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
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.
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