I just read that workaround as you posted. Thanks =)

Gary

On 8/28/06, Stephan Monette <[EMAIL PROTECTED]> wrote:
Now I remember.

I had to programm all our agents in a static queue instead of having the
agents to login. We fix the problem by having a static agent queue.

Stephan Monette
Unlimitel inc.
Tel.: 1 (877) 464-6638, x221
Fax.: (613) 443-7773


-----Original Message-----
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gary T. Giesen
Sent: Monday, August 28, 2006 5:26 PM
To: [email protected]
Subject: [on-asterisk] Asterisk gets hosed on call transfer out of queue.


Done that. I just found that it's a documented bug.
http://bugs.digium.com/view.php?id=7458 . Which is a bad thing, since it
doesn't mean it's anything I can fix at the moment.

Gary


On 8/28/06, Stephan Monette <[EMAIL PROTECTED]> wrote:
> Try with the canreinvite=no in the sip.conf file. Maybe the SIP
> devices of the agent are not SIP reinvite compatible.
>
> Stephan Monette
> Unlimitel inc.
> Tel.: 1 (877) 464-6638, x221
> Fax.: (613) 443-7773
>
>
> -----Original Message-----
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gary T.
> Giesen
> Sent: Monday, August 28, 2006 4:59 PM
> To: [email protected]
> Subject: [on-asterisk] Asterisk gets hosed on call transfer out of queue.
>
>
> I have agents in a queue, and when they receive a call, answer it, and
> then transfer it to someone else (either internal or external), the
> call is dropped and asterisk gets hosed, to the point where the queues
> dont work anymore, I can't get a listing of logged on agents or sip
> channels.
>
> The last thing I see when the call is transferred is:
>
> SIP/xxxxxxxxxx-9a13 is making progress passing it to
> Local/[EMAIL PROTECTED],2
>
> Can anyone give my ideas at what I should look at?
>
> Thanks,
>
> Gary
>
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