> > > Back in Asterisk 1.0.5, when we sent our SDP packet to the distant end, > > > we sent > > > m=audio <port> RTP/AVP <codec> 101 > > > where the 101 which indicated that we wanted to get RFC2833 DTMF from our > > > other end. > > > > > > Now it's missing, and my peer (level3) is sending me inband DTMF. > > > > > > It's not obvious to me from reading channels/chan_sip.c (in either the > > > old 1.0.5 or the current 1.2.4) how this 101 gets on the end of my Media > > > Description line or how else the peer is supposed to know that I need > > > rfc2833 DTMF. > > > > > > Can somebody please explain? > > Do you have dtmfmode=rfc2833 in sip.conf for this peer? If so, let's > get a sip debug and open a bug on bugs.digium.com.
Might also do another update as that was removed by Olle about a week ago, and then restored a few hours later. _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
