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 <[EMAIL PROTECTED]> 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: [EMAIL PROTECTED] - http://www.softins.co.uk > Play: [EMAIL PROTECTED] - http://tony.mountifield.org > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
