similar to: Queue/Agents problem with 1 agent

This just started happening today. I've got 1 queue and 6 agents. All logged in. I tell the service people to ignore my call if they see my caller id. I call the queue and watch as asterisk bounces me around the phones. Our agent ring time is 5 second timeout and a 5 second wait time before trying next agent. I get the same message in console for each agent attempt: -- Executing

Hi list, Using Asterisk 1.2.10 I am getting seriously confused by Queues and Agents. So far I configured my queue and agents, had my agents login using agentcallback. Call enters queue agent gets a call, if agent doesn't answer after 20 seconds a flag is set in AstDB (thanks to: Leo Ann Boon), call is returned to queue and the cycle continues. If the same agent doesn't

As I understand it, you'd enter the extension at which you wish to be called back at, your 9665 has nothing to do with it. Instead of dialling 28 you could dial 9665 and that would add that SIP phone as an agent to the cytelcs queue. Steve -----Original Message----- From: Harold Workman [mailto:hworkman@cytelcom.com] Sent: 22 June 2004 18:54 To: asterisk-users@lists.digium.com Subject:

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

> -----Original Message----- > From: Richard Lyman [mailto:pchammer@dynx.net] > Sent: Wednesday, December 20, 2006 4:29 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Re: Match a Numer - then continue with, > dialplan > > > Douglas Garstang wrote: > >> -----Original Message----- > >> From: David

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

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

Greetings to all. I am currently implementing call queues for a customer and have come across several "problems". The customer is an airline representative, and will be using call queues for different airline reservations. The customer requires that any agent be able to login to any number of queues. This means that queue members have to be dynamic, not using "member =>

List, I've got 2 queues with 10 agents in both queues. One of the agents is mainly responsible for queue_1, and the others mainly for queue_2 so i've defined the following in my queues.conf [queue_1] strategy=ringall member=>Agent/1,2 member=>Agent/2,1 member=>Agent/3,1 member=>Agent/4,1 [queue_2] strategy=ringall member=>Agent/1,1 member=>Agent/2,2

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 =>

** 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:

Hello list, one of our clients is going to be deploying a system with over 200 differently composed queues and 100 agents. We are going to do a full test of the viability of this solution before deployment, but I was wondering if anyone has experience of such a setup and if there are any obvious problems or no-nos. Any suggestion welcomed, l. -- Home of QueueMetrics -

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)

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

> -----Original Message----- > From: David Gomillion [mailto:dgomillion@eyecarenow.com] > Sent: Wednesday, December 20, 2006 10:27 AM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Re: Match a Numer - then continue with, > dialplan > > > I think you're making it far too difficult. > > What I do is something like this: > > [outgoing]

Hi all!! Does anyone have a short example or even better - a working AGI script that uses "GET VARIABLE' from a /var/spool/asterisk/outgoing call that uses "SetVar"? Here's what I've tried with no luck so far: sample.call ================= Channel: SIP/1000 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Application: Agi Data: playTasks.agi Callerid: Nightly Processor

Hi, my situation is: incoming call goes into the queue and is picked up by callback agent. The agent then wants to transfer the call to another device (another SIP phone). But 'transfer' button doesn't work and '#' button attempts to start channel monitor. Tried with both Queue(testq) and Queue(testq,tT). Is it meant as a feature that agents won't transfer calls at

I am originating a command via the AMI with this... Action: Login Username: xxx Secret: yyy ACTION: Originate Async: yes Timeout: 60000 Exten: callback Channel: Local/6505551212 at LegA Callerid: 849120 Context: default ActionID: 849120 My LegA context: ----------------------- context LegA { _X. => { Dial(SIP/${EXTEN}@Provider); } } And my default context:

Well because I had sooo may problems with chan_agent.c I wrote this. I'm releasing it under LGPL but if you use it or anything please let me know. It'd be interesting if anyone finds this more useful than just a pile of junk. I've included a (working) example extensions file. SIP phones are assumed to have the same identifier as their extension number, but it'd be trivial to

Greetings all, We are settng up a fair sized call center on Asterisk, but we are having some issues with our agents not knowing if they have logged in and logged out. Prior to beginning our migration to VoIP the agents logged into our nortel phones and confirmation was displayed on the phone. My question is has anyone out there done anything from Asterisk that can change the display on