similar to: member queue refresh

I am experiencing this: 1 - A,B,C are SIP users logged on QUEUEA with ringall strategy 2 - I call QUEUEA 3 - A,B,C start ringing 4 - nobody answer.... 5 - D logs on the QUEUEA 6 - D doen's receive any call, but A,B,C are still ringing How can i avoid that? I'd like that when D joins the QUEUEA it will immediately receive the call that is still ringing on other users... Thanks in

Hi, My queue members use Local channels and their queue state is "In use" while their hint value is "Idle". Since I have Ringinuse=no, I'm experiencing issues such as incoming calls waiting too much because the agent's phone isn't ringing even though it's idle/free. I read somewhere that this is a "known bug" in 1.4 and should be fixed in 1.6. I

Hi list, I'd like to have the callers to listen to the advertisement (music on hold) before the agents answer them. So, I have wrapuptime=10 in queue.conf, but the call still goes straight to the agents without delay. Here's my queue.conf: [general] persistentmembers = yes [738] musiconhold = empty ;musiconhold = default ;announce = q-738 ;strategy = ringall strategy = rrmemory

Dear all on a debian amd64 i've installed (from source) asterisk 1.4.30 On the system we have in average 50 concurrent calls in queue and 40 sip members. I'm experiencing an apparently random problem: sometimes some users receive 2 calls from asterisk, apparently ignoring the ringinuse=no settings. It appears on users that are members of many queues As you can see from the log, the

Can anyone shed any light on an issue with agent penalties? I have 2 queues set up with agents working both queues, but where agent 1 should have a penalty for queue 2 and agent 2 should have a penalty for queue 1. When a call is sent to either queue, it rings agents with and without penalties at the same time. I set up a second system and cannot replicate the issue on the test system. I

Hello I have problem with transfer call if using ACD When i using ACD with agent and queue setting, i cannot monitor call and transfer call. this's my setting - i have 2 IAX phone (phone number as 201, 202), agent.conf agent => 1001,4321,member 1 agent => 1002,4321,member 2 agent => 1003,4321,Tin then, queues.conf [MyQueue] music=default strategy=ringall timeout=30 retry=5

There is an environment Setup uses Asterisk 1.2 and doesn't want to upgrade. There is an issue while a call goes to any queue we create, the call is being disconnected after 20 seconds and it is hangup. The following is the configuration: - vi /etc/asterisk/queues_additional.conf [8] wrapuptime=0 timeout=30 strategy=ringall servicelevel=5 retry=4 reportholdtime=No queue-youarenext=

Hi - I have a queue setup with 4 agents. The people working the queue tell me that when a new call goes into the queue, both the agent and the caller hear a tone. These are static agents - could it be ringing the line again, even though the agent is on a call? We are using GXP2000's with only 1 line programmed in. thanks Todd queue.conf > [601] > announce-frequency=0

Hi all, I have 2 question. I have a call center queue with 5 agent; the following are the configuration files: *queue.conf* [name_of_queue] musicclass = default announce = queue-name_of_queue strategy = ringall servicelevel = 60 context = callcenter timeout = 60 retry = 5 wrapuptime=15 autopause=no maxlen = 0 announce-frequency = 60 periodic-announce-frequency=30 announce-holdtime = yes

hi, i'm testing asterisk i would like to log some agents in a queue and see how works the system. i've added to queue.conf Code: [700] wrapuptime=0 timeout=15 strategy=ringall retry=5 queue-youarenext= queue-thereare= queue-thankyou=queue-thankyou queue-callswaiting= music=default monitor-join=yes monitor-format= maxlen=0 leavewhenempty=no joinempty=yes

Hello list ! I have some configuration issues with queues, but I'm sure they are minor and for someone who has already configured queues it could be trivial. This is my queue configuration : [VC_support_queue] musicclass = default strategy = ringall timeout = 20 retry = 5 wrapuptime=15 autofill=yes autopause=no maxlen = 0 setinterfacevar=yes announce-frequency = 0

Hi, I have the same problem with the queue configuration When I receive 2 calls only 1 phone ring even if more agent's phone are free. The second call will go to an other agent only if the first call is pickup. Somebody have a solution ? This is my config file : Queue.conf [general] ; ; Global settings for call queues ; ; Persistent Members ; Store each dynamic agent in each queue

Hello List, First, sorry for my bad English skill, I'm French. We have an asterisk 1.8.3.2 built from sources with a simple Queue : [TestQueue] strategy=ringall timeout=15 retry=1 timeoutpriority=conf ringinuse=yes wrapuptime=2 member => SIP/002E31,0,Agent A member => SIP/1CA3F2,0,Agent B member => SIP/E08972,0,Agent C And this dialplan (extension.ael) : 3600 => {

I have something weird happening and maybe I just need another pair of eyes to see it. I have been using the queue app from CVS 06/06/2003 for a couple of months not without a problem. Every time I upgrade (Which was difficult because of my own errors), my customers get put in the queue just fine. My Agents log in just fine. But once they get connected, there is no sound. If I setup a simple

Hi, I setup a queue (number 4050) with one static agent (extension 4054). What I would like is that when someone calls the 4050 queue and there are neither "dynamic" agents logged in nor is the static agent 4054 "on-line" then the caller gets out of the queue and falls into another context (eg. voicemail or anything). Not "on-line" means that either the SIP

I'm having a problem getting my queue to function as it should. After 20 seconds or so, it should prompt the user with a message "thanks for holding..... press # to leave a message or stay on the line to continue holding". I set up the "context" in the queues.conf file, so if a user presses a digit, they should be able to leave. But I get a SIP BUSY message. Here are my

Hello Asterisk Users, I guess the subject says the most of it; here goes some more detail: - Running Asterisk 1.2.14 - Objective: record all calls managed by a specific queue - Name those files ${TIMESTAMP}-${CALLERIDNUM}-${UNIQUEID} Facts: - If the UNIQUEID chan var is used in the MONITOR_FILENAME, before calling the Queue() application, the two legs of the call are not

Here is some dialog from the Console: -- Starting simple switch on 'Zap/13-1' Feb 10 07:22:36 NOTICE[21105]: chan_zap.c:6063 ss_thread: Got event 18 (Ring Begin)... -- Executing Goto("Zap/13-1", "mainmenu|s|1") in new stack -- Goto (mainmenu,s,1) -- Executing BackGround("Zap/13-1", "thank-you-for-calling-poker -support") in new stack

Hi, I have a problem with Queue where by a call comes in to the queue and if all the phones are busy and the queue reaches the timeout, it will fork a process and leave it sitting there before going off to the next step in the dial plan and continuing normally. This doesn't cause any problems except for I assume that it will eventually use up all the memory on the machine and it messes with

----- Original Message ----- > From: "John Kiniston" <johnkiniston at gmail.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> > Sent: Tuesday, July 28, 2015 12:12:05 PM > Subject: Re: [asterisk-users] Queues don't follow dialplan if no members are registered > > In your queues.conf do you