May 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

1.6 install to Centos
by Denis Jakovlev 26 May '15

26 May '15

Hi all, Is there a chance to put on CentOS the new version 1.6? I've tried. But I have a problem on ./configure -C checking for libyuv >= 0.0.1280... configure: error: You need to install libyuv-dev. Required library Where can I find this library? I tried to put it from here https://code.google.com/p/libyuv/ But FreeSwitch does not see it. Is there a possibility to find this library to CentOS? -- S pozdravem, Ing.Denis Jakovlev mob.tel. 775-415-382

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XML_RPC connections limit
by Leonid Nasedkin 26 May '15

26 May '15

Hi, All! Im trying to use freeswitch for automated outbound calls. Im init calls via xml_rpc interface with api call. But it seems like somewhere in freeswitch hardcoded connection limit to xml_rpc . I cant create more then 16 concurrent connections. How I can increase this value? Im using: FreeSWITCH version: 1.4.15~64bit ( 64bit) CentOS 6.5 2.6.32-504.16.2.el6.x86_64 Thanks.

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FreeSWITCH Week in Review (Master Branch) May 16th-May 22nd
by Ken Rice 25 May '15

25 May '15

New Post on freeswitch.org from Kathleen check it out at http://ift.tt/1FbFRZL FreeSWITCH Week in Review (Master Branch) May 16th-May 22nd Hello, again. This passed week in the FreeSWITCH master branch we had 7 commits. We saw some neat bug fixes go in this week and I hope everyone partaking in today’s holiday enjoys their long weekend! Join us on Wednesdays at 12:00 CT for some more FreeSWITCH fun! And head over to freeswitch.com to learn more about FreeSWITCH support. Improvements in build system, cross platform support, and packaging: FS-7488 [mod_managed] Fix a build error with Windows and removed duplicate files The following bugs were squashed: FS-7541 Fixed an issue with audio gaps in native audio recordings FS-7562 [mod_sofia] Fixed an interop issue caused when using bypass media with t.38 passthru FS-7567 Fixed a rare segfault on shutdown caused by a race condition FS-7529 Fixed an error with call recording on G722 calls

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CELL number providers in latam
by Alejandro 25 May '15

25 May '15

Hi all, I'm searching for advices, about some company that provide Cell Numbers locally in LATAM. The idea is the customer can send a SMS to this number and we can receive the message. Thanks for any advice. Ale

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ringback - multiple values
by Dmitry Mordovin 25 May '15

25 May '15

Hello Is it possible to set multiple values in ringback? <action application="set" data="ringback=file_string:///opt/freeswitch/sounds/en/us/callie/voicemail/8000/vm-goodbye.wav,${us-ring}"/> I try to use " , ! ; " character for separate - Not works. My idea is play wav file and then play us-ring while bridge is calling to destination How can do it?

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no ringing then calling local user
by Dmitry Mordovin 25 May '15

25 May '15

Hello My extension <extension name="test10"> <condition field="destination_number" expression="^2222$"> <action application="answer"/> <action application="sleep" data="1500"/> <action application="set" data="hangup_after_bridge=true" /> <action application="bridge" data="sofia/internal/1004%${domain_name}"/> <action application="hangup"/> </condition> </extension> From fs_cli I originating call originate {ignore_early_media=true}sofia/gateway/sipnet/12126248024\@sipnet 1004 Soft phone 1004 registered and ringing then incoming call, but 12126248024 (aleg) is silence! In debug looks good 2015-05-25 17:46:31.990988 [DEBUG] sofia.c:6627 Channel sofia/internal/1004 entering state [proceeding][180] 2015-05-25 17:46:31.990988 [NOTICE] sofia.c:6729 Ring-Ready sofia/internal/1004! 2015-05-25 17:46:31.990988 [DEBUG] switch_channel.c:3297 (sofia/internal/1004) Callstate Change DOWN -> RINGING May I forget something?

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Difficult extension case
by Dmitry Mordovin 24 May '15

24 May '15

Hello All! I wish to implement one extension scenario, could help me? - Play prompt (its easy) - Play ringback music - Bridge to one or multiple numbers - On answer: - Play BEEP to incoming leg (stop music, play BEEP means call success, starting talks) - Play "IMPORTANT CALL!!!" to new outgoing leg - Connect incoming and outgoing legs together - On fail (timeout): - Play "Please recall later" I don't understand how can play on answer separately each leg Thank you

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RTCP and WebRTC
by Jon Lederman 24 May '15

24 May '15

Hi, I am trying to use WebRTC with Freeswitch. WebRTC relies upon RTCP for bandwidth estimation. I have tried to get forwarding of RTCP packets working with Freeswitch to make this work. However, a more fundamental issue seems to be occurring in that Freeswitch is not able to process the RTCP messages arriving from the WebRTC client. I have traced the problem to the function process_rtcp_frame in switch_rtp.c. In that function, the version of RTP appearing in the RTCP packets is 0, not 2, which is required and the packets are never processed (i.e., no fresh frames). I am not sure what the cause of this is? Could it be related to muxing of RTP/RTCP, which webrtc utilizes? Any thoughts on resolving this would be greatly appreciated. Thanks. -Jon

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XML CDR help
by Mitchelle Johnson 23 May '15

23 May '15

Hi, When processing XML CDRs sometimes(not all) the callee_id_number tag is left blank. Please tell me why does this happen? Thanks, Mitchelle

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Google Hangouts Integration
by Mitchelle Johnson 23 May '15

23 May '15

Hi all, Like skype can be integrated with FreeSWITCH. Is there a way to integrate google Hangouts also? Thanks, Mitchelle

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