<p dir="ltr">How you originate the call? Is a bridge? From which phone?</p>
<p dir="ltr">Also, please pastebin the complete sip trace (from start of leg A to end of both legs) and put here a link to pastebin<br>
</p>
<div class="gmail_quote">Il 14/Feb/2016 03:54, "Rajil Saraswat" <<a href="mailto:rajil.s@gmail.com">rajil.s@gmail.com</a>> ha scritto:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">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>
SIP/2.0 406 Not Acceptable<br>
Via: SIP/2.0/UDP<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>
send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591:<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>
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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</blockquote></div>