Could you please share your Asterisk files configuration? I'm running on problems with audio when calling from/to a webrtc script (jsSIP) and Asterisk 12. I'm using an extension attached to a SIP trunk and my calls do fine but there is no audio on any side. -- Enviado desde la aplicación myMail para Android miércoles, 09 septiembre 2015, 04:27p.m. -05:00 de Carlos Chavez < [email protected]> :
>On 9/9/15 4:22 PM, D'Arcy J.M. Cain wrote: >> On Wed, 9 Sep 2015 16:11:03 -0500 >> Carlos Chavez < [email protected] > wrote: >>> I am having a small problem that is driving me nuts. I can make >>> calls over my Webrtc client without any problems and audio sounds >>> fine. The only problem I have is that when I call an internal SIP >>> extension on my PBX I do not hear the ring while I wait for the call >>> to be answered. My dial command does include the rR options. If I >>> make an external call to a land line or a mobile phone I do hear the >>> ring sounds, only internal extensions have this problem. Why would >>> the webrtc client ignore the ringing when calling another SIP >>> extension? Any ideas? >> I had a similar problem. Turned out that my indications.conf file was >> empty. Check that out. >> >The file is full of definitions for many countries. It specifically has >one for Mexico but we usually use the same one as the USA. > >-- >Telecomunicaciones Abiertas de México S.A. de C.V. >Carlos Chávez >+52 (55)9116-91161 > > >-- >_____________________________________________________________________ >-- Bandwidth and Colocation Provided by http://www.api-digital.com -- >New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
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