asterisk users - Jul 2004 - 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

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" <chad@idworld.net>
To: <asterisk-users@lists.digium.com>
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
>
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