8 maj 2007 kl. 15.40 skrev Joshua Colp:
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.
...psst...
There's a patch in the bug tracker that I believe is what you want.
Please test and review, add your comments.
/O
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