similar to: How to check if a local channel member of a queue?

Hi, I have a queue with one agent added using AddQueueMember (FAO|Local/1001 at from-sip|0||Agent/602). My extensions.conf is [general] static=yes writeprotect=yes autofallthrough=no clearglobalvars=no priorityjumping=no [from-sip] exten => 100001000,1,Dial(SIP/100001000,,t) exten => 1001,1,Dial(SIP/1001,,t) exten => 1002,1,Dial(SIP/1002,,t) exten => 1003,1,Dial(SIP/1003,,t) exten

Hi, I am using asterisk-1.4.15, My sip configs is like [2501] type=friend username=2501 secret=2501 canreinvite=no host=dynamic dtmfmode=rfc2833 context = sip disallow=all allow=ulaw incominglimit=1 nat=1 queue.conf is like [gen-enq] joinempty = yes musiconhold = default strategy = rrmemory servicelevel = 60 timeout = 60 retry = 5 wrapuptime=5 announce-frequency = 90 announce-holdtime = yes

Following on from a posting yesterday from Kevin, I have the following in the dialplan: exten => 709,1,AddQueueMember(SomeQueue|Local/706@AgentQ) I am on extension 706. From the CLI: SomeQueue has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime), W:0, C:0, A:3, SL:0.0% within 60s No Members No Callers I call 709, get a console message

Paul wrote > >;Pause/unpause Queue >exten => 424,1,PauseQueueMember(|SIP/${CALLERID(num)}) >exten => 424,2,Playback(unavailable) >exten => 424,3,Hangup >exten => 425,1,UnPauseQueueMember(|SIP/${CALLERID(num)}) >exten => 425,2,Playback(available) >exten => 425,3,Hangup > Following your suggestion and a number of postings and articles I have

Hi, Can some one tell me what is the meaning of all the fields of show queue callcenter? for example in my system it gives: callcenter has 0 calls (max unlimited) in 'roundrobin' strategy (33s holdtime), C:429, A:12, SL:0.0% within 0s How is the holdtime calculated? what is A and SL? Also how can I see which of my zap interfaces are busy currently? I did a zap show channels I get

Hi, I am using asterisk-1.4.15, and using AddQueueMember to add SIP interface to the queue. Each sip interface is member of multiple queues The queue does not recognize that an agent is busy and keeps trying to call the busy agent. I have identified two patches that can fix the problem, one at http://www.scopserv.com/download/asterisk-1.4.17-state_interface.diff in thread

Hello, I add a member to a queue with AddQueueMember, but the Queue still indicates "joinempty" : Add member to queue : /-- Executing [queueadd at sub-GetParams:2] AddQueueMember("SIP/sip17-00005c1e", "myqueue11,member3") in new stack -- Executing [queueadd at sub-GetParams:3] NoOp("SIP/sip17-00005c1e", "AQMSTATUS = ADDED") in new stack/ ...

Hello everyone. I got a little problem in here: I want to set up a queue so that if anything of these happens: a) No agents logged in b) All agents busy Then the user gets diverted somewhere. I used this (for testing purposes only, of course): exten => 7080,1,Answer() exten => 7080,n,Queue(teste) exten => 7080,n,Goto(${QUEUESTATUS}) exten => 7080,n(ERROR),NoOp(${QUEUESTATUS}) exten

Hi, I'm currently migrating to 1.4 and have problems changing deprecated AgentCallbackLogin to AddQueueMember. I have dynamic queues and dynamic agents (MySQL Realtime), and pseudo-dynamic agents.conf (with huge amount of possible agent numbers). Agent login is done trough manager API: * AgentCallbackLogin * QueueAdd In 1.4 seems AddQueueMember can do all the same, but there is no such

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

Queue not sending call to Agent I am having an issue and i am not sure if it is a bug or a config issue. I was originally running Asterisk 1.8.1.1 when I noticed this issue. I upgraded to 1.8.4.2 to see if that would fix it but it didn't. The issue is that I have a call queue and the agent dials a number to log into the queue. When someone calls the queue the first time the call is

I am trying to set up Hunt Groups and I am having some issues. Here is what I am trying to do. All my users actually register with OpenSIPS. Asterisk is using Realtime and I have set up a MySQL View Table so that Asterisk see's all the SIP users info that OpenSIPS has. This is what I have configured queues.conf ---------------------------------- [irock.com] strategy=leastrecent

I have set up an operator queue for our receptionist. That way, if she takes a break or is out, by logging out of the queue, calls to the "Operator" can be handled by other agents. I have set strategy = roundrobin in queues.conf. According to "the book" ATFoT, roundrobin always starts with the first agent in the queue. This is the desired result. I want all calls to start

Hi, I have a typical call center with queues and agents added via AddQueueMember. One of my requirement is to implement a forgot password function. If a caller does not remember the password, he calls up an unauthenticated line and the agent manually authenticates him. Then the caller should have a provision to reset his password. The requirement is that the agent should not know the new password

I am having a problem with the Call Queue Feature. If I let a user into the Queue prior to an agent being available for them to take the call, they experience the following: 1) they hear that they are the first in line 2) when the agent finally logs in (the caller on hold in the queue is sent to the users phone) 3) the AGENT is still in the login phase hearing that they are "successfully

I have a queue setup to 'roundrobin' (NOT roundrobin with memory). I have three agents. We'll call them 101, 102, and 103. When a call comes in.. I want it to always try 101 if no answer try 102.. if no answer try 103, etc. However, what it is doing is... it will ring 101... if 101 answers, next time a call comes in it will go to 102. This isn't at all what I want. Any ideas

I am running r79979 of Asterisk Trunk, and I am having problems trying to use app_queue.so. I want to use the extension 510 to be a line where users can call technical support. Extensions 511 and 512 are used by the operators to dynamically make themselves a Queue Member or not. So, operators call 511, and they should get added to the Queue as a Queue member. When users call 510 then, it

Hi, ZAP/DAHDI extension 3210 calls an Asterisk queue 4050 with one SIP agent 4053 added via ->QueueAdd("4050", "Local/4053 at from-internal/n", 1) (not via agents.conf). SIP extension 4053 rings, answers and then decides to blind-transfer to ZAP/DAHDI extension 3666. The "show queue" command still displays 4053 as "In use". However, if 3210 calls 4050

Hi, Is it possible to define maximum talk time in a queue? ie any one who joins a queue should not be able to talk more than say 5 minutes to the agent. raj

Hi, I am running a 5 seater inbound call center on 1.0.9-BRIstuffed-0.2.0-RC8h and it's running well. I am now trying to upgrade it to 1.2.4. So I installed 1.2.4 from source and copied all config files from original to the new server. But when a caller lands inside the queue no queue message is getting played. The gsm files are present in proper locations, whcih I am able to play using