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
[EMAIL PROTECTED]
[EMAIL PROTECTED] 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/[EMAIL PROTECTED] 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
>
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