[Asterisk-Dev] Queue - Transfer & server stability
Dov Bigio
dovb at terra.com.br
Mon Nov 21 11:43:57 MST 2005
Actually I am using 1.0.9... Anyway, I'll reproduce the error and check all
logs & debugs.
As I am waiting to receive Asterisk Business Edition in the next few days,
I'll try to reproduce the problem there and contact Digium's support in case
I really need it...
Thank you,
Dov
----- Original Message -----
From: "Olle E. Johansson" <oej at edvina.net>
To: "Dov Bigio" <dovb at terra.com.br>; "Asterisk Developers Mailing List"
<asterisk-dev at lists.digium.com>
Sent: Monday, November 21, 2005 2:57 PM
Subject: Re: [Asterisk-Dev] Queue - Transfer & server stability
> Dov Bigio wrote:
> >
> > Hello,
> >
> > When I use the Queue application, I have taught all agents to make
> > transfers using the # key, and use the parking feature to make attended
> > transfers.
> >
> > But, sometimes, they make mistakes and use the SoftPhone's (EyeBeam)
> > XFer button.
> >
> > Everytime this happens, after a few seconds, the Queue application stops
> > answering, and I have to restart Asterisk.
> >
> > Is this a bug? Shuldn't Asterisk just ignore the SIP transfer instead of
> > locking up?
> >
> > In this situation, the queue_log file displays a ZOMBIE channel...
> >
> >
1132584647|1132584552.19305|cobranca|SIP/ivan.ferreira-d831<ZOMBIE>|COMPLETE
AGENT|12|60
> > 1132584779|1132584111.19200|operacoes|Agent/4917|COMPLETECALLER|9|646
> > 1132584804|1132584781.19328|cobranca|NONE|ENTERQUEUE||"Cobranca 202"
<202>
> > 1132584828|1132584781.19328|cobranca|Agent/5450|CONNECT|24
> > 1132584836|1132584813.19338|cobranca|NONE|ENTERQUEUE||"Cobranca 339"
<339>
> > 1132584997|1132584235.19222|cobranca|Agent/5110|COMPLETECALLER|212|527
> > 1132585329|1132584781.19328|cobranca|Agent/5450|COMPLETECALLER|24|501
> > 1132586395|NONE|NONE|NONE|QUEUESTART|
> > 1132586396|NONE|NONE|NONE|CONFIGRELOAD|
> > Thank you for your help,
>
> In order to fully understand what happens, I need to see a full debug of
> a transaction - set verbose 4, set debug 4 and turn on sip debug.
> Capture all output and we'll analyze.
>
> Also, always tell us which version of Asterisk you are using and on
> which platform. As you are mailing to asterisk-dev, I suppose you are
> using CVS head, the development code, but I do not know the platform.
> THanks.
>
> /O
>
>
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