November 2014 - freeswitch-dev@lists.freeswitch.org

[Freeswitch-dev] 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-dev] 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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[Freeswitch-dev] 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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