December 2020 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Fwd: [CFP] FOSDEM 2021, RTC devroom, speakers, volunteers neeeded
by FOSDEM RTC Team 23 Dec '20

23 Dec '20

FOSDEM - Real Time Communications devroom CfP ============================================= NOTE: we have extended the deadline but we will give preference to people who submitted earlier when we set the time for each talk. Please submit your proposal ASAP so that you can get your preferred time slot. Overview -------- [FOSDEM](https://fosdem.org) is one of the world's premier meetings of free software developers, with over five thousand people attending each year. FOSDEM 2021 takes place 6-7 February 2021 and for the very first time, it will be online. This document contains information about: - Real-Time Communications developer room (devroom) - speaking opportunities - volunteering New rules for the online edition -------------------------------- This year FOSDEM will be fully online instead of being held in Brussels, here are the most important things to know about this (quite significant) change: - The reference time will be Brussels local time (CET) - Talks will be pre-recorded in advance, and streamed during the event - The Q/A session will be live - A facility will be provided for people watching to chat between themselves - A facility will be provided for people watching to submit questions Call for participation - Real Time Communications (RTC) ------------------------------------------------------- The Real-Time devroom is about all things involving real-time communication, including: XMPP, SIP, WebRTC, telephony, mobile VoIP, codecs, peer-to-peer, privacy and encryption. **We are looking for speakers for the devroom and volunteers who can help manage the scheduling and live Q&A sessions.** The devroom is only on Saturday, 6th of February 2021. To discuss the devroom, volunteer or ask questions, please join the [Free-RTC mailing list](http://lists.freertc.org/mailman/listinfo/discuss). ### Key dates - 20th Dec: Submission deadline (extended to 8 January) - 24th Dec: Announcement of selected talks - 15th Jan: Presentations upload deadline - 6th & 7th Feb: Conference dates (online) - 6th Feb: RTC devroom date (online) ### Speaking opportunities Note: if you used FOSDEM Pentabarf before, please use the same account/username Real-Time Communications devroom: deadline 23:59 UTC on 20th of December. Please use the [Pentabarf](https://penta.fosdem.org/submission/FOSDEM21/) system to submit a talk proposal for the devroom. On the "General" tab, please look for the "Track" option and choose "Real Time Communications devroom". ### First-time speaking? FOSDEM devrooms are a welcoming environment for people who have never given a talk before. Please feel free to contact the devroom administrators personally if you would like to ask any questions about it. This year this is more true than ever, being able to record your presentation offline without an audience in front can greatly help build up one's confidence! ### Submission guidelines The Pentabarf system will ask for many of the essential details. Please remember to re-use your account from previous years if you have one. In the "Submission notes", please tell us about: - The purpose of your talk - Any other talk applications (devrooms, lightning talks, main track) - Availability constraints and special needs You can use HTML and links in your bio, abstract and description. If you maintain a blog, please consider providing us with the URL of a feed with posts tagged for your RTC-related work. We will be looking for relevance to the conference and devroom themes, presentations aimed at developers of free and open source software about RTC-related topics. Please feel free to suggest a duration between 20 minutes and 55 minutes but note that the final decision on talk durations will be made by the devroom administrators based on the number of received proposals. As the two previous devrooms have been combined into one, we may decide to give shorter slots than in previous years so that more speakers can participate. Please note FOSDEM aims to record and live-stream all talks. The CC-BY license is used. ### Recording help The devroom organization is able to provide help with recording your session. The recording would be performed at a scheduled time with one of us, so you won't be alone giving your presentation. Minimal edits will be possible, but the ideal plan is to record it in one shot. Thanks Dan Jenkins for providing us with the means to do this! Volunteers needed ----------------- To make the devroom run successfully, we are looking for volunteers. This year many things be done for the first time, so all the help we can get is more than welcome. Spread the word and discuss --------------------------- If you know of any mailing lists where this CfP would be relevant, please forward this document. If this devroom excites you, please blog or microblog about it, especially if you are submitting a talk. If you regularly blog about RTC topics, please send details about your blog to the planet site administrators: - All projects https://planet.freertc.org planet(a)freertc.org - XMPP https://planet.jabber.org ralphm(a)ik.nu - SIP https://planet.sip5060.net planet(a)sip5060.net Please also link to the Planet sites from your own blog or web site as this helps everybody in the free real-time communications community. Contact ------- For any private queries, contact us directly using the address **fosdem-rtc-admin(a)freertc.org** and for any other queries please ask on the [Free-RTC mailing list](http://lists.freertc.org/mailman/listinfo/discuss). The devroom administration team: - Saúl Ibarra Corretgé <s(a)saghul.net> - Ralph Meijer <ralphm(a)ik.nu> - Daniel-Constantin Mierla <miconda(a)gmail.com> - Daniel Pocock <daniel(a)pocock.pro> - Guus der Kinderen <guus.der.kinderen(a)gmail.com>

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VLC Errormessage
by R G 20 Dec '20

20 Dec '20

Hi all, i have a problem with the mod_vlc Plugin. Regardless of whether i try to record a conference to stream or a file i get the following error message: 2020-12-19 22:08:09.523574 [INFO] mod_conference.c:433 Auto recording file: vlc://#standard{access=file,mux=raw,dst=/var/freeswitch/meetings/test.wav} 2020-12-19 22:08:09.523574 [DEBUG] conference_loop.c:1338 Setup timer soft success interval: 20 samples: 960 from codec opus 2020-12-19 22:08:09.523574 [DEBUG] conference_record.c:304 Setup timer success interval: 20 samples: 960 2020-12-19 22:08:09.523574 [ERR] mod_vlc.c:772 VLC error: cannot initialise VLC handle 2020-12-19 22:08:09.523574 [ERR] conference_record.c:277 Error Opening File [vlc://#standard{access=file,mux=raw,dst=/var/freeswitch/meetings/test.wav}] 2020-12-19 22:08:09.523574 [ERR] switch_core_timer.c:117 Timer is not properly configured. 2020-12-19 22:08:09.523574 [INFO] conference_record.c:425 Recording of vlc://#standard{access=file,mux=raw,dst=/var/freeswitch/meetings/test.wav} Stopped This is my conference.conf.xml: <param name="auto-record" value="vlc://#standard{access=file,mux=raw,dst=/var/freeswitch/meetings/test.wav}"/> I use a Ubuntu 16.04 and libVLC is installed: libvlc-dev/xenial-updates,xenial-security,now 2.2.2-5ubuntu0.16.04.4 amd64 [installed] libvlc5/xenial-updates,xenial-security,now 2.2.2-5ubuntu0.16.04.4 amd64 [installed,automatic] libvlccore-dev/xenial-updates,xenial-security,now 2.2.2-5ubuntu0.16.04.4 amd64 [installed] libvlccore8/xenial-updates,xenial-security,now 2.2.2-5ubuntu0.16.04.4 amd64 [installed,automatic] vlc-data/xenial-updates,xenial-updates,xenial-security,xenial-security,now 2.2.2-5ubuntu0.16.04.4 all [installed,automatic] Kind regards, Robert

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Tracking Call Leg through Callcenter
by Nathan Stratton 19 Dec '20

19 Dec '20

On calls flowing them my FreeSWITCH box I use: <action application="set" data="sip_h_In-Reply-To=${sip_call_id}"/> Before bridge, to correlate my call legs between my public and private networks. The problem is if I put that before application callcenter it is not added. Is there any way to make correlate legs going into and then out from callcenter? <extension name="HesslerKnivesSams-queue_en"> <condition field="destination_number" expression="^HesslerKnivesSams-queue_en$"> <action application="export" data="dialed_extension=$1"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="bypass_media=false"/> <action application="set" data="record_concat_video=true"/> <action application="set" data="enable_file_write_buffering=false"/> <action application="set" data="core_video_blank_image=/media/frame.png"/>  <action application="callcenter" data=" HesslerKnivesSams-queue_en(a)troc.vocinity.com"/> <action application="export" data="nolocal:absolute_codec_string=VP8,OPUS"/> <action application="bridge" data="sofia/gateway/gateway/HesslerKnivesSams-busy_en"/> </condition> </extension> ><> nathan stratton

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Reasons for call dropped/RFC2543 incompatible destination
by Robert Fitzjohn 19 Dec '20

19 Dec '20

Hello, FreeSwitch v. 1.6.20 I'm calling my own mobile no. via my localphone gateway, and while this used to work when I last tested it a while ago, recently the call no longer goes through, with freeswitch dropping the call with the RFC2543 warning and then 'incompatible destination'. Codecs on both sides look good & calling other mobile numbers on other networks via the same gateway works fine, for what it's worth. The fuller/anonymized paste is at: https://pastebin.freeswitch.org/view/ddafee03 but the bits which I think matter are below. Could someone give me any hints as to whether this is a FS issue or whether the problem lies with my gateway or otherwise. Thanks - recv 1205 bytes from udp/[94.0.0.0]:5060 at 10:21:18.751270: - ------------------------------------------------------------------------ - SIP/2.0 183 Session Progress - Via: SIP/2.0/UDP 45.0.0.0:5080;rport=5080;branch=z9hG4bKHNUQK4208H7Br - Record-Route: <sip:95.0.0.0:5070 ;lr;ftag=lp-2k9-5ee88476-00002a55-0005d2b3R2d70439b.a>,<sip:94.1.1.1;lr=on>,<sip:94.0.0.0;lr=on;ftag=B3N7U1NHeem0F;an=YWJjREVENjg5JiUnPDkyE2laNj48LXtVYWEGaiYpXA--> - To: <sip:447970000000@proxy.localphone.com >;tag=lp-2k9-5ee88476-00002a55-0005d2b3R2d70439b.b - From: "Robert" <sip:1234567@localphone.com>;tag=B3N7U1NHeem0F - Call-ID: 45818239-b574-1239-8a8e-5600002a6bbb - CSeq: 29244045 INVITE - Allow: PUBLISH,MESSAGE,PRACK,SUBSCRIBE,REFER,INFO,NOTIFY,OPTIONS,BYE,INVITE,ACK,CANCEL - Contact: <sip:94.2.2.2;did=2d6.a9862935> - Content-Type: application/sdp - Content-Length: 474 - - v=0 - o=UK1-SBC-15-1 70259913 70259914 IN IP4 0.0.0.0 - s=sip call - c=IN IP4 0.0.0.0 - t=0 0 - m=audio 0 RTP/SAVP 18 8 0 101 - a=rtpmap:18 G729/8000 - a=fmtp:18 annexb=no - a=rtpmap:8 PCMA/8000 - a=rtpmap:0 PCMU/8000 - a=rtpmap:101 telephone-event/8000 - a=fmtp:101 0-15 - a=ptime:20 - m=audio 0 RTP/AVP 18 8 0 101 - a=rtpmap:18 G729/8000 - a=fmtp:18 annexb=no - a=rtpmap:8 PCMA/8000 - a=rtpmap:0 PCMU/8000 - a=rtpmap:101 telephone-event/8000 - a=fmtp:101 0-15 - a=ptime:20 - a=nortpproxy:yes - ------------------------------------------------------------------------ - 2020-12-10 10:21:18.738771 [DEBUG] sofia.c:7084 Channel sofia/external/447970000000 entering state [proceeding][183] - 2020-12-10 10:21:18.738771 [NOTICE] sofia.c:7192 Ring-Ready sofia/external/447970000000! - 2020-12-10 10:21:18.738771 [DEBUG] switch_channel.c:3346 (sofia/external/447970000000) Callstate Change DOWN -> RINGING - 2020-12-10 10:21:18.738771 [DEBUG] sofia.c:7084 Channel sofia/external/447970000000 entering state [proceeding][183] - 2020-12-10 10:21:18.738771 [DEBUG] sofia.c:7094 Remote SDP: - v=0 - o=UK1-SBC-15-1 70259913 70259914 IN IP4 0.0.0.0 - s=sip call - c=IN IP4 0.0.0.0 - t=0 0 - m=audio 0 RTP/SAVP 18 8 0 101 - m=audio 0 RTP/AVP 18 8 0 101 - - 2020-12-10 10:21:18.738771 [WARNING] switch_core_media.c:3951 RFC2543 from March 1999 called; They want their 0.0.0.0 hold method back..... - 2020-12-10 10:21:18.738771 [NOTICE] sofia.c:7273 Hangup sofia/external/447970000000 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] - 2020-12-10 10:21:18.738771 [DEBUG] switch_core_state_machine.c:584 (sofia/external/447970000000) Running State Change CS_HANGUP (Cur 4 Tot 65) - 2020-12-10 10:21:18.738771 [DEBUG] switch_core_state_machine.c:850 (sofia/external/447970000000) Callstate Change RINGING -> HANGUP - 2020-12-10 10:21:18.738771 [DEBUG] switch_core_state_machine.c:852 (sofia/external/447970000000) State HANGUP - 2020-12-10 10:21:18.738771 [DEBUG] mod_sofia.c:438 Channel sofia/external/447970000000 hanging up, cause: INCOMPATIBLE_DESTINATION

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forward registration
by Joli Martinez 19 Dec '20

19 Dec '20

Hello, We have a several FQDN's that are pointing to a Freeswitch box and users are registering to it. I need forward those registrations for one domain only to a new freeswitch box I am building. Not sure how to do this.

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reINVITE
by David Villasmil 17 Dec '20

17 Dec '20

Hello all, Is there any way to know which commands trigger a reINVITE? I.e.: out of these: - uuid_break - uuid_bridge - uuid_broadcast - uuid_deflect - uuid_displace - uuid_record - uuid_transfer Anyone knows off the top of their head? Thanks David -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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FreeSWITCH call recording missing header information
by Chad Phillips 16 Dec '20

16 Dec '20

I have a dialplan setup that records a bridged call in FreeSWITCH, which sets the 'api_hangup_hook' variable to call a lua script when the call hangs up, and the called lua script moves the recorded .wav file from it's temporary location to a final destination. This works, but recently I noticed that some of my .wav recordings have no header information, which results in many players choking when trying to play it. I eventually figured out that the header information for the .wav file appears to be written *after* the call is hung up, and file move operation in the api_hangup_hook moves the file before the header information has been written. This occurs sporadically. I've found that introducing a 1 second sleep in my lua script fixes this race condition, but that seems a bit of an ugly hack. Curious if anybody else has run into this, and/or can offer me a cleaner solution than my sleep hack. I'm attaching a brief summary of the dialplan and lua script for anyone that wants to see the code.

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Re: [Freeswitch-users] Copy incoming audio from participants for ASR
by David P 16 Dec '20

16 Dec '20

How do you currently forward audio to kaldi? If you use an FS conference, I think there is no way to separate the audio per-user. I agree it would be very useful if there were a way to process each user's audio separately. After a few years of using FS, I haven't come across a way to do it. --------- Forwarded message ---------- > From: R G <mitachundkrach(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Bcc: > Date: Mon, 14 Dec 2020 12:40:38 +0100 > Subject: [Freeswitch-users] Copy incoming audio from participants for ASR > Hello everyone, > > i have a technical question about freeswitch and would like to know if > my idea is technically possible. > > I've got a project at the university and my goal is to improve automatic > speech recognition for bigbluebutton. > > In the summer semester a group of students of my university made a > subtitling plugin[1] for bigbluebutton. This plugin adds a participant > into the meeting and grabs his audiostream to send it to the kaldi ASR > Software. When more than one person speaks at a time the detection > decreases. > > My idea is to copy the incoming freeswitch audiostreams from every > participant and send them directly as a copy to the ASR (kaldi can > handle multiple audiostreams at once). > > In the Confluence-Wiki i searched for something in this direction but > only found things like Dialplan to ring multiple telephones at once. > > I hope my idea is formulated understandably. > > > Kind regards, > > Robert > > [1] https://github.com/3wille/bbb-kaldi-connector > >

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Full Trickle ICE via mod_sofia supported?
by Alexander Haugg 16 Dec '20

16 Dec '20

Hello, Currently I'm trying to implement the Full Trickle mechanism as described here https://tools.ietf.org/id/draft-ietf-mmusic-trickle-ice-sip-11.html Since I haven't succeeded so far, I have the following questions. Is Full Trickle supported via mod_sofia? If so, which settings or sip headers are required? Thanks Alex

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disable NOTIFY message
by Aidar Kamalov 16 Dec '20

16 Dec '20

Hello, is it possible to disable NOTIFY messages from freeswitch to external sip profile? NOTIFY sip:aaaa@aaaaaaa;intercom=true SIP/2.0 .... Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Event: talk Allow-Events: talk, hold, conference, refer Subscription-State: terminated;reason=noresource Content-Length: 0 -- Aydar A. Kamalov

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