Hello ppl,
Any way to do a re-invite and make RTP bypass Asterisk, after call
establishment.
In other words, I would like to control when to do the bypass work for
peer-peer RTP flow.
The issue is that I need to send DTMFs after dialing the user because most of
the users are behind PBXes (having individual extensions) themselves and almost
all of the PBXes send a 200 OK and then play out the PBX messages.
So I need to send the extension DTMFs first, bridge the calls and then
re-invite users for them to do a peer-peer rtp conversation.
TiA,
- Ben.
____________________________________________________________________________________
Be a better friend, newshound, and
know-it-all with Yahoo! Mobile. Try it now.
http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
