Rohan Hathiwala wrote:
Hi,
I need asterisk to instruct the other side to send RTP to a conference
server running on a different machine. The conference server does not
understand SIP so I cannot use the SIP REFER method.
I have another question. Suppose when processing a SIP INVITE we want to use
asterisk only for call control and let another server handle the RTP is
there a clean way to do this in asterisk.
Regards,
Rohan Hathiwala.
Asterisk/chan_sip wasn't designed to be able to do this. You're going to
end up modifying things... potentially a lot. If the conference server
does SIP though you can just dial it, make sure canreinvite is set to
yes, and audio should go direct.
Joshua Colp
Software Developer
Digium, Inc.
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