similar to: Queue App - Free agent and waiting calls

Asterisk 11.1.0 One queue with strategy=leastrecent. (Full queues.conf below.) Occasionally (several times today), a caller will get "stuck" in the queue - there are operators available to take the call, but the caller stays in the queue for a long time. Any idea what might cause this, or where I can start looking to debug it? I'm going to start digging through the queue log

First of all I would like to thank Mark for getting roundrobin to go roundrobin. Good job. Now we have some options here for leastrecent and fewestcalls strategy. It needs some work on the logic and Mark recommend that I ask the list and get some input before he makes any changes to it. fewestcalls from what I have seen would always ring the agent with the fewestcalls first then go into

Is there any way to do this? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120307/77764e4b/attachment.htm>

Is a normal functionality? when I do #dahdi_cfg -vvvvvv In my Asterisk console shows this.... [Jul 24 13:39:08] NOTICE[30263]: chan_dahdi.c:9461 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 If I do this a lot of times...then [Jul 24 13:39:20] NOTICE[30263]: chan_dahdi.c:9461 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 [Jul 24

Hi ! I have a python script that create and move .call files to /var/spool/asterisk/outgoing Sometimes...(in this case after 500 successfull calls) Asterisk don?t make the calls and the .call files are in the "outgoing" forever... Any Ideas? I'm using Asterisk 1.4.22 (in 1.4.36 was the same behavior) In my python script I move .call files using ... import shutil

I have a server with an asterisk ss7 link connected to a Siemens working well for over a year. A few days ago I started having problems with signaling. I found the following logs in / var / log / messages Sep 12 11:49:25 call3 kernel: [1018427.030959] dahdi: Master changed to TE2/0/2 Sep 12 11:49:25 call3 kernel: [1018427.120740] dahdi: Master changed to TE2/0/1 Sep 12 11:49:26 call3 kernel:

I try to configure that only registered sips can make calls. How can I do that? I was looking in sip.conf but I didn?t found wath opition configure this functionality. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071212/cfed2687/attachment.htm

Hi Everyone, We have three different Queues set to "leastrecent" strategy and from time to time I hear someone complain that they receive short rings (partial ring cycle) and since it's not their turn even if they pickup the phone the call is not given to them since the Queue is actually hitting someone else at the same time. Is this short ring an indication of some sort for

I have this [TRONCAL-SIP] exten=>225/91,1,Answer exten=>225/91,2,Echo exten=>225/91,3,Hangup exten=>225/92,1,Answer exten=>225/92,2,Playback(conf-invalid) exten=>225/92,3,Hangup When I make a call CLI> -- Recv IAM CIC=8 ANI=91 DNI=225 RNI= redirect=no/0 complete=1 Dont work If I add this rule exten=>225,1,Answer Works ok -------------- next part --------------

I have a simple 4 person ACD queue using the AgentCallback function. No matter what strategy I use, anytime someone calls into the queue asterisk dials the agents in the order that they are listed in the agents.conf file. This doesn't seem right to me, or am I wrong.

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 problem with the queues on Asterisk. The setup is Asterisk@Home v1.0 with Asterisk 1.0.7. I have 1 queue (4500) set up, with leastrecent strategy. There are no agents configured in this queue. Agents log in by dialing 4500* on their phones. All incoming calls are sent to the queue. Calls wait 120 seconds in the queue, and are then sent to voicemail extension 310. My problem is

I want the callers don't join in a queue when the agents are busy. I suposse it is easy but i can't get the solution for this. Can you suggest me something? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081229/3545ab8c/attachment.htm

I have an Asterisk 1.6.0.28 system, with a queue called 'marketing'. Everything appears normal, but the status of the members never changes from 'not in use', even if they are being rang or are in a call. Members are added like so: queue add member SIP/1406 to marketing penalty 0 as SIP/1406 state_interface SIP/1406 And they are present as a hint: exten =>

How can I execute an EAGI script with params from extensions.conf Example python script: InfMsg -s 1 in my extensions.conf exten => 492,1,Answer exten => 492,2,eagi,InfMsg -s 1 exten => 492,3,Hangup() It doesn?t work my * report... -- Executing [92@telpin-112:2] EAGI("Zap/4-1", "InfMsg -s 1") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/InfMsg

Hello all, we're using asterisk 1.2.12.1 in an Inbound callcenter using the queue application. If there are many calls in the queue, it sometimes takes up to 30 Seconds before a call is distributed to an agent. For example there are 10 callers in the queue, an Agent is finishing a call and it takes up to 30 seconds before his phone rings again. We're already set the

Hi , I;ve Asterisk 1.6.0 with static agents (sip softphones with extns 100 & 101 ) in a queue..When a caller arrives in queue , it lands on first 100 , 100 then does a blind transfer to 101 .. so that the caller can converse with 101 .. strangely enough the queue_log shows : 1253814090|1253814090.12|55365|NONE|ENTERQUEUE||98221232123

We're moving a somewhat complicated call center over to an Asterisk system, and I'm looking for documentation on queue/agent configuration. So far I haven't found anything on the Digium or Asterisk websites, and I was hoping that someone could point me in the right direction. Thanks, Devon

Hi, I've recently upgraded Asterisk to the latest version 1.4.9 on a PBX that manages several queues, but at least on one queue strategy (leastrecent) it doesn't seem to be distributing the calls has it should. I think this strategy should work like this: a) Make a list of available agents and their idle time (time since last call) and