[Asterisk-Users] SIP Redirect/Transfer

[BJ Weschke]

 Olle has said he has a working patch for this scenario, but it will
be a couple of weeks yet before it's ready to be merged into the HEAD
tree so it will be a post 1.2 thing.

On 11/10/05, Tony Mountifield <tony at softins.clara.co.uk> wrote:
> 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
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