[Asterisk-Users] agent with queues remain unavailable during transferred call

[Mario.Spoljar at hypo-alpe-adria.com]

I've had same problem, but I realised that problem was in fact that my
agents used * just for handling incoming queue call and my agent phones
have been registered on another legacy  PBX (Alcatel 4400) interconected
with * through ISDN PRI. Because of that transfer function is handled on
legacy PBX (Alcatel) and Asterisk does not 'know' if agent talks to callee
or if I transfer incoming call. Do you using some other PBX  connected to
Asterisk PBX? That may be the case.

My topology loks like:
                   +----------+              +---------+
--PSTN-PRI-------->|  ALCATEL |<----PRI----->| ASTERISK|
                   +----+-----+              +---------+
                        ^
                        |
                        ¡
                   +----+-----+
                   |  AGENT   |
                   +----------+

This kind of topology were used because:
* agents was used their station on Alcatel before
* through Asterisk I added some additional features to my call center
without need to pay expensive licences to Alcatel
* I need functionality of billing application connected to Alcatel PBX

____________________
Mario Spoljar
mario.spoljar at hypo-alpe-adria.com

asterisk-users-bounces at lists.digium.com wrote on 03/01/2005 15:53:16:

> Hi,
>
> I'm seeing something I'd like suggestions on:
>
> I have a queue with agents that log in using agentcallbacklogin. The
> extension that is logged in with is a Local channel. Now, if a call
> comes in to the queue and is handled by an agent (in our case using
> Cisco 7960 SIP phones) and transferred (attended) to another extension,
> the agent remains unavailable during the remains of the call. Using show
> agents gives this:
>
> 103          (TIC 3) logged in on MGCP/aaln/1 at 00059002798e-1 talking to
> Zap/20-1 (musiconhold is 'default')
>
> As you can see, the Agent is shown with the transferred call, and is
> unavailable for new calls. However, the phone _is_ on hook and free.
>
> I am using a 1.0.2. version (bri-stuff rc2b)
>
> Any suggestions are welcome.
>
> Florian
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users