May 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Multi-user video conference
by happy.neko 29 May '16

29 May '16

Hello, I have following requirements for multi-user video conference: 1. One Teacher and up to 10 Students are in the same room. 2. Teacher hears everyone in the room and sees video streams from every student (total bandwidth = 1 audios stream + n video streams). 3. Student hears everyone in the room and sees video stream from Teacher only (total bandwidth = 1 audio stream + 1 video stream). 4. There is no CPU intensive video encoding process on the server. Teacher receives video streams as-is plus muxed audio stream. 5. Conference is auto-recorded on the server. (I don't know exactly how, mp4 with multiple video streams?) Is it possible to implement this with FreeSwitch? If no what requirements should be relaxed? Also what exactly is the difference between "mux", "transcode" and "passthrough" video modes? -- Best regards, I. Sokolov

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FreeSWITCH using outbound proxy but to bypass media
by Muhammad Naseer Bhatti 28 May '16

28 May '16

I am trying to use an outbound SIP proxy for FreeSWITCH by adding fs_path to the bridge. It’s working fine and I am able to send calls using the proxy. On the proxy (which is also FreeSWITCH) I am receiving the call in a lua script, create a new session with the destination gateway and bridge two sessions (old, incoming and the newly created) The idea is to have multiple FreeSWITCH servers (all public IP address) originating calls while all calls going to the carrier via the proxy. The problem is that all media is also passing through the proxy. Is there a way I can only send signaling through proxy and media to flow directly between the endpoints? I have tried bypass_media and bypass_media_after_bridge but it’s not working. Is this even doable? Thanks. -- Muhammad Naseer Bhatti

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How FreeSwitch starts on Debian 8
by Valter Nogueira 27 May '16

27 May '16

I followed the instructions of https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie Which is pretty simple, indeed (exept by -y on apt-get install): apt-get update && apt-get install -y curl curl https://files.freeswitch.org/repo/deb/debian/freeswitch_archive_g0.pub | apt-key add - echo "deb http://files.freeswitch.org/repo/deb/freeswitch-1.6/ jessie main" > /etc/apt/sources.list.d/freeswitch.list apt-get update && apt-get install freeswitch-meta-all After installation, Freeswitch was up and running, but I don't see any /etc/init.d script to start it. When I killed the process, it automatically restarted! How does it happen? Thank you Valter

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TTS module using SAPI?
by Alex Pierry 27 May '16

27 May '16

Hi All, what about a Freeswitch module that integrates with a TTS engine through SAPI enviroment? Do you know if exists some initiative in this way? It would be great to use Microsoft's TTS engine that has a good quality and a large number of languages available. Thanks, Alex. <http://pt.slideshare.net/marketingteclan> <http://blog.teclan.com.br/>

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[Default Directory XML] All internal extensions looking at only 1 extensions xml file while dialling numbers internally
by Daniel Nazareth 26 May '16

26 May '16

Hi , My problem is that each of my extensions(ranging from 3001-3015) looks up only extension 3001.xml in /etc/freeswitch/directory/default while dialling internal extensions. This means that although calls are routed perfectly, even for a call not involving 3001 only the caller id, caller_id_name etc for 3001 is seen on the IP phone. All the other extension xml's have the exact same pattern but their details are ignored, can anyone explain this behaviour? I've pored over the wiki for directory XML but cannot find any clues. Sample XML's are at *http://pastebin.com/MkYFChh9 <http://pastebin.com/MkYFChh9>* Thanks Daniel

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Time is running out for ClueCon Speaker Proposals
by Anthony Minessale 26 May '16

26 May '16

Get them in soon! -- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA

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mod_verto webphone
by magnusadilsom＠gmail.com 26 May '16

26 May '16

I’m developer a CallCenter manager system with FS and I need add in the panel a web phone. I'm try do it using mod_verto I already install FS, and load mod_verto install verto in my http server https://www.my_web_site.com/verto/#page-main <https://www.my_web_site.com/verto/#page-main> but in FS console I see error [INFO] mod_verto.c:4025 181.95.222.30:51652 Client Connect. [INFO] mod_verto.c:1978 181.95.222.30:51652 Starting client thread. [DEBUG] mod_verto.c:1825 181.95.222.30:51652 WS SETUP FAILED [] [INFO] mod_verto.c:2005 181.95.222.30:51652 Ending client thread. Thanks

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verto.js and Phonegap/Cordova iOS
by Oivvio Polite 26 May '16

26 May '16

Is anyone here using verto.js in conjunction with Phonegap/Cordova on iOS? regards Oivvio Polite

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bypass_media + SDP change workaround
by Sergey Safarov 26 May '16

26 May '16

FS is not propagates SDP change to aleg when used bypass_media(FS-9203 <https://freeswitch.org/jira/browse/FS-9203>). Are workaround is known?

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C API documentation
by Luis Daniel Lucio Quiroz 25 May '16

25 May '16

Hello, Where I can read how to start coding a module. I know c/c++, but I ignore the FreeSWITCH API

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