[Freeswitch-users] Freeswitch doesnt transcode

[Rajil Saraswat]

The siptrace is at http://pastebin.com/xiGqtj1Y

The call is being made from 303 (Android/CSipsimple with OPUS codec)
to 208 (pjsua test client with PCMU codec). The error is on line 545.

On 14 February 2016 at 11:28, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
> How you originate the call? Is a bridge? From which phone?
>
> Also, please pastebin the complete sip trace (from start of leg A to end of
> both legs) and put here a link to pastebin
>
> Il 14/Feb/2016 03:54, "Rajil Saraswat" <rajil.s at gmail.com> ha scritto:
>>
>> Hello,
>>
>> I have a remote sip phone (Linksys SPA3102) which only supports PCMU.
>> When I call to this remote sip phone i get a 406 error that opus is
>> not supported as shown by the sip trace below. However, if I force the
>> codec to absolute like this
>> {absolute_codec_string='PCMU,PCMA'}sofia/internal/303 at 192.168.1.5
>> the call works fine.
>>
>> Is there anyway I can make FreeSWITCH to automatically transcode
>> without forcing the codec string in the dial plan?
>>
>> The codec preferences is set as  global_codec_prefs=OPUS,PCMU,PCMA and
>> outbound_codec_prefs=PCMU,PCMA,GSM
>>
>> ---------------------------siptrace--------------------------------
>>
>> recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499:
>>
>> ------------------------------------------------------------------------
>>    SIP/2.0 406 Not Acceptable
>>    Via: SIP/2.0/UDP
>>
>> 192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej
>>    Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
>>    From: "202" <sip:202 at 192.168.1.111>;tag=DFX0FUvr2vNcm
>>    To: <sip:303 at 192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
>>    CSeq: 87372504 INVITE
>>    Content-Length:  0
>>
>>
>> ------------------------------------------------------------------------
>> send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591:
>>
>> ------------------------------------------------------------------------
>>    ACK sip:303 at 192.168.1.5 SIP/2.0
>>    Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej
>>    Max-Forwards: 68
>>    From: "202" <sip:202 at 192.168.1.111>;tag=DFX0FUvr2vNcm
>>    To: <sip:303 at 192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
>>    Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
>>    CSeq: 87372504 ACK
>>    Content-Length: 0
>>
>>
>> ------------------------------------------------------------------------
>> 2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel
>> sofia/internal/303 at 192.168.1.5 entering state [terminated][406]
>> 2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup
>> sofia/internal/303 at 192.168.1.5 [CS_CONSUME_MEDIA]
>> [SERVICE_NOT_IMPLEMENTED]
>>
>> Thanks
>> Rajil
>>
>> _________________________________________________________________________
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>>
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>
>
> _________________________________________________________________________
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> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
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