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<body class='hmmessage'>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.. <BR>
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<BR>
this is supposed to solve the issue.. <BR>
if you didn't have problems with queues before upgrading.. then it's a matter of how you log your agents to your queue<BR>
Salam<BR>
Tarek Sawah<BR><BR><BR>
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<P align=center> </P><A href="http://www.tareksawah.com/"></A></DIV><BR><BR>
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> Date: Mon, 30 Jun 2008 00:54:31 +0300<BR>> From: atis@iq-labs.net<BR>> To: asterisk-users@lists.digium.com<BR>> Subject: Re: [asterisk-users] 1.4.21 + Realtime Queues = Agents Not Ringing?<BR>> <BR>> On Sun, Jun 29, 2008 at 7:02 PM, Sherwood McGowan<BR>> <sherwood.mcgowan@gmail.com> wrote:<BR>> > Sherwood McGowan wrote:<BR>> >> Gentlemen,<BR>> >> I'm using 1.4.21 SVN Tag, and have the queues set up to use Realtime.<BR>> >> This system works fine with 1.2.28, and everything loads fine with no<BR>> >> errors, but when I log an agent in I see the extra message "(not in<BR>> >> use)" by their listing and they are not rang by asterisk when their<BR>> >> queue is called.<BR>> >><BR>> >> Any ideas?<BR>> >><BR>> > Nobody else?<BR>> ><BR>> <BR>> Have you checked call-limit and state information for SIP peers? That<BR>> was changed between 1.2 and 1.4, and could affect queue state. See the<BR>> UPGRADE notes.<BR>> <BR>> Otherwise You'll have to set "core set debug 2" and "core set verbose<BR>> 3", and post full log (debug+verbose) where agents got logged in (if<BR>> you have also realtime members, just execute "queue show xxxx" on CLI.<BR>> Then you'll have to give one call to agent, talk for little and<BR>> disconnect. Then just post that log here.<BR>> <BR>> Regards,<BR>> Atis<BR>> <BR>> -- <BR>> Atis Lezdins,<BR>> VoIP Project Manager / Developer,<BR>> atis@iq-labs.net<BR>> Skype: atis.lezdins<BR>> Cell Phone: +371 28806004<BR>> Cell Phone: +1 800 7300689<BR>> Work phone: +1 800 7502835<BR>> <BR>> _______________________________________________<BR>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --<BR>> <BR>> AstriCon 2008 - September 22 - 25 Phoenix, Arizona<BR>> Register Now: http://www.astricon.net<BR>> <BR>> asterisk-users mailing list<BR>> To UNSUBSCRIBE or update options visit:<BR>> http://lists.digium.com/mailman/listinfo/asterisk-users<BR><br /><hr />The other season of giving begins 6/24/08. Check out the i’m Talkathon. <a href='http://www.imtalkathon.com?source=TXT_EML_WLH_SeasonOfGiving' target='_new'>Check it out!</a></body>
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