Olivier
2009-Feb-09 21:43 UTC
[asterisk-users] Is "a=fmtp:101 0-15" a legal option in SDP ?
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 ? regards * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090209/76c3070c/attachment.htm
Raj Jain
2009-Feb-09 22:17 UTC
[asterisk-users] Is "a=fmtp:101 0-15" a legal option in SDP ?
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. -- Raj Jain