January 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Unable to hear audio with mod_portaudio
by Pete Kay 13 Jan '16

13 Jan '16

Hi, I tried to use mod_portaudio to bridge a local sound card/buffer to a channel, but no media is heard. Could someone please give me some idea what could go wrong? Is there any working example I can follow? How do I best debug this issue? Any help will be greatly appreciated. P

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Dial multiple users for "conference" call
by Michael Nielsen 13 Jan '16

13 Jan '16

Is it possible to dial multiple users at once, for initiating a conference call - without the need for a conference room. And without having the client as "connection-point" for all the parties involved, but having FreeSWITCH handling the merging of calls, so the RTP stream for each client is similar to a 1-1 call for all parties.

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GSMOpen restarting freeswitch on network loss (?)
by Jude Mukundane 13 Jan '16

13 Jan '16

Hello Freeswitch Users, I have come upon an issue that I need clarification on. I am using mod_gsmopen with huawei E173u-1 Modems. They work fine. I noticed though for one of the networks that are weak, every time the signal is lost, (double green blink on modem), freeswitch restarts. some thing triggers "/bin/bash /etc/init.d/freeswitch restart. Is this by design in GSM open? I was considering moving this line to a GoIP if that is the case so that Freeswitch is not made unavailable for all the other lines by one line. regards, Jude Mukundane CTO - RootIO http://rootio.org

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Mod_verto + Haproxy POOL Error
by Murugan Pandian 13 Jan '16

13 Jan '16

Hi, We have tried to run Freeswitch mod_verto(WS) behind Haproxy(WSS). But When i tried to run the client(verto.js websocket connection) disconnected and getting reconnected for every 60sec(1 min). If i make a call, it is getting disconnected and reconnected again for every 1 min. If i use mod_verto directly without Haproxy it is working fine without any issues. Note: My mod_verto is running on port 8081(WS). But when i tried to run behind my Haproxy running on WSSi am getting the above issue. Thanks.

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Building ESL library by itself?
by Nathan Neulinger 13 Jan '16

13 Jan '16

Has there been any consideration given to adding a sub 'configure' to the libs/esl/ dir in build tree and possibly extracting a tarball for just libesl? The requirements for that configure could be significantly reduced, allowing the esl libs (and wrappers) to be built by themselves on other platforms - even those where building/supporting the full FS (especially 1.6) is problematic. Is there any strong reason AGAINST having this capability? -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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deflect vs redirect vs bridge
by ik 12 Jan '16

12 Jan '16

Hello, I have an incoming call that on some occurrences, instead of answering it, I need to "move" the call to a different Freeswitch server. I have tried to use deflect, and redirect, however the call with deflect just hangup immediately, and with redirect, it arrive to timeout and then hangup. I'ved noticed at the wiki page (confluence) that the call must be answered with deflect. Only when I use bridge, the other Freeswitch see's the SIP request and handle the call properly, but as a statful request. How can I make the original Freeswitch out of the routing or at least make it stateless redirect (302) instead of stateful one? Thanks, Ido

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FS conference limitations
by Bote Man 12 Jan '16

12 Jan '16

This is probably best answered by Anthony or Mike J. I need to make extensive use of the "relate" conference API command to eliminate feedback between a bunch of VoIP speakers that have microphone talkback capability. Is there a practical limit on how many "relations" I can set up to mute conferees from hearing each other? I have 16 endpoints in one building and probably 8 to 10 of them need to be muted relative to their nearest neighbors so that list can get fairly large I imagine. I just don't know how far I can push it. Then I have to become a wizard at Lua script programming in a matter of a few hours to make this all work. Pointers to help are greatly appreciated. If I get stuck look for a bounty request :-) I'm thinking I can set up a linked-list populated with the extension number (which I know and is fixed) paired with the conference user id (which varies with each call, but is used by the 'relate' command). I'm just going to make it very simple with everything defined in the script header since it won't change. Once this is done I hope to donate it back to the community in the rare chance that somebody else finds it useful. Thanks! Bote

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FreeSWITCH’s Birthday!!
by Ken Rice 12 Jan '16

12 Jan '16

New Post on freeswitch.org from Kathleen King check it out at http://ift.tt/1TQ0aEy FreeSWITCH’s Birthday!! Hey FreeSWITCHers! Last week FreeSWITCH turned ten! It has been a long road, With many, many lines of code. The community has been there through thick and thin, And witnessed all the hard work put in. Long nights, no break, But, a wonderful piece of software they did make. Blood, sweat, tears, and no perks, Would result in a switch that just works! We had no idea they would be so merciless, So, we apologize on behalf of Rachael from Cardholder services. Convey your appreciation with some flair, Show Tony how much you care, By clicking on link below, And, send him something as thanks for Verto. http://goo.gl/3WCNnI

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WebRTC client to traditional video phone, no video
by SamyGo 11 Jan '16

11 Jan '16

Hi all, I'm trying to test video calls between WebRTC clients and the video capable IP phones. Originating calls from H264 based IP Phone to LinPhone results in two way video stream no errors on console, replacing LinPhone with WebRTC client makes video disappear, black screen only but two-way audio only. I'm testing with following version: *FreeSWITCH Version* 1.7.0+git~20151103T161102Z~cffc673879~64bit (git cffc673 2015-11-03 16:11:02Z 64bit) *OS version:* Distributor ID: Debian Description: Debian GNU/Linux 8.2 (jessie) Release: 8.2 Codename: jessie WebRTC Client is SIPML5 on Firefox. mod_vpx, mod_av, and mod_h26x are loaded. I've tried unload mod_h26x and see if it makes any difference but doesn't change anything. Console logs along with sip traces can be found on this pb link: https://pastebin.freeswitch.org/24517 Since I've no trouble with video call between LinPhone and IP Phone I suspect its more of a problem related to WebRTC and not VP8-H264 transcoding issue. Need some guidance in this regard. Thanks, Sammy

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ClueCon Weekly – Jan 6, 2016 – Dave Horton
by Ken Rice 11 Jan '16

11 Jan '16

New Post on freeswitch.org from Kathleen King check it out at http://ift.tt/1JFNAr2 ClueCon Weekly – Jan 6, 2016 – Dave Horton Dave Horton discusses his open-source, node.js-based SIP load balancer for FreeSWITCH servers (http://ift.tt/1JFNxLQ…) which is currently deployed in several service provider networks.

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