-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling aka ManxPower Sent: Tuesday, June 28, 2005 6:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] RTP session between two end users
Erdem HAKİ wrote: > > -----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? > >How are you determining if RTP audio is going thru Asterisk? >Remember, SIP signaling will always go thru Asterisk. >Also do a "sip show channels" during a call to confirm that the codecs >are the same. -- >Always do right. This will gratify some people and astonish the rest. Mark Twain _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ************************ Hi, I determine signaling with ethereal and i am sure that both sides use the same codec. By the way, i searched forum again and i read something below; > In wiki pages it is stated that The audio channels (RTP) may go directly > from phone to phone or may go through Asterisk's media bridge. > > Currently with my settings, I notice that all rtps are passing through > my asterisk. How could I achieve that they go directly from phone to > phone? I assume this way, my machine will have less load and therefore > could handle more calls. As bkw pointed out, use canreinvite=yes for each sip phone definition. But, that will only work if the phones can reach each other directly (the phones and/or asterisk can't be behind a nat/firewall box). Thanks Erdem HAKI [EMAIL PROTECTED] _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users