[Freeswitch-users] Try to get rid of Speex codec on invite

[Peter Steinbach]

Thanks Brian,

that did the trick.

Best regards
Peter


Brian West schrieb:
> You are making the wrong conclusion.  The SDP is being verbose.  That SDP is offering PCMA, PCMU, Speex on 98, GSM, DTMF on 101 and CN
>
> What you need to set is the variable verbose_sdp=true so that we do a complete filled out SDP because the device you're speaking to is being intellectually challenged.  See the codec list in the m= line?
>
> /b
>
> On Jan 11, 2011, at 4:35 PM, Peter Steinbach wrote:
>
>   
>> Hello,
>>
>> when our Freeswitch sends an INVITE to a phone, it only offers speex
>> codec for voice:
>> m=audio 12068 RTP/SAVP 8 0 98 3 18 101 13
>> a=rtpmap:98 SPEEX/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>>     
>
>
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>   


-- 
With kind regards
Peter Steinbach 

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