[asterisk-users] how to transfer a caller out of a queue ?

Stefan-Michael. Guenther (in-put GbR) asterisk at in-put.de
Mon Sep 18 13:08:04 MST 2006

Hi Rick,

Am Montag, 18. September 2006 21:30 schrieb Rick Smith:
> can't the agent just transfer the caller to another extension, whether that
> be another queue or a person ?
yes, that's the easy part. But my client wants the caller (!) to be able to 
transfer himself into another context. The reason for this is, that the 
caller must decide whether he wants to be transfered from a free support line 
to a support line for which he would have to pay.


> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
> Stefan-Michael. Guenther (in-put GbR) Sent: Monday, September 18, 2006 3:19
> PM
> To: asterisk-users at lists.digium.com
> Subject: [asterisk-users] how to transfer a caller out of a queue ?
> Hi,
> I would like to give a caller the chance to leave a queue after an agent
> has already accepted the call.
> The caller enters the queue by dialing 333:
> [from-sip]
> exten => 300,1,Answer()
> exten => 300,2,Queue(q1|tT)
> When the caller presses # and e.g. 1, asterisk is looking for this
> extension in the context where the call came in. In my configuration this
> means, that my office phone is ringing:
> exten => 1,1,Answer()
> exten => 1,2,DIAL(CAPI/@8304499:8304498,30,tTr)
> exten => 1,3,Hangup
> But in this case not the caller, but the agent has been transferred!
> Isn't there a chance for the caller to stop the conversation e.g. because
> the agent told him that he has called the wrong queue and that he should
> dial #1 to get to the right queue or directly to another person?
> If the agent does this, the caller get's transfered to the office phone, as
> expected.
> As far as I understand the documentation, the context that is assigned to a
> queue in queue.conf is only valid before an agent has accepted the call.
> I'm still running Asterisk 1.0.6, which is the current version for SuSE
> 9.3. Maybe Asterisk 1.2.x would help?
> Thanks for help & hints,
> Stefan


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