Sorry for the top-post... If you do a core show application AddQueueMember from the cli, you'll see the option I was referring to.
You'll also need to make sure you're properly reporting device state to asterisk. I think this means you need to set a call-limit for each sip peer that you want to monitor in sip.conf (we use 25 so there are no accidental limits actually applied), and setup hints in your extensions.conf for each peer. Thanks, --Warren Selby On Oct 14, 2010, at 11:36 PM, Matt Darnell <[email protected]> wrote: > Warren, > > I tried using AddQueueMember to add agents. > > If they a user is on a call asterisk shows: > Members: > SIP/101 (dynamic) (Not in use) has taken no calls yet > No Callers > > We are using 1.4.36. > > What did you use to keep track of the extension state? Didn't see any > option for that at > http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20AddQueueMember > > Thanks for the help. > > -Matt > > > On Thu, Oct 14, 2010 at 6:04 PM, Warren Selby <[email protected]> wrote: >> What version of asterisk are you using and method are you using to login >> your agents? I recently had this issue with a 1.4.33 install where the >> agents logged in with agentcallbacklogin. In the end I had to move them away >> from chan_agent altogether, using dynamic agents and AddQueueMember, which >> has a parameter for designating a device to keep track of the state for that >> member. Seems to be working for now. >> >> Thanks, >> --Warren Selby >> >> On Oct 14, 2010, at 10:13 PM, Matt Darnell <[email protected]> wrote: >> >>> We have a queue that agents log into through the dial plan. Extension >>> Sip/101 logs in as Agent/101 >>> >>> We have 'ringinuse = no' in the queues.conf file. >>> >>> The issue is that when Ext 101 is on a 'non queue' call (they placed a >>> call, someone called their DID, etc) they still receive queue calls. >>> >>> Is there a way to stop this from happening? >>> >>> -Matt >>> >>> -- >>> _____________________________________________________________________ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> New to Asterisk? Join us for a live introductory webinar every Thurs: >>> http://www.asterisk.org/hello >>> >>> 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 -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> 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 -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
