[Asterisk-Users] RTP session between two end users

[Eric Wieling aka ManxPower]

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