- FreeSWITCH-users - FreeSWITCH Mailing Lists

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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ClueCon Weekly – December 16, 2015 – Italo Rossi and the Evolux Call Center System
by Ken Rice 11 Jan '16

11 Jan '16

New Post on freeswitch.org from Kathleen King check it out at http://ift.tt/1UJHnuy ClueCon Weekly – December 16, 2015 – Italo Rossi and the Evolux Call Center System

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ClueCon Weekly – December 9, 2015 – Tsahi Levent-Levi
by Ken Rice 11 Jan '16

11 Jan '16

New Post on freeswitch.org from Kathleen King check it out at http://ift.tt/1UJHhTK ClueCon Weekly – December 9, 2015 – Tsahi Levent-Levi The challenges in testing WebRTC services How do you view WebRTC? As a VoIP technology or an Internet one? There’s a huge divide between the two, especially when it comes to testing them. WebRTC sits in the intersection between them, making it doubly hard to grok – and test. Tsahi will outline in this session a few of the main challenges, with suggestions on how to fix them. He will also use the opportunity to show how testRTC.com addresses these challenges.

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Developer friendly SIP trunk provider for experimentation?
by Oivvio Polite 11 Jan '16

11 Jan '16

I'm just getting started with FS and want to hook up a couple of SIP trunk for experimentation. Right now I'm looking for a provider that'll let me pick numbers from a web interface and pay with credit card. Many of the providers I come across when googling seem to only sell SIP trunks in bulk and want me to contact a sales rep to get started. That might make sense later on in the project but right now I'm looking for something more light weight. Any tips?

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Re: [Freeswitch-users] Developer friendly SIP trunk provider for experimentation?
by Lyle Pratt 11 Jan '16

11 Jan '16

Try Twilio. They provide traditional SIP Trunking services now and signing up only takes a couple of minutes. https://www.twilio.com/sip-trunking Regards, Lyle Pratt

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gateway var in channel vars
by Ximo Peral 11 Jan '16

11 Jan '16

Hi, I'm trying save in cdr_csv a gateway variable, but not work for me. This is the conf for sofia: <gateway name='XXXXX'> <variables> <variable name="gateway_id" value="3"/> </variables> </gateway> I'm trying to save variable from the dialplan with: <extension name="national" continue="true"> <condition field="destination_number" expression="^([98765]\d{8}|[23]\d{3})$" break="on-true"> <action application="set" data="exec_after_bridge_app=set gateway_id=\${sofia_gateway_data \${sip_gateway_name} var gateway_id}"/> <action application="bridge" data="sofia/gateway/${distributor(XXXX ${sofia(profile external gwlist down)})}/${destination_number}"/> </condition></extension> In console it seems to work, but not generate channel variable: 2016-01-11 14:45:57.802303 [DEBUG] switch_cpp.cpp:1103 sofia/external/xxxxx(a)xx.xx.xx.xx destroy/unlink session from objectEXECUTE sofia/external/965662225(a)78.136.81.1 set(exec_after_bridge_app=set gateway_id=${sofia_gateway_data ${sip_gateway_name} var gateway_id})2016-01-11 14:45:57.802303 [DEBUG] mod_dptools.c:1498 SET sofia/external/xxxxx(a)xx.xx.xx.xx [exec_after_bridge_app]=[set gateway=]EXECUTE sofia/external/965662225(a)78.136.81.1 bridge(sofia/gateway/XXXX/xxxxxx)2016-01-11 14:45:57.802303 [DEBUG] switch_channel.c:1237 sofia/external/xxxxx(a)xx.xx.xx.xx EXPORTING[export_vars] [t38_passthru]=[true] to event2016-01-11 14:45:57.802303 [DEBUG] switch_ivr_originate.c:2127 Parsing global variables2016-01-11 14:45:57.802303 [NOTICE] switch_channel.c:1091 New Channel sofia/external/622073738 [56086a63-c6e9-4c11-b1a4-316165d3bc20]2016-01-11 14:45:57.802303 [DEBUG] mod_sofia.c:4749 sofia/external/xxxxxx setting variable [gateway_id]=[1] Is this possible?

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Knowing who is talking in a conference
by ik 11 Jan '16

11 Jan '16

Hello, How can I know what channel (uuid) is talking at a specific conference room? Thanks, Ido

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Spanish Language error
by Annus Fictus 09 Jan '16

09 Jan '16

Hello, I'm trying to use spanish prompts (maria voice) in FreeSWITCH without success. When i call voicemailmain, I can see in FreeSWITCH console, this error: [ERR] switch_xml.c:3250 Can't find phrases tag. How can i solve? Regards

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Need Help with mod_portaudio
by Pete Kay 09 Jan '16

09 Jan '16

Hi, I tried to stream audio from aloop to outside and stream outside audio to bloop. Then I want to do : originate user/1000 bridge:portaudio/auto_answer inline Here is my pa devlist, but I tried pa outdev and pa indev and i got "indev not set (invalid value)" Does anyone know how I should set it up so media can stream? pa indev 5 indev not set (invalid value) freeswitch@vps57327> pa devlist 0;Loopback: Loopback PCM (hw:0,0)(ALSA);32;32;r,i,o 1;Loopback: Loopback PCM (hw:0,1)(ALSA);2;32; 2;amix(ALSA);0;2; 3;asnoop(ALSA);2;0; 4;aduplex(ALSA);2;2; 5;aloop(ALSA);128;128; 6;bloop(ALSA);2;0; 7;card0(ALSA);128;128; 8;default(ALSA);128;128;

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