[Asterisk-Users] Attended transfer and queue

[Matt]

On 6/12/06, BJ Weschke <bweschke at gmail.com> wrote:
> On 6/12/06, Matt <mhoppes at gmail.com> wrote:
> > AHHH!  We use the Xfer button on our Aastra 9133is to do transfers....
> > for some reason (see another post I just made) when I hit * queue
> > calls disconnect.
> >
> > On 6/12/06, aston martin <astonmartin_2006 at yahoo.com> wrote:
> > >
> > > 1. Call comes in the queue (command Queue(...) gets executed)
> > > 2. Call reaches extension at which agent is registered (I'm using
> > > AgentCallbackLogin), where Dial(SIP/.....,,tT) command gets executed
> > > 3. Agent answers the call and enters *2 (that's my default for attended
> > > transfers as set in features.conf) + a number to which the call should be
> > > transfered
> > > 4. Agent hangs up
> > > 5. Agent stays in 'busy' state
> > >
> > >
>
>  With 1.2.X, the * was hardcoded into chan_agent to drop the call.
> With /trunk and 1.4, this is now a configurable option.

Uhhh who's bright idea was that?