November 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_av passthrough
by Madovsky 14 Nov '17

14 Nov '17

Hi folks, is it possible to pass through with mod_av? ex leg a and b has already H264, avoid to re encode again thanks

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Options for initiating call from browser (not WebRTC)
by Oivvio Polite 14 Nov '17

14 Nov '17

I have a simple call recording solution set up with a Twilio-like PaaS. The user interacts with a web app where she enters a destination number. My web app then instructs the PaaS to 1. Call the users phone 2. Wait for the user to answer 3. Start a recording. 4. Connect the destination number. The resulting wav is available for the user to download in the web app. I now want to replace the PaaS with FreeSwitch. I've done a lot of experimenting with FS during the last year, and managed to get callrecording of PSTN and WebRTC calls to work. With those calls origination was from PSTN or WebRTC. Now I want to do origination with a HTTP request. When leafing through the FreeSwitch book from PACKT I see that I have multiple options for doing this. 1. fs_cli Have the browser talk to a separate web app (written in node or python or whatever) that the initiates the call with fs_cli. 2. mod_httapi Have the browser talk to a webapp that talks to FS via mod_httapi. In this scenario I can't find any info on how to originate calls in mod_httapi though. 3. mod_event_socket Have a the browser talk to a webapp that talks to FS via mod_event_socket. In all of these scenarios I figure that I'll have the webapp running on the same box as FS and that I'll do all authentication/authorization of the user in the webapp. But which option should I go with? regards Oivvio

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Announcing switchio and the friends-of-freeswitch group on Github
by Tyler Goodlet 13 Nov '17

13 Nov '17

Hey all, This is to announce a new group a few of us ex-Sangoma people have created on Github: https://github.com/friends-of-freeswitch If any of you were familiar with Sangoma's project `switchy` this project has now been hard forked and evolved into this one: https://github.com/friends-of-freeswitch/switchio This new edition completely rewrites the core to use Python's 3.5+ new asyncio and coroutines and gets a performance boost due to dropping reliance on the ESL SWIG module. The project contains a full-blown auto-dialer and was originally built for stress testing using FreeSWICH clusters. The docs are still being updated after the recent rewrite so bear with us. Our vision for this group is to start aggregating FreeSWITCH related projects on Github into one place so as to provide a one-stop shop for community maintained tools. If anyone is interested in adding an existing project and joining the group fire us an email and we'll gladly welcome you! We'd also appreciate any opinions on or help with any of the projects if you find them useful or interesting to build-on. Currently our small crew is chatting on riot: https://riot.im/app/#/room/#freeswitch:matrix.org Most of us can also be found in the standard IRC and hipchat rooms. Thanks! Tyler Goodlet

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DID same freeswitch routing
by Christopher Trower 13 Nov '17

13 Nov '17

What is the best way to route DID's internally on freeswitch? For example two pbx system each assigned their own DID. And from the upstream sip provider to freeswitch and then pbx system works fine. Same way when dialing out from the pbx system. But when you dial from one PBX system with that DID to another PBX system with a different DID it connects but then also attempts to connect with the upstream SIP provider. Currently the pbx are setup via ACL.

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Cannot connect to ESL via perl
by 王聡 13 Nov '17

13 Nov '17

Hey all, I had made an ivr system based on perl and esl connectivity, which looks like below: while(1){ if(!$self->{esl}->connected()){ $self->{esl} = new ESL::ESLconnection("127.0.0.1", "8021", "ClueCon"); print "connecting...\n"; next; sleep(1); } ... ... my $t_from = $self->{esl}->api("uuid_getvar $row[0] sip_from_user")->getBody(); if(funca($t_from)){ $self->{esl}->api("uuid_kill $row[0]"); } .... .... sleep(1); } Most of the time it works well, but sometimes the esl connection returns unconnected and the whole ivr system got freeze. [ !$self->{esl}->connected() ] returned false. I had checked console and no error found. Actually it will recover by itself after about 1 hour, but it makes me annoying for the random call failure. Is there any solution about this connection failure? FreeSWITCH version : FreeSWITCH Version 1.4.26+git~20160205T175853Z~ca9207aa32~64bit (git ca9207a 2016-02-05 17:58:53Z 64bit) Regards.

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Record module for video calls
by 王聡 13 Nov '17

13 Nov '17

Hi All, Recently I need to record all contents for video calls on my FS server to make sure no illegal contents, and I found mod_vlc (https://wiki.freeswitch.org/wiki/Mod_vlc <https://wiki.freeswitch.org/wiki/Mod_vlc>) which might helps. However, I got some troubles during my build, it shows no dependencies for this mod. making all mod_vlc make[4]: Entering directory '/usr/src/freeswitch/src/mod/formats/mod_vlc' Makefile:892: *** You must install libvlc-dev to build mod_vlc. make[4]: Leaving directory '/usr/src/freeswitch/src/mod/formats/mod_vlc' Makefile:653: recipe for target 'mod_vlc-all' failed make[3]: *** [mod_vlc-all] Error 1 make[3]: Leaving directory '/usr/src/freeswitch/src/mod' Makefile:563: recipe for target 'all-recursive' failed make[2]: *** [all-recursive] Error 1 make[2]: Leaving directory '/usr/src/freeswitch/src' Makefile:3132: recipe for target 'all-recursive' failed make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory '/usr/src/freeswitch' Makefile:1402: recipe for target 'all' failed make: *** [all] Error 2 I had already installed the libvlc-dev but it still can’t built correctly. Is there any idea? Or any other way for video call recordings? Regards.

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Session Interval Too Small
by Supun Madushanka 10 Nov '17

10 Nov '17

Hi I installl freeswitch debian jessie 8.9 server. i register 2 extension (1001,1002)using tryjs sip when i call one to other it response SIP/2.0 422 Session Interval Too Small Via: SIP/2.0/WSS 7j9v0vaho6jt.invalid;branch=z9hG4bK2460026;received=123.231.122.157;rport=43751 From: "jay" <sip:1002@webrtc01.resolve1.com>;tag=gc5ckl91vp To: <sip:1001@webrtc01.resolve1.com>;tag=mtB2UD7UDgpHB Call-ID: s8slq1jnq7o7ic92lbo6 CSeq: 5609 INVITE User-Agent: FreeSWITCH-mod_sofia/1.9.0-1785-9d96741~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Min-SE: 120 Content-Length: 0 I change the below line <param name="session-timeout" value="1800"/> <param name="enable-timer" value="true"/> <param name="minimum-session-expires" value="120"/> but still get that respone could you help me -- Best Regards, *Supun Madushanka* [Undergraduate] University of Moratuwa. http://www.mrt.ac.lk Department of Computer Science and Engineering. http://cse.mrt.ac.lk Mobile: +94 71 1135012 <%2B94%280%29%20711135012> E-mail: supun.12(a)cse.mrt.ac.lk

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read vs. write codec channel variables clarification needed for CDR purposes
by Andrew Keil 10 Nov '17

10 Nov '17

To FreeSWITCH Users, I am just looking for confirmation of the following: 1. If I receive an inbound call to FreeSWITCH (the codec is negotiated and the call is answered by FreeSWITCH) and there is NO call bridging done, then am I correct that only the read_codec & read_rate channel variables apply (for CDR purposes)? 1. If I make an outbound call from FreeSWITCH (using originate api command) and once again there is NO call bridging done, then am I correct that only the write_codec & write_rate channel variables apply (for CDR purposes)? 1. When using mod_conference (for conferencing two inbound calls together or one inbound and one outbound (using originate api command) together) does this affect the read_codec & read_rate and write_codec & write_rate channel variables in any way? The reason I ask this is I noticed the following on an outbound call made (using the originate api command) recently and wanted to better understand what the read_codec & read_rate are used for when NO call bridging is done: read_codec: L16 read_rate: 8000 write_codec: PCMA write_rate: 8000 I appreciate the clarification. Andrew

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disturbance in conference
by Ketan Kothari 10 Nov '17

10 Nov '17

Hello there, We tried to do conference call using freeswitch that time i heard some disturbance but how can we remove that disturbance.So which parameter need to set for clear sound. We find the solution but how to configure that solution in freeswitch.We have find solution in below link. https://wiki.freeswitch.org/wiki/VAD_and_CNG

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Hangup Causes 408
by Supun Madushanka 10 Nov '17

10 Nov '17

Hi I install freeswitch debian 8.9 jessy. i register two user using jsSip(1001 and 1002). i can ring the one party to other and can answer. but few time it close. no audio no video HangUp causes = 408 could you helpme -- Best Regards, *Supun Madushanka* [Undergraduate] University of Moratuwa. http://www.mrt.ac.lk Department of Computer Science and Engineering. http://cse.mrt.ac.lk Mobile: +94 71 1135012 <%2B94%280%29%20711135012> E-mail: supun.12(a)cse.mrt.ac.lk

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