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: [EMAIL PROTECTED] > > To: [email protected] > > 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
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_______________________________________________ -- 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
