August 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

March 23rd Cluecon Weekly lost?
by Tom Lynn 17 Dec '16

17 Dec '16

I've tried to play back the March 23rd entry of Cluecon Weekly on Youtube.com but keep getting "An error Occurred. Please try again later." Can this episode be reposted? or is it lost? Thank you. Tom

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FS6947 - Tuning Opus bandwidth
by Emrah 26 Oct '16

26 Oct '16

Hi list, I re-read FS6947 and don't understand how this problematic was addressed and the issue fixed. The scope is simple. There should be a setting in the dialplan that allows downsampling of Opus for applications that do not require the 48khz / 2 channels framework. I.e.: terminating to the PSTN with Opus to take advantage of low bandwidth and great PLC. There seems to be a lot of confusion around bandwidth in general there. It doesn't matter if the internal clock of the device is always sampling at 48khz / 2ch. There are settings that can facilitate a lower bandwidth consumption for particular use cases, and it seems the reason it is not being implemented in FS is just a matter of being confused about the intent of the 48khz 2ch base. Please revisit this issue. FS should allow tuning of Opus audio / network bandwidth in the dialplan. It would optimize greatly lots of use cases. If I'm calling the PSTN, I'd rather have my client downsample and stream at a lower bandwidth, even if my audio capture would still be at 48khz / 2ch as per the RFC, and save on bandwidth, than transcode the full 48khz spectrum into PCM on my FS and minimize processing power on the client's side. Jira here: https://freeswitch.org/jira/browse/FS-6947 Emrah

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Capturing the Current Call Audio
by Leonardo Lima Ribeiro 13 Oct '16

13 Oct '16

Hi! I’m tying to capture the current call audio but I have no idea on how to do that. My main achievement is to get the current call audio and them stream this audio bytes to an API via HTTP Post. I was doing some tries with mod_shout and mod_vlc, and I was successful in the streaming part - I tested some wavs and mp3 files. But the mainly part is that I need the current call audio, and it has no file in the system to stream right? I was thinking if there’s a command that I can get the audio bytes from the current call but I really don’t know, I could not find any option like that. Could you please help me? Thank you, Leonardo Ribeiro

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Kernel Hyperthreading ON or OFF?
by Raimund Sacherer 06 Oct '16

06 Oct '16

Hi, we have a new Freeswitch cluster with Dell R410. It has 2 CPUs (Xeon E5620, QuadCore. it's 8 cores withouth HT and 16 cores with HT. I googled with hyperthreading and freeswitch but some say turn it on it will give you a boost, others say turn it off, you will gain performance. Does anyone know if it's better to leave it on or turn it off? And some small explanation? I would be very gratefull for your help, best regards Raimund --

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Audio Control on mp3/wav
by Jai Rangi 03 Oct '16

03 Oct '16

Checking to see if it is possible to have controls on audio files played with playback or filestream. Expected controls are keys to rewind, forward, fast-forward, pause, play. Jai

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Conference recording in stereo
by Oleg Stolyar 09 Sep '16

09 Sep '16

Hi guys, is there a way to record conferences to wav files in stereo? Documentation for RECORD STEREO variable mentions Leg A and Leg B, so I am not sure it will work for a conference.

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Bottleneck between mod_xml_radius and freeradius client?
by Juan Pablo Godoy 05 Sep '16

05 Sep '16

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console siptrace filtering
by David Villasmil 02 Sep '16

02 Sep '16

Hello guys, You can get all SIP messages coming and going to a specific profile by typing on the console: “sofia profile [PROFILE] siptrace on” But if you have traffic on the box, this is a NIGHTMARE, as you get ALL messages… So I made a modification to mod_console.c to add a filtering for siptraces. By enabling the filter like so: “console filter [expression]” Where “expression” can be anything present in the packets you want to get, be it username, dialed number, whatever. Only those messages on which the expression was found will be shown: freeswitch@MacBookPro-David> console USAGE: -------------------------------------------------------------------------------- console help console loglevel [[0-7] | <loglevel_string>] console filter <filter-expression> <<<——— NEW functionality console colorize [on|off|toggle] ———————————————————————————————————————— freeswitch@MacBookPro-David> console filter USAGE: -------------------------------------------------------------------------------- console filter <filter-expression|off> Enable, disable filtering, Only messages containing the 'filer-expression' will be printed out. set 'filter-expression' to 'off' to stop filtering ———————————————————————————————————————— Typing: console filter 1001 Would result in only seeing messages which include 1001 EXPRESSION (1001) FOUND IN: <<<recv 1255 bytes from tcp/[127.0.0.1]:52917 at 21:43:11.739980: ------------------------------------------------------------------------ INVITE sip:1001@127.0.0.1 SIP/2.0 Call-ID: 6f9cdc511446251e254e9e617d26647c@0:0:0:0:0:0:0:0 CSeq: 2 INVITE From: "1000" <sip:1000@127.0.0.1>;tag=45dd865b To: <sip:1001@127.0.0.1> Max-Forwards: 70 Contact: "1000" <sip:1000@127.0.0.1:52917;transport=tcp;registering_acc=127_0_0_1> User-Agent: Jitsi2.8.5426Mac OS X Content-Type: application/sdp Via: SIP/2.0/TCP 127.0.0.1:52917;branch=z9hG4bK-343435-1dc76531757a26482fa8ff1bac9acf3b Proxy-Authorization: Digest username="1000",realm="127.0.0.1",nonce="1a2be9b9-fa12-4763-b00c-dfeacc79f9fd",uri="sip:1001@127.0.0.1",response="fce7bf74772a149fe39ec7e50ec61c69",algorithm=MD5,qop=auth,cnonce="xyz",nc=00000001 Content-Length: 566 v=0 o=1000-jitsi.org 0 0 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 5119 RTP/AVP 96 a=rtpmap:96 opus/48000/2 a=fmtp:96 usedtx=1 a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=rtcp-xr:voip-metrics m=video 5121 RTP/AVP 97 99 a=recvonly a=rtpmap:97 H264/90000 a=fmtp:97 profile-level-id=4DE01f;packetization-mode=1 a=imageattr:97 send * recv [x=[0-1920],y=[0-1200]] a=rtpmap:99 H264/90000 a=fmtp:99 profile-level-id=4DE01f a=imageattr:99 send * recv [x=[0-1920],y=[0-1200]] ------------------------------------------------------------------------ >>> My goal is to extend this to also filter other messages, like call processing, etc. (ERR messages would not be filtered out) I’m wondering whether it is possible to do a pull request on this? Regards, David ᐧ

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Getting, sofia_glue.c:1031 Error creating HANDLE!
by devang nathwani 02 Sep '16

02 Sep '16

Hello, While routing the call from opensips server to freeswitch server i am getting this error in fs_cli log, 2016-08-26 14:05:13.611542 [CRIT] sofia_glue.c:1031 Error creating HANDLE! url_str=[sip:357xxxxxxxx@178.xxx.xxx.xxx] call_id=[N/A] to_str=[<sip:357xxxxxxxx@178.xxx.xxx.xxx>] from_str=["<null>" <sip:<null>@172.16.3.2>] invite_contact=[<sip:gw+IPComms@172.16.3.2:5060;transport=udp;gw=gtw-name>] I have tried configuring 1.6.8 and 1.6.9 versions of freeswitch but getting the same error. What am i missing?

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One-way audio with Verto + Chrome (with video enabled)
by Joel Serrano 02 Sep '16

02 Sep '16

Hi, I'm getting started with WebRTC, Verto, etc. I'm facing an issue with chrome+video, I get one-way-audio. Scenario: I login to Verto and I call a conference that has other users in it (those users enter the conference by SIP). 1- Chrome: If video is enabled, I can send audio and other users will hear me but I can't hear them. 2- Chrome: If video is disabled, all works correctly. 3- Firefox: with video enabled, all works correctly. This definitely looks like a config issue but I have no idea where to start looking. I found this JIRA https://freeswitch.org/jira/browse/FS-7679 but in my case, I think the correct candidate is being selected. Should I attach the full logs from all 3 test calls? Please let me know also if I should open a new JIRA for this, I just want to make sure this isn't a config issue first. Thanks! Joel.

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