Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

VoIP Mailing List Archives
Mailing list archives for the VoIP community
 SearchSearch 

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


 
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users
View previous topic :: View next topic  
Author Message
astguy at gmail.com
Guest





PostPosted: Wed Feb 06, 2008 6:05 am    Post subject: [asterisk-users] Directing SIP/RTP sessions b/w UA Reply with quote

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
Back to top
Display posts from previous:   
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users All times are GMT - 5 Hours
Page 1 of 1

 
Jump to:  
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum


Powered by phpBB © 2001, 2005 phpBB Group

VoiceMeUp - Corporate & Wholesale VoIP Services