[Asterisk-Users] SIP Redirect/Transfer

[Tony Mountifield]

I have a question which may be about the SIP protocol, or may be about
SIP features supported in Asterisk, I don't know.

Let's say I have three Asterisk boxes, A, B and C, which pass calls to
each other using SIP.

A call comes into box A from somewhere, and A determines that the call
should be routed to box B.

When box B receives the call, it does some operations internally, and
decides that in fact the call should be handled by box C instead.

I know B could easily dial a new call to C and pass the contents of
the call back and forth between A and C.

However, is it possible for box B to redirect the original call to
box C so that A is talking directly to C, and B is no longer involved?

In fact, A and C might not be Asterisk, but other kinds of SIP switch.
Box B definitely is Asterisk, and is the box over which I have control.

Thanks in advance for any ideas.

Cheers
Tony
-- 
Tony Mountifield
Work: tony at softins.co.uk - http://www.softins.co.uk
Play: tony at mountifield.org - http://tony.mountifield.org