>> > So, does anyone know of a way to detect whether a call from a SIP phone >> > is the first step of an attended transfer or an original call? > > It could probably work if you put a SIP proxy in between (ref. Kamilio).
Another way might be to set up a special transfer extension that all users use to perform transfers. To do a transfer, all users would first transfer to that special transfer extension. The transfer extension could then read the intended destination and compare the source and destination in a series of GotoIf statements. The GotoIf statements would check the source and destination of the transfer, and if it's ok, use the transfer() app. If not, playback a message that the transfer is not allowed. It means a lot of very specific dialplan logic, and a change of procedures for the users, but it's one way to do it. - Noah _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
