[asterisk-users] 1.4.21 + Realtime Queues = Agents Not Ringing?

Tariq .. tareksawah at hotmail.com
Mon Jun 30 10:22:09 CDT 2008


i have been through this with Dynamic agents and callback .. i used to use addQueueMember but it caused me troubles when joining a queue.. because sometimes the agent would be in the queue twice.. 
my suggestion to you is to check if you can make calls between two members of the queue .. then do a call from the agent to the queue number "if dialable" and then see if the queue would forward your calls to the agents.. if not.. i would suggest that you use static Agents like ... member=SIP/xxxx,0
this is supposed to solve the issue.. 
if you didn't have problems with queues before upgrading.. then it's a matter of how you log your agents to your queue
Salam
Tarek Sawah

 

> Date: Mon, 30 Jun 2008 00:54:31 +0300> From: atis at iq-labs.net> To: asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] 1.4.21 + Realtime Queues = Agents Not Ringing?> > On Sun, Jun 29, 2008 at 7:02 PM, Sherwood McGowan> <sherwood.mcgowan at gmail.com> wrote:> > Sherwood McGowan wrote:> >> Gentlemen,> >> I'm using 1.4.21 SVN Tag, and have the queues set up to use Realtime.> >> This system works fine with 1.2.28, and everything loads fine with no> >> errors, but when I log an agent in I see the extra message "(not in> >> use)" by their listing and they are not rang by asterisk when their> >> queue is called.> >>> >> Any ideas?> >>> > Nobody else?> >> > Have you checked call-limit and state information for SIP peers? That> was changed between 1.2 and 1.4, and could affect queue state. See the> UPGRADE notes.> > Otherwise You'll have to set "core set debug 2" and "core set verbose> 3", and post full log (debug+verbose) where agents got logged in (if> you have also realtime members, just execute "queue show xxxx" on CLI.> Then you'll have to give one call to agent, talk for little and> disconnect. Then just post that log here.> > Regards,> Atis> > -- > Atis Lezdins,> VoIP Project Manager / Developer,> atis at iq-labs.net> Skype: atis.lezdins> Cell Phone: +371 28806004> Cell Phone: +1 800 7300689> Work phone: +1 800 7502835> > _______________________________________________> -- 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
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