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Hello Group, I have a requirement to dialout some external number, once the call is answered the same has to be forwarded to an Internal Queue. Please help me. I have tried calling with two SIP end point forwarding , even that is not working, My dial plan line is , Dial(SIP/19201/19202,300) -- *Best regards,* *Ruban.S* -------------- next part -------------- An HTML attachment

Hi, Thanks for your info, What is the impact of the following line in dialplan, Dial(SIP/19201/19202,300) On Thu, Sep 3, 2015 at 7:20 PM, Vinicius Fontes <vinicius at aittelecom.com.br> wrote: > You might want to use the Originate() application instead. Check its usage > by issuing the command 'core show application originate' on Asterisk CLI. > > 2015-09-03 9:09 GMT-03:00 Kantharuban Ruban <kanth.ruban at gmail.com>: > >> Hello Group...

I have several large customers (200+ extensions) running on vSphere without issue. Not sure about OpenVZ, thought. 2015-04-07 11:36 GMT-03:00 Mitul Limbani <mitul at enterux.in>: > Show him this freaking thread, or else ask him to prove it otherwise. > > We all here have decades of exp dealing with asterisk. > > Mitul > On 07-Apr-2015 7:27 PM, "Ikka

...erence in my case is Asterisk server has a public IPv4 and the browser is behind a single NAT. I'm forwarding my configuration below (which I posted previously on asterisk-users). How can we debug ICE negotiation? ---------- Forwarded message ---------- From: Vinicius Fontes <vinicius at aittelecom.com.br> Date: 2015-07-27 13:54 GMT-03:00 Subject: No audio on SIP over WebRTC To: Asterisk Users Mailing List - Non-Commercial Discussion < asterisk-users at lists.digium.com> I'm following this tutorial ( https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5) to de...

hello, i'm facing strange problem asterisk13.5 + chan_sip wss transport + SIPML5 1.5.230 person1 to person3 are behind different NATs audio devices double checked call from person1(chrome) to person2(chrome) works call from person1(chrome) to person 3(chrome) - no audio on both side (RTP flowing only in one direction) call from person2(chrome) to person 3(chrome) - no audio on both side

I'm having an issue with CDR. Basically, I expect to have all "legs" of a call having the same linkedid and differing only by the sequence value. That does happen, but I'm getting null dst values after doing an attended transfer. I'm not sure if this is a bug or I'm doing something wrong. I'm running Asterisk 13.2.0. Here's the console log, step by step: First,

Have you tried Asterisk 13? The bridging mechanism has been completely rewritten on Asterisk 12, so there's no longer channel masquerading and zombie channels. Might be worth a try. 2015-04-07 20:33 GMT-03:00 Alex Villac??s Lasso <a_villacis at palosanto.com>: > El 07/04/15 a las 17:38, Alex Villac??s Lasso escribi?: > > I am trying to collect enough information about an