> Well you could do a supervised transfer, or 3 way call. Basically, you > place the one leg of the call on hold, dial the extension you are to > transfer to, then if you successfully connect, bring the call to three > way, and then excuse yourself.
Ok, I understand that one. Overkill but it would work. I know how to do this on a Zap interface (hookflashes) , but how does one do it on a SIP or IAX interface? > Or you could create a kind of Macro for transfers where it stores the > originating part of the transfer, and upon failed connect, does a return > dial. This would bypass normal call routing where a direct call would go > to voicemail if it misses a person at the end. I thought of that too, but I don't think it'll work: exten 101,1,Setvar(myexten) exten 101,2,Dial(Zap/1,10,t) exten 101,103,getvar(myexten) exten 101,104,Dial(myexten) If extension 101 is busy it will immediately try to dial me, but I'm still on the phone since the transfer didn't complete... Or did I misunderstand "return-dial" ? -A. _______________________________________________ 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
