[Asterisk-Users] RTP session between two end users

[Erdem HAKİ]

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] 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
***************