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.
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