similar to: Calling same queue member all the time

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

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 =>

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, 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

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

** 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 have 1.2.9.1 installed. It always rings first available agents for 15 seconds, then rings and hangs up the next agents straight away, then ring the next agents for 15 seconds. It goes as a loop. Any one has the following same problem? Thanks. Agents.conf [general] persistentagents=yes [agents] autologoff=60 wrapuptime=15000 ackcall=no group=1 agent => 7130,7130,agent1 agent =>

We are using G711 for phones to talk to Asterisk and G729 licenses at asterisk to talk to ITSP Could you please suggest transcoder to use from G711 and G729 and which is comptible with Asterisk. We will like to avoid using TDM if possible Also i remember that initially we didn't have G729 and were using only 711 for with vicidial but then also we had same problems. at that time it was only 2

The OpenSSH and ssh.com agents use a completely different set of messages for dealing with DSA keys, so I don't expect the OpenSSH client to be able to get DSA keys from ssh-agent2. However, if I'm running "ssh-agent2 -1", I expect OpenSSH to be able to use RSA keys stored in ssh-agent2's agent1 compatibility box. And it does. However, there's a problem. If I'm

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

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

Does IAX2 support a video call ? Jerry

I am having trouble with the AgenCallBackLogin app. I can't seem to define a context for the queue. Here is the relevant configs: queues.conf: [general] [default] [q_lo_1] music = default strategy = ringall context = c_in_1 timeout = 15 retry = 2 maxlen = 0 member => Agent/@3 agents.conf: [agents] autologoff=10 wrapuptime=15000 group=1 agent => 1001,1234,Agent1 agent =>

I am reading at http://www.asteriskguru.com/tutorials/queues.html They are using member in both static and dynamic method. member => <technology>/XXXX -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110519/ec8d19e1/attachment.htm>

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)

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, I am setting up a test call center using *. I am using one Zap channel (Wildcard TDM400P REV E/F -- 4 FXO modules) for incoming call and sip phones (SjPhone) for call agents. I have setup queues and agents. While testing I found that if the agent presses * key in soft phone while attending calls Zap channel gets hung up, and another customer can call. But if the caller hangs up (for example

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