November 2020 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Replace stun.freeswitch.org in default configuration since it's gone.
by mario_fs 10 Nov '20

10 Nov '20

For the last couple of weeks, I noticed stun.freeswitch.org <http://stun.freeswitch.org/> is no longer responding. I do verification weekly tests of freeswitch development and this is a big problem since the tests are mostly automated. Also, this is not a good thing for new users who follow the wiki to setup their first freeswitch. I am not sure if this should be a problem report but does anyone else think if stun.freeswitch.org <http://stun.freeswitch.org/> no longer responds shouldn’t the default config be changed? Mario G

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Is there any way to break a bridged call without hangup caller?
by 王聡 10 Nov '20

10 Nov '20

Hello all, We’re trying to make a call system with pairing function based on freeswitch, the function we designed likes below: 1.Create a call to end user and hold (standby). [Use endless_playback] 2.Bridge 2 standby users and let them talk with each other. [Use uuid_bridge] 3.Stop bridge and let users back to standby status for next pair. [Under design] We had test function 1 and 2 successfully, but no idea for function 3. Hangup app would hangup both immediately, and break app seems not working. For some reason it is not easy for us to switch to conference, so I wonder if there are any advice for this function based on uuid_bridge. Best regards.

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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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Re: [Freeswitch-users] SIP Chronicles, recap and upcoming episode on Friday @ 7pm UTC
by Maxim Sobolev 05 Nov '20

05 Nov '20

OOPS, typo, the link to Sandro's episode is below: https://youtu.be/6EKVDyAx2Yo -Max On Thu, Nov 5, 2020 at 11:01 AM Maxim Sobolev <sobomax(a)sippysoft.com> wrote: > 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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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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