Steve Davies wrote: > In the cases previously mentioned, the user is doing an attended > transfer using the handset features, and not Asterisk. I do not know > whether SIP even allows the Caller ID to be changed at the point when > two separate calls are bridged to one...
It does, but Asterisk does not currently support that behavior (even in the development branch). I believe Olle's SIP transfer re-write may provide this functionality when Asterisk 1.4 is released, but I am not positive. _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
