<HTML><BODY><p style='margin-top: 0px;' dir="ltr">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. </p>
<p dir="ltr">I'm using an extension attached to a SIP trunk and my calls do fine but there is no audio on any side. </p>
<p dir="ltr">--<br>
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miércoles, 09 septiembre 2015, 04:27p.m. -05:00 de Carlos Chavez <<a href="mailto:cursor@telecomabmex.com">cursor@telecomabmex.com</a>>:<br><br><blockquote style='border-left:1px solid #FC2C38; margin:0px 0px 0px 10px; padding:0px 0px 0px 10px;' cite="14418340560000000460">
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<div id="style_14418340560000000460_BODY">On 9/9/15 4:22 PM, D'Arcy J.M. Cain wrote:<br>
<div class="mail-quote-collapse">> On Wed, 9 Sep 2015 16:11:03 -0500<br>
> Carlos Chavez <<a href="/compose?To=cursor@telecomabmex.com">cursor@telecomabmex.com</a>> wrote:<br>
>> I am having a small problem that is driving me nuts. I can make<br>
>> calls over my Webrtc client without any problems and audio sounds<br>
>> fine. The only problem I have is that when I call an internal SIP<br>
>> extension on my PBX I do not hear the ring while I wait for the call<br>
>> to be answered. My dial command does include the rR options. If I<br>
>> make an external call to a land line or a mobile phone I do hear the<br>
>> ring sounds, only internal extensions have this problem. Why would<br>
>> the webrtc client ignore the ringing when calling another SIP<br>
>> extension? Any ideas?<br>
> I had a similar problem. Turned out that my indications.conf file was<br>
> empty. Check that out.<br>
><br>
</div>The file is full of definitions for many countries. It specifically has <br>
one for Mexico but we usually use the same one as the USA.<br>
<br>
-- <br>
Telecomunicaciones Abiertas de México S.A. de C.V.<br>
Carlos Chávez<br>
+52 (55)9116-91161<br>
<br>
<br>
-- <br>
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