[asterisk-users] Agent channel...

[Carlos Chavez]

	I have been looking for the busy-limit directive you mention but cannot
find it in any documentation for Asterisk.  I can only find something
called "busy-level" which by its description might be what I need.

On Wed, 2008-07-16 at 15:20 +0000, Tariq .. wrote:
> Try adding "busy-limit=1" to your SIP users as it will let the agent
> to report the "Busy" as a hint.
> the "call-limit=1" only allows one channel to the agent.. but then if
> the agent is not "busy" the queue will try to call them and it will
> bypass the CW service so the Agent channel will receive the call and
> drop it immediately.
> adding the busy-limit=1 will send the "busy here" hint to the queue
> when it tries to call it .. and then the queue will try another
> agent. 
> Salam
> Tarek Sawah
> 
> 
>                                    
> 
> 
> 
> 
> 
> ______________________________________________________________________
> > Date: Tue, 15 Jul 2008 10:54:34 +1000
> > From: pdhales at optusnet.com.au
> > To: asterisk-users at lists.digium.com
> > Subject: Re: [asterisk-users] Agent channel...
> > 
> > 
> > From memory, I have seen something similar done with the SIPPEERS 
> > function (curcalls) but it's too fuzzy for me to remember it fully.
> > 
> > Paul Hales
> > NTS
> > 
> > 
> > Carlos Chavez wrote:
> > > I have a customer with a small outgoing call center. Usually only
> 3 to
> > > 5 agents online. We are still using Agent/XXX channels in this
> > > application on Asterisk 1.4.18. I have an autodialer that is
> making the
> > > outgoing calls and then dropping them into a Queue where all the
> agents
> > > are logged on.
> > >
> > > My problem is that when an agent makes a call on his/her phone the
> > > queue always says that the agent is "Not in use". I have
> call-limit set
> > > to 1 on all sip phones that are used for agents but I can see that
> the
> > > queue tries to send a call to the agent. Since the agent has a
> limit of
> > > one the call gets rejected but instead of going back to the queue
> it is
> > > dropped.
> > >
> > > How can I make sure the agent will show "In Use" when they make a
> call?
> > >
> > > 
> > >
> ------------------------------------------------------------------------
> > >
> > > _______________________________________________
> > > -- Bandwidth and Colocation Provided by http://www.api-digital.com
> --
> > >
> > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> > > Register Now: http://www.astricon.net
> > >
> > > asterisk-users mailing list
> > > To UNSUBSCRIBE or update options visit:
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
> > 
> > _______________________________________________
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com
> --
> > 
> > AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> > Register Now: http://www.astricon.net
> > 
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
> ______________________________________________________________________
> Use video conversation to talk face-to-face with Windows Live
> Messenger. Get started.
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: application/pgp-signature
Size: 197 bytes
Desc: This is a digitally signed message part
Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20080723/32d66df2/attachment.pgp