Re: Repost: Re: [asterisk-dev] How does RFC2833 get indicated to the SIP peer
On 2/17/06, Ed Greenberg [EMAIL PROTECTED] wrote: Can somebody who understands chan_sip.c please explain this to me? THanks. --On Thursday, February 16, 2006 6:20 AM -0800 Ed Greenberg [EMAIL PROTECTED] wrote: 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. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --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
Re: Repost: Re: [asterisk-dev] How does RFC2833 get indicated to the SIP peer
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
Repost: Re: [asterisk-dev] How does RFC2833 get indicated to the SIP peer
Can somebody who understands chan_sip.c please explain this to me? THanks. --On Thursday, February 16, 2006 6:20 AM -0800 Ed Greenberg [EMAIL PROTECTED] wrote: 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? Thanks, /edg ___ --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 ___ --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
