tried to replicate the crash by dialing in with multiple phones and put the call on hold and disconnect with another phone. On all my attempts the conference application recognized the disconnect and did not crash. Not even when i force quit a sip client. Has anyone ever experienced this. Or is anyone able to explain why this happens. I use freeswitch version 1.4.12 on debian 7. The machine already succesfully handled different conferences with a total of 80 participants. So performance does not seem to be the issue. Second i would like to know if it is possible to "disable" the switch_assert in production builds of freeswitch and have a more elegant way of handling problems in threads. Because when looking to my problem. Freeswitch aborted because one channel was not available when it was supposed to be. I understand that thread has to fail because it has no channel but it should not affect the whole freeswitch application. If anyone has some ideas to get this done or a solution please let me know. Thanks, Miguelo Sana

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Software/Hardware interoperability
by Mario Gasparoni Jr 03 Nov '14

03 Nov '14

Hello guys, I am trying to use FreeSWITCH as a conference server for group calls. I'm using mod_conference, and I know it works fine with multiple SIP clients (for testing purposes I use Ekiga and Jitsi on both Windows and Linux). Audio is mixed individually for each user, and video is switched based on the active talker, and the same video is sent to all clients. The problem shows up when I join the group call using a Polycom hardware. If I configure FreeSWITCH to support H.263 only, it works fine, my Polycom can display the video sent by FreeSWITCH, including when the active talker is itself. The other SIP clients work fine as well. When I configure FreeSWITCH to support H.264 only, the video switching only works when Polycom is the active talker, otherwise all the users just see the last video frame sent by Polycom. If I do a peer-to-peer call between the Polycom and a SIP client, both configured to use H.264, everything works just fine, which points me that the problem could be related to the transition between viewing a video stream with a given resolution and viewing another video stream with a different resolution, or something related to the non-continuity of video packets timestamp. Is there anybody out there that have experienced similar issues? Do you guys have any suggestion or anything that could help me to make it work properly? Thanks in advance. -- Att. Mário Gasparoni. www.inf.ufrgs.br/~mcgjunior

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FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 31 Oct '14

31 Oct '14

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit https://conference.freeswitch.org and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see https://confluence.freeswitch.org/display/FREESWITCH/Community for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

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FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 17 Oct '14

17 Oct '14

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

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Phone Feature Synchronization
by Kevin Wormington 16 Oct '14

16 Oct '14

Hi, I'm trying out the "feature synchronization" functionality of the new .73 beta firmware for the T46G that allows the phone to notify FreeSWITCH of DND and Forwarding status. The phone is using SUBSCRIBE messages to do this (example below) and FS processes them in sofia_presence.c and generates a SWITCH_EVENT_PHONE_FEATURE_SUBSCRIBE event. I don't see that this event is handled anywhere in mod_sofia. I would like to add a configurable feature at let FS emulate key system behavior and in the case of DND update the presence to be DND/Busy so the BLF lights on other UA would indicate busy. I know this is not necessarily what BLF means but most old key systems have this behaviour. I have not made an extensive review of mod_sofia but I would think this would be implemented there where the csta stuff is. I can play around with it, but wanted to see if anyone else had an opinion on it or if I'm completely missing some other way to do it other than using ESL to catch events and send new ones. Here is an example of DND being activated: ------------------------------------------------------------------------ SUBSCRIBE sip:1000@192.168.1.175:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:5062;branch=z9hG4bK2975756387 From: <sip:1000@192.168.1.175>;tag=2420321046 To: <sip:1000@192.168.1.175>;tag=YKBD9deNLnnN Call-ID: 3592119465(a)192.168.1.106 CSeq: 5 SUBSCRIBE Contact: <sip:1000@192.168.1.106:5062> Authorization: Digest username="1000", realm="192.168.1.175", nonce="71a73470-5561-11e4-9781-81e6a6ef6342", uri="sip:1000@192.168.1.175:5060", response="78e97ccca503de4e7490b0bbbf2f35ef", algorithm=MD5, cnonce="0a4f113b", qop=auth, nc=00000004 Content-Type: application/x-as-feature-event+xml Max-Forwards: 70 User-Agent: Yealink SIP-T46G 28.73.0.16 Expires: 3630 Event: as-feature-event Content-Length: 185 <?xml version="1.0" encoding="ISO-8859-1"?><SetDoNotDisturb xmlns="http://www.ecma-international.org/standards/ecma-323/csta/ed3"><doNotDisturbOn>true</doNotDisturbOn></SetDoNotDisturb> And turning off: ------------------------------------------------------------------------ SUBSCRIBE sip:1000@192.168.1.175:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:5062;branch=z9hG4bK264963740 From: <sip:1000@192.168.1.175>;tag=2420321046 To: <sip:1000@192.168.1.175>;tag=YKBD9deNLnnN Call-ID: 3592119465(a)192.168.1.106 CSeq: 6 SUBSCRIBE Contact: <sip:1000@192.168.1.106:5062> Authorization: Digest username="1000", realm="192.168.1.175", nonce="71a73470-5561-11e4-9781-81e6a6ef6342", uri="sip:1000@192.168.1.175:5060", response="4042dfcf7590c1d5187aa4b260275991", algorithm=MD5, cnonce="0a4f113b", qop=auth, nc=00000005 Content-Type: application/x-as-feature-event+xml Max-Forwards: 70 User-Agent: Yealink SIP-T46G 28.73.0.16 Expires: 3630 Event: as-feature-event Content-Length: 186 <?xml version="1.0" encoding="ISO-8859-1"?><SetDoNotDisturb xmlns="http://www.ecma-international.org/standards/ecma-323/csta/ed3"><doNotDisturbOn>false</doNotDisturbOn></SetDoNotDisturb>

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Latest 1.4 still fails at FS-6849. Does anyone else notice it?
by Nuno Reis 13 Oct '14

13 Oct '14

Hi guys. I've just upgrade to FS 1.4.12 (b33a86c) and I see that JIRA FS-6849 <https://jira.freeswitch.org/browse/FS-6849> still happens. Isn't there anyone able to fix the issue using only the feedback provided by me on that JIRA? Do you really thing git bisect is the only way to go here? I really hope not. Looking forward to here from you. Best Regards, -- *Nuno Miguel Reis* | *Unified Communication** Systems* M. +351 913907481 | nreis(a)wavecom.pt WAVECOM-Soluções Rádio, S.A. Cacia Park | Rua do Progresso, Lote 15 3800-639 AVEIRO | Portugal T. +351 309 700 225 | F. +351 234 919 191 *GPS <http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a…> | www.wavecom.pt <http://www.wavecom.pt/>** <http://www.wavecom.pt/>* [image: Description: Description: WavecomSignature] <http://www.wavecom.pt/pt/wavecom/premios.php> [image: Publicity] <http://www.wavecom.pt/pt/mail_eventos.php> On Thu, Sep 25, 2014 at 4:54 AM, Nuno Reis <nreis(a)wavecom.pt> wrote: > Using gdb I've figured out that in mod_python.c:281: > ... > result = PyEval_CallObjectWithKeywords(function, arg, (PyObject *) NULL); > Py_DECREF(function); > ... > > variable result is always null when a python script trying to use the > session run the first time right after freeswitch startup. > That will make it enter the else logic in line 291 in the same file ending > with a crash. > I'll still have to drill more to find out why this is happening now. > Anyone has a clue on what's going on? > > BR, > > -- > > *Nuno Miguel Reis* | *Unified Communication** Systems* > M. +351 913907481 | nreis(a)wavecom.pt > WAVECOM-Soluções Rádio, S.A. > Cacia Park | Rua do Progresso, Lote 15 > 3800-639 AVEIRO | Portugal > T. +351 309 700 225 | F. +351 234 919 191 > *GPS > <http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a…> > | www.wavecom.pt <http://www.wavecom.pt/>** <http://www.wavecom.pt/>* > > [image: Description: Description: WavecomSignature] > <http://www.wavecom.pt/pt/wavecom/premios.php> > > [image: Publicity] <http://www.wavecom.pt/pt/mail_eventos.php> > > > > On Tue, Sep 23, 2014 at 3:53 AM, Nuno Reis <nreis(a)wavecom.pt> wrote: > >> This is the output after calling the script right after start: >> >> 2014-09-23 03:49:45.801129 [NOTICE] mod_python.c:212 Invoking py module: >> teste >> 2014-09-23 03:49:45.801129 [NOTICE] switch_cpp.cpp:1328 Testing123... >> 2014-09-23 03:49:45.801129 [NOTICE] switch_cpp.cpp:1328 Dumping current >> CDR INFO >> 2014-09-23 03:49:45.801129 [ERR] mod_python.c:293 Error calling python >> script >> 2014-09-23 03:49:45.801129 [ERR] mod_python.c:164 Python Error by calling >> script "teste": <type 'exceptions.AttributeError'> >> Message: 'PySwigObject' object has no attribute 'getXMLCDR' >> Exception: None >> >> Traceback (most recent call last) >> File: "/opt/freeswitch/scripts/teste.py", line 12, in handler >> >> All the following script calls work just fine. >> >> *Note:* If you want to test the code snippet i've added before don't >> forget to add: >> >> import freeswitch >> >> Looking forward to hear from you. >> >> Best Regards, >> >> -- >> >> *Nuno Miguel Reis* | *Unified Communication** Systems* >> M. +351 913907481 | nreis(a)wavecom.pt >> WAVECOM-Soluções Rádio, S.A. >> Cacia Park | Rua do Progresso, Lote 15 >> 3800-639 AVEIRO | Portugal >> T. +351 309 700 225 | F. +351 234 919 191 >> *GPS >> <http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a…> >> | www.wavecom.pt <http://www.wavecom.pt/>** <http://www.wavecom.pt/>* >> >> [image: Description: Description: WavecomSignature] >> <http://www.wavecom.pt/pt/wavecom/premios.php> >> >> [image: Publicity] <http://www.wavecom.pt/pt/mail_eventos.php> >> >> >> >> On Tue, Sep 23, 2014 at 3:44 AM, Nuno Reis <nreis(a)wavecom.pt> wrote: >> >>> Hi guys. >>> >>> I've reported this before and i can tell that it is still happening in >>> the latest v1.4. >>> >>> How reproduce: >>> >>> >>> 1. just create a dialplan that calls a python script >>> 2. dial that dialplan entry >>> >>> >>> Example code (works with anything that invokes *session*): >>> >>> #!/usr/bin/env python >>> # -*- coding: utf-8 -*- >>> >>> from __future__ import absolute_import >>> # http://docs.python.org/whatsnew/pep-328.html >>> >>> def handler(session, args): >>> freeswitch.consoleLog("notice", "Testing123...") >>> freeswitch.consoleLog("notice", "Dumping current CDR INFO") >>> freeswitch.consoleLog("info", str(session.getXMLCDR())) >>> freeswitch.consoleLog("notice", "The END!") >>> >>> >>> mod_pyhton crashes everytime the script is called right after a >>> freeswitch start (the very first time session is invoked inside a >>> mod_python python script). >>> After the first time (error), everything starts to work just fine. >>> This doesn't happen in e6b8b3e4f2deac73547bc9b9a2cb3e1d3147d101 from >>> v1.2.stable >>> >>> Should i open a jira on this? >>> >>> -- >>> >>> *Nuno Miguel Reis* | *Unified Communication** Systems* >>> M. +351 913907481 | nreis(a)wavecom.pt >>> WAVECOM-Soluções Rádio, S.A. >>> Cacia Park | Rua do Progresso, Lote 15 >>> 3800-639 AVEIRO | Portugal >>> T. +351 309 700 225 | F. +351 234 919 191 >>> *GPS >>> <http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a…> >>> | www.wavecom.pt <http://www.wavecom.pt/>** <http://www.wavecom.pt/>* >>> >>> [image: Description: Description: WavecomSignature] >>> <http://www.wavecom.pt/pt/wavecom/premios.php> >>> >>> [image: Publicity] <http://www.wavecom.pt/pt/mail_eventos.php> >>> >>> >>> >> >

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[Announcement] FreeSWITCH 1.4.12 Released!!
by Ken Rice 10 Oct '14

10 Oct '14

Hi FreeSWITCHers! The FreeSWITCH Team would like to announce FreeSWITCH 1.4.12! This is a bug fix release and addresses a possible DoS. Thanks to Travis Cross for finding and patching this issue. commit b5294c53d694499fe0f4a807554154ca8c14dfa6 Author: Travis Cross <tc(a)traviscross.com> Date: Fri Oct 10 18:25:11 2014 +0000 Fix crash on transport=tls with non-TLS profile We use the transport of the Contact header of the remote UAC to decide which of our own Contact addresses we should use when replying to a SUBSCRIBE or sending a presence NOTIFY. If TLS is not enabled on a Sofia profile, then the TLS Contacts for that profile are NULL. Unfortunately we were using these NULL values uncritically when the remote UAC sent us a Contact header with a TLS transport and our own Sofia profile did not have TLS enabled. With this commit we fall back to our TCP Contact address when the remote Contact is TLS and our Sofia profile does not have TLS enabled. There are a few other minor bug fixes also to address a media resume after hold problem that is limited in scope and fixes to fs_encode for G722 file encoding. Source tarballs are available via http://files.freeswitch.org/ and packages are available via the FreeSWITCH Yum and Debian repositories. Have a Great Weekend! K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 10 Oct '14

10 Oct '14

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

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