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