similar to: Queue reload

How do I clear a global variable for good? I have a situation of needing to use global variables to aide in channel communication, but will be changing the name within a defined scope. Additional Background... I want to get a variable from a channel (child) that is created by another channel (parent), however the execution of the parent channel does not continue until the child channel is gone.

Hey guys, i'm wondering if there is any good way to get app_queue working in real roundrobin strategy. The idea is to specify a call list of, lets say, 3 agants. Those agents should always be called in the correct defined order. So all calls have to get the following agent priority: 1st Agent -> 2nd Agent -> 3rd Agent I've actually solved that by defining penelty for the accounts,

Hello everyone, This is is my first email to this group. I'm am writing a small php program to pull some info out of our Asterisk's queue_log. I'm having trouble figuring out what some of the parameters mean. Here's an example: 1118098465|1118098465.47|salesq|NONE|ENTERQUEUE||"Ray Balbin 25" <(716)250-3405> I found a doc that tells me about everything from

Does anyone know what can cause queue members to go into a status of "Unknown"? pbxtel-01*CLI> queue show cs has 2 calls (max unlimited) in 'rrmemory' strategy (24s holdtime), W:0, C:447, A:20, SL:91.7% within 60s Members: SIP/1405 (dynamic) (Unknown) has taken no calls yet SIP/1420 (dynamic) (paused) (Not in use) has taken no calls yet SIP/1442

I am trying to figure out which one of our agents is answering the calls. According to http://www.voip-info.org/wiki/view/Asterisk+log+queue_log the only time the queue_log puts the channel (agent) is during logoff & logon. There is the connect & completeagent message, but it doesn't show which channel (agent) answered the phone. I can't even figure it our cross referencing the

Hello, Polycom ip soundpoint support ACD login/logout . Can we configure asterisk with polycom ACD support? Regards Harry ___________________________________________________________________________ Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et l'international. T?l?chargez sur http://fr.messenger.yahoo.com

Hello all, I need to write a queue_log parser that is going to implement more or less the functionalities described here http://lists.digium.com/pipermail/asterisk-users/2003-July/014965.html of course not everything from scratch, but this is where I'd like it to go. I am looking for - previous work (maybe it's ready somewhere and I've never heard of it) - suggestions -

Hello, I am using domain event notifications from libvirt-event API in my application and it seems to work fine, except for 'suspend' and 'resume' events where I keep receiving duplicated notifications. Similarly, the example provided with libvirt (in examples/object-events/event-test) also produces twice as much notifications for 'suspend' and 'resume' than for

Hi, If an agent doesn't take a call.. is there some way I can PAUSE them instead of logging them out?

Hey all! I'm trying to do a bit of logic here so that a user only has to dial one code to pause/unpause in a queue (e.g. *0 will (un)pause depending on the users's state). My logic looks fine to me but every time ${PQMSTATUS} shows up empty. Here's the extensions.conf part.... exten => *0,1,NoOp(${PQMSTATUS}) exten => *0,n,Macro(user-callerid,SKIPTTL,) exten =>

Guys, I have a problem. I have a queue on asterisk 1.8 that members are added dynamically via the AMI QueueAdd. When you run the CLI a "reload app_queue.so" all members who were in the queue disappear. This is a bug or some parameter that I do not know? Would have another way to do the reload queue without any risk to members who are already in it? tks Ed -------------- next part

I need to send calls to a bank of servers in a round robin fashion. Has anybody implemented this in dialplan language? If not, in some other fashion. </edg>

Hi, I have several agents, who all log into multiple queues. What I want to happen (but doesn't seem to be) is: Agent 5 is logged into queues 1,2,3 Agent 4 is logged into queues 1,3 A call comes into queue 1, and goes to agent 5. Agent 5 answers the call and finishes it. A call comes into queue 3. I want this call to go to Agent 4, as opposed to going to agent 5 (which is what it is doing

Hi, I'm a little bit stuck with Queue app. I'm putting callers into the queue and have them hear music on hold when all (static) agents are busy. This is easy. But when agent become available I want the caller to hear a ringing tone (with message that his call has been routed to the support representative). Is this somehow doable? Thanks, David

When a caller in the queue is connected to an agent, the call is placed to the extension and context specified using Agentcallbacklogin. This allows for me to add extra things to the diaplan *before* calling the agent. Now, I want to be able to use a device, rather than agents. So I can use addQueueMember and add my SIP device. However, I still want to do a couple of things before the device

Hi all, I am seeing a strange problem with Asterisk queue. I am not sure if it's my configuration which is wrong or there's something with Asterisk. I am using Asterisk 1.4.2 and i have a queue with one MGCP member. When i tried to call the extension number directing to the queue, the MGCP phone is not ringing. However, it is fine to call the MGCP phone directly. The strange thing is

Asterisk: 1.6.2.15 On the production server I've modify the /etc/asterisk/queues.conf file. Now in CLI I wan't to reload queue configuration gracefully. I did: virtual-pbx*CLI> queue reload members office virtual-pbx*CLI> But `queue show office` tells me that nothing has changed. I tried to reload all -- `queue reload all': virtual-pbx*CLI> queue reload all [Dec 28

Hi, I'm writing a Java client/server application that talks to the Asterisk manager interface via the asterisk-java stuff. The idea being it will give you an app to run on your desktop that monitors your phone essentially. Once I've got something vaguely working it will be released under the GPL and hopefully people will contribute to it etc... As part of this, I'm currently

Hi all, I am having the following trouble with recording calls: When calls come into the support line did number, the call starts to record on the first queue, but appears to hang up when the call actually connects to the engineer (ie I see "got hangup request" on the cli and then mixmonitor ends.) I am guessing this has to do with the announce file that is played to the engineer

I run a centos6.2 guest VM in xen4.1.2, the id of centos-domU is 22. The info of dom0 is: 2.6.32-131.21.1.el6.xendom0.x86_64 #1 SMP Tue Dec 13 18:09:29 EST 2011 x86_64 x86_64 x86_64 GNU/Linux Centos-domU: 2.6.32-220.7.1.el6.x86_64 #1 SMP Wed Mar 7 00:52:02 GMT 2012 x86_64 x86_64 x86_64 GNU/Linux At dom0, I went like this: [malei@xentest-4-1 Tue Apr 30 ~]$ sudo gdbsx -a 22 64 9999 Listening on