Erdem HAKİ wrote:
Is it possible that a RTP session between two end users (so i want to use
asterisk as a signaling proxy and bypass RTP sessions)?
I used "canreinvite=yes" but it didn't work.
Description from asterisk conf. File;
(canreinvite=yes ; allow RTP voice traffic to bypass
Asterisk)
It's sip.conf. reinvites only work if the codec is the same for the
two endpoints and Asterisk does NOT have to listen for DTMF (no t or T
on the dial line, no meetme, etc.)
--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
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