- FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_python and logs
by David Villasmil 09 Nov '20

09 Nov '20

Hello all, So I'm using fluentd to ship all my logs and i enabled the printing of the UUID to the log on every line. Thing is when printing from a python script, there's no UUID (the script is triggered by an incoming call). Looking at the code i don't think there's a way of printing it via config, but I wanted to confirm this. Thanks and Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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use-vbr vs adjust-bitrate in opus.conf.xml
by E. Schmidbauer 06 Nov '20

06 Nov '20

in opus.conf.xml what is the difference between adjust-bitrate and use-vbr ? doesn’t variable bit rate (vbr) adjust the bitrate? or do you need both enabled for that to actually work? or do they contradict each other? Any info on this would be greatly appreciated and i would love to document the answer on confluence https://freeswitch.org/confluence/display/FREESWITCH/mod_opus

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Hey Join Me tomorrow 10 am central (11/06/2020)
by Brian West 05 Nov '20

05 Nov '20

Join us here https://cluecon.cantina.video/rooms/ClueConCafe Lets have some fun and fix some bugs. SEE YOU THERE! /b -- Brian West | Co-founder and Developer Need Commercial support? email sales(a)freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&e…> Email: brian(a)freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> [image: https://www.facebook.com/signalwireinc?src=email] <https://www.facebook.com/freeswitch> [image: https://twitter.com/freeswitch] <https://twitter.com/freeswitch>

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SIP Chronicles, recap and upcoming episode on Friday @ 7pm UTC
by Maxim Sobolev 05 Nov '20

05 Nov '20

Hey SipTubies, Just a quick recap: if you missed any of latest episodes of SIP Chronicles here we go: Very interesting historical-looking firegrill-side chat with Paul Kyzivat of IETF: https://youtu.be/kvLuJhsnF-c Great conversation with Sandro Gauci (SIPVicious, Enable Security) about recent developments: https://youtu.be/kvLuJhsnF-c For #12 I think we'll do something different: turn the lens around and do a live coding / testing / whatever jam session with Turbomax at controls doing all 3 of those things at once as well as answering questions from the audience. The official goal is to benchmark our RFC8760 implementation under simulated fire and see how it fares to the old-good master branch with the sub-target to measure a relative drop/increase of peak performance of processing digest auth. In the process I expect to get OpenSIPS up flying at x00,000 CPS altitude and land it safely at few improvements. Some pics of the workhorse we are going to bench with are attached, this is a small but mighty 8-core Xeon-D 1541 with 64GB of ECC RAM. We'll see if simulated fire erupts into a real one. We will go over the process of setting up OpenSIPS on Ubuntu 20.x and using either https://github.com/sippy/b2bua or https://github.com/sobomax/microsippy or combination of both to source incoming traffic and measure latency. https://youtu.be/SYzqka4I34M Bring your own beer, snacks - it's going to be a long session and see you tomorrow. :) -Max

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Regarding verto issue
by Alaghu Sundram 05 Nov '20

05 Nov '20

Hello, We configured both verto and signlawired(sip profile). Signalwire registerd. We tried to connect through verto client . We cant get any error regarding verto. Even we configured verto as per https://freeswitch.org/confluence/display/FREESWITCH/mod_verto. In above attachement you can found configuration of verto and sip_profiles. Please help us to solve this issue Thanks

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Detect when an incoming call has been forwarded through another number
by Chad Phillips 04 Nov '20

04 Nov '20

Is there any way in FreeSWITCH to detect when an incoming SIP call has been forwarded through another number? Example: Phone1 --> Phone2 --> FreeSWITCH Usually caller ID info comes through for Phone1, which makes sense. But we'd like to *also* know that the call came through a forward by Phone2. Use case: occasionally someone will fat-finger a phone forward and forward their phone to our customer service number. We'd like to toss those calls or play some kind of "sorry, you can't do this" message. Chad

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Regarding freeswitch error
by Alaghu Sundram 03 Nov '20

03 Nov '20

Hello all, New to Freeswith and I am working on setting up a sip trunk to register freeswicth to external service provider. I able to register. Please check following freeswitch(a)fsm.intelliverse.com> sofia status profile Profile::Gateway-Name Data State Ping Time IB Calls(F/T) OB Calls(F/T) ================================================================================================= external::example.com sip:joeuser@example.com NOREG 0.00 0/0 0/0 external::vitelity-inbound sip:inte_01@inbound32.vitelity.net REGED 0.00 0/0 0/0 external::vitelity-outbound sip:inte_01@outbound.vitelity.net REGED 0.00 0/0 0/0 external::signalwire-inbound sip:d038652d108e925f7188aa0458e5c63b@orchestrate-c9b1086b33234be59b9be2c4193f7fbd.sip.signalwire.com REGED 0.00 0/0 0/0 external::signalwire-outbound sip:d038652d108e925f7188aa0458e5c63b@orchestrate-c9b1086b33234be59b9be2c4193f7fbd.sip.signalwire.com NOREG 0.00 0/0 0/0 ================================================================================================= 5 gateways: Inbound(Failed/Total): 0/0,Outbound(Failed/Total):0/0 Whenever I login through verto. I got terminate register message. Please help us to solve this issue. I provide error in attachments Regards, R.Alaghu sundram

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distributed sqlite
by Abaci B 03 Nov '20

03 Nov '20

I was wondering if anyone is working on implementing in freeswitch support for dqlite https://github.com/canonical/dqlite to support a distributed freeswitch system for core database.

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FreeSWITCH patch: `uuid_fileman` to preserve pitch on playback speed change
by Gulyás Attila 02 Nov '20

02 Nov '20

Good Morning, Would it be possible to merge this pull request, and add it to the next FreeSWITCH release? https://github.com/signalwire/freeswitch/pull/244 Tried to get a response on Slack and by commenting in the pull request, but no joy. I totally understand that all the devs are busy, and that this is probably one of the lowest priority issues out there, but here's some context why I am being a pest about this: Our Access News phone service is for visually and print impaired individuals who are unable to consume conventional print materials. We just replaced an old, on-premises system with aFreeSWITCH solution, and it has been a dream to work with, except this one flaw. Non-visual users rely heavily on changing the playback speed to efficiently browse through vast amounts of audio content, and the current behaviour is very disruptive, especially because many legally blind folks have other disabilities as well (such as cognitive and hearing impairments). Please let us know if there's anything we could do to facilitate this process. Thank you! Appreciatively, Attila Gulyas | IT/Program Assistant Email: agulyas(a)societyfortheblind.org Phone: (916) 889-7510 Access News helpdesk: (916) 889-7519 Access News service: (800) 665-4667 (916) 732-4000 https://societyfortheblind.org/programs-services/access-news/ Society for the Blind 1238 S Street Sacramento, CA 95811 SFTB Main Phone: (916) 452-8271 Fax: (916) 492-2483 www.societyfortheblind.org Our mission is to empower individuals living with low vision or blindness to discover, develop and achieve their full potential.

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TALK event not generated properly by freeswitch
by Shahnawaj Khan 02 Nov '20

02 Nov '20

Hi, I am trying to detect voice using VAD in freeswitch. I have set VAD for both in sip profile. <param name="vad" value="both"> and my dialplan looks like this <extension name="test"> <condition field="destination_number" expression="^(5080)$"> <action application="set" data="rtp_enable_vad_out=true"/> <action application="set" data="rtp_enable_vad_in=true"/> <action application="export" data="fire_talk_events=true"/> <action application="answer" /> <action application="playback" data="https://www2.cs.uic.edu/~i101/SoundFiles/BabyElephantWalk60.wav"/> </condition> </extension> But whenever a music file is played it generates the TALK event for music but not for the user voice. I tried with <extension name="test"> <condition field="destination_number" expression="^(5080)$"> <action application="set" data="rtp_enable_vad_out=true"/> <action application="set" data="rtp_enable_vad_in=true"/> <action application="export" data="fire_talk_events=true"/> <action application="answer" /> <action application="park"/> </condition> </extension> It does not generate TALK events in such a dialplan. Please let me know if I am missing something. Or any other way to detect user voice as well. Thanks & Regards, Shahnawaz

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