9 feb 2009 kl. 23.17 skrev Raj Jain: > On Mon, Feb 9, 2009 at 4:43 PM, Olivier <[email protected]> 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 _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
