<p dir="ltr">Look at the invite starting at 480. You are only offering opus to the callee. </p>
<div class="gmail_quote">On Feb 14, 2016 2:05 PM, "Rajil Saraswat" <<a href="mailto:rajil.s@gmail.com">rajil.s@gmail.com</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">The siptrace is at <a href="http://pastebin.com/xiGqtj1Y" rel="noreferrer" target="_blank">http://pastebin.com/xiGqtj1Y</a><br>
<br>
The call is being made from 303 (Android/CSipsimple with OPUS codec)<br>
to 208 (pjsua test client with PCMU codec). The error is on line 545.<br>
<br>
On 14 February 2016 at 11:28, Giovanni Maruzzelli <<a href="mailto:gmaruzz@gmail.com">gmaruzz@gmail.com</a>> wrote:<br>
> How you originate the call? Is a bridge? From which phone?<br>
><br>
> Also, please pastebin the complete sip trace (from start of leg A to end of<br>
> both legs) and put here a link to pastebin<br>
><br>
> Il 14/Feb/2016 03:54, "Rajil Saraswat" <<a href="mailto:rajil.s@gmail.com">rajil.s@gmail.com</a>> ha scritto:<br>
>><br>
>> Hello,<br>
>><br>
>> I have a remote sip phone (Linksys SPA3102) which only supports PCMU.<br>
>> When I call to this remote sip phone i get a 406 error that opus is<br>
>> not supported as shown by the sip trace below. However, if I force the<br>
>> codec to absolute like this<br>
>> {absolute_codec_string='PCMU,PCMA'}sofia/internal/<a href="mailto:303@192.168.1.5">303@192.168.1.5</a><br>
>> the call works fine.<br>
>><br>
>> Is there anyway I can make FreeSWITCH to automatically transcode<br>
>> without forcing the codec string in the dial plan?<br>
>><br>
>> The codec preferences is set as global_codec_prefs=OPUS,PCMU,PCMA and<br>
>> outbound_codec_prefs=PCMU,PCMA,GSM<br>
>><br>
>> ---------------------------siptrace--------------------------------<br>
>><br>
>> recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499:<br>
>><br>
>> ------------------------------------------------------------------------<br>
>> SIP/2.0 406 Not Acceptable<br>
>> Via: SIP/2.0/UDP<br>
>><br>
>> 192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej<br>
>> Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124<br>
>> From: "202" <<a href="mailto:sip%3A202@192.168.1.111">sip:202@192.168.1.111</a>>;tag=DFX0FUvr2vNcm<br>
>> To: <<a href="mailto:sip%3A303@192.168.1.5">sip:303@192.168.1.5</a>>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q<br>
>> CSeq: 87372504 INVITE<br>
>> Content-Length: 0<br>
>><br>
>><br>
>> ------------------------------------------------------------------------<br>
>> send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591:<br>
>><br>
>> ------------------------------------------------------------------------<br>
>> ACK <a href="mailto:sip%3A303@192.168.1.5">sip:303@192.168.1.5</a> SIP/2.0<br>
>> Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej<br>
>> Max-Forwards: 68<br>
>> From: "202" <<a href="mailto:sip%3A202@192.168.1.111">sip:202@192.168.1.111</a>>;tag=DFX0FUvr2vNcm<br>
>> To: <<a href="mailto:sip%3A303@192.168.1.5">sip:303@192.168.1.5</a>>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q<br>
>> Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124<br>
>> CSeq: 87372504 ACK<br>
>> Content-Length: 0<br>
>><br>
>><br>
>> ------------------------------------------------------------------------<br>
>> 2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel<br>
>> sofia/internal/<a href="mailto:303@192.168.1.5">303@192.168.1.5</a> entering state [terminated][406]<br>
>> 2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup<br>
>> sofia/internal/<a href="mailto:303@192.168.1.5">303@192.168.1.5</a> [CS_CONSUME_MEDIA]<br>
>> [SERVICE_NOT_IMPLEMENTED]<br>
>><br>
>> Thanks<br>
>> Rajil<br>
>><br>
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><br>
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</blockquote></div>