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
[email protected]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
