[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 * ___ -- 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
Re: [asterisk-users] Is "a=fmtp:101 0-15" a legal option in SDP ?
On Mon, Feb 9, 2009 at 4:43 PM, Olivier 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 ___ -- 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
Re: [asterisk-users] Is "a=fmtp:101 0-15" a legal option in SDP ?
9 feb 2009 kl. 23.17 skrev Raj Jain: > On Mon, Feb 9, 2009 at 4:43 PM, Olivier 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
