You just described an supervised transfer. You need to be pressing # then dialing bob and hanging up.
I suspect your sip devices don't support supervised transfers properly. bkw ----- Original Message ----- From: "Chad Scott" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, July 01, 2004 6:36 PM Subject: [Asterisk-Users] Hangup on transfer... > I've got another issue I can't quite figure out and a search of the > archives and Google turn up nothing... > > Say a call comes in (these are all via SIP) and is sent into a > Queue(myqueue,t,,,300). Note the "t" to allow whomever receives the > call to transfer it. > > The call is enqueued, and the logged in agents ring as expected. An > agent answers the call, and begins chatting. He says, "oh, you need to > speak to Bob, let me transfer you." He hits 'transfer,' dials Bob's > extension, waits for an answer, briefs Bob, then hits 'transfer' again. > > Asterisk then promptly drops all parties. > > This is such a common use case I'm sure I'm missing something incredibly > simple, but I just can't spot it. > > Bob's extension is defined as Dial(SIP/1234,20,t). > > It looks, to me anyway, like it should work. Should I file a bug or can > someone hit me over the head and show me the way it should be > configured? > > Thanks in advance, > Chad > > _______________________________________________ > 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 > _______________________________________________ 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
