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astguy at gmail.com Guest
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Posted: Wed Feb 06, 2008 6:05 am Post subject: [asterisk-users] Directing SIP/RTP sessions b/w UA |
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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
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