[asterisk-users] Is "a=fmtp:101 0-15" a legal option in SDP ?

[Johansson Olle E]

9 feb 2009 kl. 23.17 skrev Raj Jain:

> On Mon, Feb 9, 2009 at 4:43 PM, Olivier <oza-4h07 at myamail.com> wrote:
>>
>> Hi,
>>
>> My patton 4638 is sending :
>> v=0
>> o=MxSIP 0 46 IN IP4 192.168.100.52
>> s=SIP Call
>> c=IN IP4 192.168.100.52
>> t=0 0
>> m=audio 4984 RTP/AVP 8 101
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15
>> a=sendrecv
>> m=image 4986 udptl t38
>> a=T38FaxUdpEC:t38UDPRedundancy
>> a=sendrecv
>>
>>
>> Asterisk (1.4.22.1) replies :
>> Got unsupported a:fmtp in SDP offer
>>
>> Shall I care ?
>
> This error is somewhat benign. Basically, the end-point is telling
> that it can receive RFC 2833 events in the range of 0-15 (DTMF tones)
> and Asterisk is ignoring that range.

To clarify: We're not ignoring DTMF. We're just not parsing the fmtp  
header.
This is a message while doing debug, so don't bother with it.

/O