[asterisk-users] Directing SIP/RTP sessions b/w UA

[ast guy]

Hi,
 Let me explain what I'm looking for a solution using asterisk.

I have one third party SIP based server (A) and on Asterisk server (B).
1. Extension-1 --> Server A calls Server B.
2. Server B does some processing and calls/sends back to Server A --->
Extension-2
3. SIP session has been established b/w two Extension-1 and Extension-2.

Now is there any config that I can do in sip.conf which causes direct
sip/rtp communication between Extension-1 and Extension-2 without involving
Server-B

Exten-1-------> |
        |  Server A     | <---->|ServerB |
Exten-2<------- |


-ag
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080206/83b51057/attachment.htm