similar to: IAX2 support of video

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

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

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 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 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 using Asterisk 1.4.18. I am implementing Penalties for my agents. What is happening: two agents configuired one agent with penalty 1 and the other with penalty 2. All the calls must go first to Agent 1 and if his line is busy then only then agent 2 will get the call. However my queues are not behaving in this manner. I have impmemnted ringall strategy. Now when first call comes it

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

Perhaps I was a little too hasty in my conclusions of dysfunctional fax on the SPA-2000. It turns out I have a one way audio problem on line one of my SPA-2000. I have all the correct settings according to the comments in the wiki, but the problem persists. However, if I do a hook flash out of and back in to the call that isn't transmitting audio, it works fine. My sip.conf entry for the

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

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

Hello. I've got 1 queue setup with 2 possible agents. Agent 1 is logged in and awaiting a call via AgentCallback. Agent 2 has not logged in. An outsider (caller A) calls in and is placed in the queue, cytelcs. Agent 1's phone rings and Agent1 and A talk. While they are talking, caller B calls in. Caller B is correctly placed in the queue and hears music, however this shows up in asterisk

Hello, I ran into an issue today as I began to transition into a production environment from my Puppet testbed. I am using Puppet Enterprise 2.7 for Ubuntu (x64) and ran through the Installer and configured the Console, Cloud Provisioner, and Master on the same box. This all went well. I then began setting up agent1 for testing and after installing PE, updating the environment in the

** 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:

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>

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> --

Thanks Scott. I was able to get the basic concept to run. However, it seems PJSIP INVITE for the Dial also does not support added headers. The Local channel dial plan did have the channel variable values. I added them as SIP headers, then Dial(PJSIP/Agent). The INVITE for the Dial on PJSIP continues to not include the SIP Headers I added. For chan_sip, I have no problem with this. Even the

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

Greetings. Anyone have any recommendations for studying for the dCAP Certification? Other than the expensive Digium courses, there doesn't seem to be anything online. Thanks, Michael Gilleran -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130607/3b4766ea/attachment.htm>