-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling aka ManxPower Sent: Monday, June 27, 2005 8:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] RTP session between two end users
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.) *************** We use same codec and don't use meetme etc... So what else should i do? Thanks Erdem HAKI *************** _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
