similar to: Static Vs Dynamic queue confusion

How to get rid on following.. why its Invalid ? holler*CLI> queue show queue1 queue1 has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s Members: Agent/7201 (Invalid) has taken no calls yet Agent/7202 (Invalid) has taken no calls yet No Callers -------------- next part -------------- An HTML

Hello, I'm trying to setup a queue where call goes from agent to agent in strictly set order. I have queue (roundrobin): Agent1 penalty 1 Agent2 penalty 2 Agent3 penalty 3 When I call to this queue Agent1 rings. If this agent does not take the call, after set timeout same Agent1 is dialed again. The call never goes to Agent2 (only when Agent1

Hi, I have a scenario that I am failing to implement using the Queue app, but which I had thought would be commonplace... 1) (this bit works fine) I want a queue caller to have access to the basic set of agents initially, with an overflow to additional agents if they are busy - This is done using penalty: queues.conf: member => SIP/dev1,0,Agent1 member => SIP/dev2,0,Agent2 member =>

Hello, I was planning to use the output of asterisk -rx "show queues" in a script when I noticed that sometimes asterisk only outputs the first line of the response. e.g: debian:/# asterisk -rx "zap show channels" Chan Extension Context Language MusicOnHold debian:/# asterisk -rx "zap show channels" Chan Extension Context Language

Good Day list, I have a friend who is interested in implementing an asterisk implementation at his offices. The configuration would consist of the following Site A ---- Asterisk Box With 12 incoming lines and 15 phones Extensions 101-115 Site B ---- Asterisk Box With 4 incoming lines and 7 phones Extensions 201-207 Site C ---- Asterisk Box With 4 incoming lines and 6 phones

Seeing as the "leastrecent" strategy does not work for us (it will *always* call the leastrecent agent, even if that agent is busy, and will not move to the next agent) I thought I'd try the roundrobin strategy. This seemed to work - however, I had my supervisor telling me today that a certain agent seemed to be getting most of the calls - and when I checked, out of 50 inbound

** I installed PE Master on one VM and Agents on two VMs pointing to master . Agent1 VM 64 bit works fine , but agent2 VM 32 bit fails with below error. Only difference is architecture. One more note both the agent nodes were accepted from Dashboard,so master has both the certificates. Any help will be greatly appreciated. puppet-enterprise-3.1.0-el-6-i386]# puppet agent --test Info:

Hi, I am new in puppet, and I just wonder whether it is possible to create multiple levels of puppet masters. Can puppet work this way? First-level(master): root-master Second-level(masters): master1, master2 Third-level nodes(as agents): agent1, agent2, agent3, agent4 All master nodes in the second-level are agents of root-master, and each of third-level

Hi i have an issue getting agentLogin working /etc/asterisk/queues.conf member => Agent/1001 member => Agent/1002 extension.conf exten => 110,1,Wait,1 exten => 110,2,AgentLogin() now, i call 110 by a firefly client, trying to login in as 1001 agent: Aug 12 16:31:36 DEBUG[1103408048]: chan_sip.c:4423 build_route: build_route: Contact hop: <sip:sip3@192.168.1.151:5060> --

Hi, We are using Vicidial and sometime even when agent disconnects, outgoing call originated by dialer is still active. Since call was initiated by dialer and then bought into meetme conference of agent and we can't corelate this call to any agent channel. When agents are dialing, channels doesn't show calls vicidial2*CLI> show channels Channel Location

https://bugzilla.mindrot.org/show_bug.cgi?id=3153 Bug ID: 3153 Summary: Prefer user specified keys to avoid the agent overloading MaxAuthTries before even trying the key that was specified Product: Portable OpenSSH Version: 8.2p1 Hardware: Other OS: Linux Status: NEW

I''ve been confused by this question for near two days ...my puppet master version is 2.7.9-1.el6 and client versiong is 2.6.16-2.el5. This is what my command lines shows: *[root@agent1 ~]# puppet agent --server=edward --test --waitforce 30* info: Creating a new SSL key for agent1 warning: peer certificate won''t be verified in this SSL session info: Caching certificate for ca

On Mon, 2003-11-03 at 21:53, Dave St John wrote: > Anyone know of or how to capture the pid via the command line in bash? pidof ??? $ pidof httpd 827 825 824 788 -- Drew <p>--- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage: http://www.icecast.org/ To unsubscribe from this list, send a message to 'icecast-request@xiph.org' containing only

Hi, I am able to install my RPM via this puppet code ... [root@agent1 ~]# puppet apply -v install_named_conf.pp info: Loading facts in /opt/puppet/share/puppet/modules/stdlib/lib/facter/facter_dot_d.rb info: Loading facts in /opt/puppet/share/puppet/modules/stdlib/lib/facter/puppet_vardir.rb info: Loading facts in /opt/puppet/share/puppet/modules/stdlib/lib/facter/root_home.rb info: Loading

Paul wrote > >;Pause/unpause Queue >exten => 424,1,PauseQueueMember(|SIP/${CALLERID(num)}) >exten => 424,2,Playback(unavailable) >exten => 424,3,Hangup >exten => 425,1,UnPauseQueueMember(|SIP/${CALLERID(num)}) >exten => 425,2,Playback(available) >exten => 425,3,Hangup > Following your suggestion and a number of postings and articles I have

Hi! I want to know the way that calls are answer in this case... I have queue1 and queue2, one agent that receive call from both queues. queue1 <- call1 queue1 <- call2 queue2 <- call3 queue2 <- call4 In my test the agent answer calls in this order: call1,call3,call2 and call4. I think this must be in this order call1,call2, call3, call4 like a big FIFO. Its ok this behavior? Could

Ex-girlfriend logic not working in latest CVS? Incoming sip calls don't work. Anyone else seen this problem? Extension logic looks good: exten => 6153248305/_931NXXXXXXX,1,Queue(queue1); exten => 6153248305/_615NXXXXXXX,1,Queue(queue2); ;exten => 6153248305,1,Queue(queue3); show dialplan looks good: -- Added extension '6153248305' priority 1 (CID match

Hi all, I'm trying to make some simple MOH (Music On Hold) working. So far I've failed miserably - so I turn here for help. Basically I've been using the wiki and all the sample confs I could from there and via google. The queue system seems to work fine with my limited setup. Just 2 IAX2 clients where I keep Client B busy (by making it listen to mp3 via ext. 777) but logged into

> -----Original Message----- > From: Watkins, Bradley [mailto:Bradley.Watkins@compuware.com] > Sent: Thursday, June 15, 2006 10:36 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] DUNDi Not Able to > HandleComplexFailoverSituations > > > Is it possible for you to explain in more detail the > situation involved.

I have both the PJSIP add and the chan_sip way of adding SIP headers in there. The Verbose is showing the variable value is there. The INVITE to PJSIP/Agent1 does not include either X-My-DNID or X-My-DNID2 headers. exten => 1234,1,Verbose(X-My-DNID:${MY_DNID}) same => n,Set(X-My-DNID=${MY_DNID}) same => n,Set(PJSIP_HEADER(add,X-My-DNID2)=${MY_DNID}) same => n,Dial(PJSIP/Agent1)