[asterisk-users] No ring sound when calling SIP extensions over Webrtc

[Alfonso Sandoval]

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 < cursor at telecomabmex.com> :

>On 9/9/15 4:22 PM, D'Arcy J.M. Cain wrote:
>> On Wed, 9 Sep 2015 16:11:03 -0500
>> Carlos Chavez < cursor at telecomabmex.com > 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
>
>
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