[Asterisk-Users] Hangup on transfer...

[Chad Scott]

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