May 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Goggle QUIC Protocol
by Abdul Hakeem 15 May '16

15 May '16

Hello, Has anyone ever tried running FS using QUIC rather than UDP/TCP/SCTP ?. Cheers, Abdul Hakeem

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T.38 Problem
by Stefan Kainz 15 May '16

15 May '16

Hello, i've a quick question. Are there any known issues when faxing from asterisk to freeswitch using T.38? FS Version 1.6.8 git / Asterisk 13.8.2. We're trying very hard getting this to work, but it just wont. Everything else fax-related seems to be working very well, but in that case we keep getting result (3) Timed out waiting for the first message. */When using Version FS 1.2 this problem doesnt seem to exist./* The setup is as following: ATA ( cisco spa ) -> Asterisk -> Freeswitch1.6.8 ( rxfax ) regards,

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Using freeswitch as Generic SIP SBC/RtpProxy without predefined endpoints
by Taisto Qvist 14 May '16

14 May '16

Hi, I am an OpenSIPS/Kamailio user that has spent the whole day trying to figure out which software is most suitable to build an SBC that only acts as a full b2bua with rtp-relay, with an undefined number of destinations(entire domains with lots of unknown proxies) on each side. I've been looking at yate, freeswitch, SEMS, but I am overwhelmed by the gazillions of configuration-files that seem to be needed. I have absolutely NO need of pbx/user functionality, just a B2BUA and rtp-proxy capabilities(with full codec-transparency), with the addition that I must be able to add some custom logic to decide the DNS or IP(v6) address of the next-hop sip-node. All examples I've found on freeswitch SBC config, has predefined gateways/trunks/proxies as "predefined next-hops", and I want the SBC to be able to send anywhere, depending on custom application logic. I'd stick with kamailio/rtp-proxy and the standard topology-hiding module, but unfortunately I need the full topology hiding of a b2bua. Is this easily achievable with Freeswitch? Any help with be immensely appreciated! Regards Taisto Qvist

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Verto - ringing
by Gregor Nanger 13 May '16

13 May '16

Hi! I need some advice. I am using verto and created client by myself from base of demo. Now I have one problem and cannot solve it. If user is in call and he gets another call in background (call waiting). Audio from active call is lost. Call is active, but cannot hear audio. Is this something that I overlooked in javascript? Best regards, Gregor

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Debian, FreeSWITCH, and email
by Steven Schoch 13 May '16

13 May '16

I'm setting up a new system to replace our previous one (which was FreeSWITCH 1.4.15 on CentOS 6.6) to a better supported configuration (latest FreeSWITCH on Debian jessie). I'm getting to the part where I'm configuring email, to be used by FreeSWITCH voicemail notifications, the LUA freeswitch.email function, and future FAX support (assuming I can get it to receive FAXes dependably so I can get rid of my POTS FAX line). I found that exim4 comes with Debian, but I also found a (old) warning in https://freeswitch.org/confluence/display/FREESWITCH/mod_voicemail talking about potential problems, that may have been fixed. So I was wondering if I should use exim4, or install postfix. Any opinions? -- Steve

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Codec transcoding verto - provider
by Gregor Nanger 13 May '16

13 May '16

I would please you for opinion. If verto client use OPUS to freeswitch and then call is bridged to provider that use ulaw, than transcode happens. Regarding performance, what do you think, should we also force verto client to use ulaw, so there is no transcoding? But I think OPUS is more reliable for transportation and quality? Best regards, Gregor

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Checking authorisation on external profile
by Artur Mega 13 May '16

13 May '16

Good day, How can I check incoming call to the External profile? Theoretically, both authorised and unauthorised users can call via external profile. Tried to detect authorised users with ${sip_authorised} variable, but it doesn't help. :( -- Arthur

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Javascript in freeswitch
by Nduwayezu, Joselyne 13 May '16

13 May '16

Hello, How do you write scripts in javascript language? I tried this example: #!/usr/bin/js session = new Session("sofia/external/xxxxxxx(a)yyyyyyy.fr"); session.waitForanswer(10000); if(session.ready()){ NDUWAYEZU Joselyne -- Haute École de Bruxelles

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Support 2000 concurrent calls on single server
by Abhishake Agarwal 13 May '16

13 May '16

Hi, I want to support 2000 concurrent calls for my client. Can we support by using single centos server. What should be server config (Dedicated) & what optimization have to be done. Our server will be used as proxy and may required transcoding for some calls. Thanks, Abhishake

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Disable VAD for Conference??
by Alexander Haugg 13 May '16

13 May '16

Hi, At the moment i can't disable VAD. I Try <action application="set" data="rtp_disable_vad_in=true" /> <action application="set" data="rtp_disable_vad_out=true" /> Before the conference will entering. I set the vad = none in the sofia profile I set the comfort-noise in the conference config = 0 But the VAD in the Conference is working again. Is there another idea? Thanks a lot

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