similar to: Realtime Queue Members Not Ringing

It's my understanding that the backport is available now in 1.4. However, seem to be having some issues with it. Just wondering if I have everything setup right. I'm running 1.4.26.2 realtime. queue_members: `uniqueid` int(10) unsigned NOT NULL auto_increment, `membername` varchar(40) default NULL, `queue_name` varchar(128) default NULL, `interface` varchar(128) default NULL,

Hi all, Someone know how can I check for available members on a queue Before I queue the call, so I can do something else with it? Note that is not the case for joinempty Thanks, Gabriel Ortiz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090718/462b725b/attachment.htm

I have some cases when I want to pause a queue member and automatically unpause the queue member after a specified time. Right now I am doing it by a AMI script, but was thinking if it is possible to add a parameter to PauseQueueMember like, PauseQueueMember([queuename],interface[,options[,reason[,time]]]) where time will be how long (in seconds) the interface will be paused. before brought back.

Imagine that you have this code: exten => _X!,n,Set(foo=${QUEUE_WAITING_COUNT(${QueueName})})) If ${QueueName} happens to be unset, this will cause a warning: [Jul 22 14:26:17] ERROR[8114]: app_queue.c:5187 queue_function_queuewaitingcount: QUEUE_WAITING_COUNT requires an argument: queuename The obvious solution: exten => _X!,n,ExecIf($["${QueueName}" !=

I'm looking for a function I can put in my dial plan that tells me if a device is a member of a queue, but I can seem to find one. Basically I want to be able to dial to join a queue and if I'm already on the queue, leave...... exten => 4,1,GotoIf(${is_queue_member(queuename,SIP/${ext})}?leave:join) exten => 4,n(leave),RemoveQueueMember(queuename,SIP/${ext}) exten => 4,n,Hangup

We love our Digium phones and DPMA - but we really need it to work on our Realtime Platform. Otherwise we lose all the cool features and they are just standard SIP phones. Anyone working on a solution for this? Or anyone from Digium see this on the roadmap?

Hi, Surely, I must be overlooking something. If I run the following SQL queries I don't get the same number of rows. Is this coherent? mysql> select * from queue_log where queuename = '4010' and FROM_UNIXTIME(time) between 20080308000000 and 20080313145900 group by callid; 357 rows in set (0.01 sec) mysql> select * from cdr where dst = 4010 and calldate between 20080308000000

Has anyone observed a problem where using Local channels with AddQueueMember results in missing TRANSFER events? Right now I'm using straight SIP channels when I call AddQueueMember(). I'm contemplating moving to Local channels because the non-state-based wrapuptime blows when you have a channel in multiple queues (they can hang up and get a call immediately so long as it's from a

To all who need more queue functionality, We are contracting Digium to enhance the queue app for our call center needs. Please read the following email conversation and give your ideas. Unless a glaring omission is found in my specification we will have them start tomorrow (Wednesday). I may not have thought of something important. It will be released to all Asterisk users by Digium. Thanks for

Hi We're using 1.8.23.1 on CentOS 5 and are trying to get accurate stats for queues. For a particular customer, when I run queue show <queue_name> I get the following numbers: <queue_name> has 0 calls (max unlimited) in 'ringall' strategy (17s holdtime, 94s talktime), W:0, C:175, A:44, SL:48.6% within 45s So from that data we look at 17s holdtime And assume that is the

hi,all i have just upgrade from 1.4.28 to 1.6.2.1. all works fine now except realtime queue. it seems queue_table works fine, but queue_member_queue not work, the two tables works fine when in 1.4.28. is that something changed related to realtime queue configuration? more detail about two table definition and data stored in , please see: http://pastebin.com/m33f9539e the extconfig.conf file,

Hi to all i'm using Asterisk 1.4 and need to announce something like 'The operator answering to you call is XXX' to the caller, is it possible to do that using an AGI script ? The syntax in Asterisk 1.4 is Queue(queuename[|options][|URL][|announceoverride][|timeout][|AGI]) So, setting up an appropriate AGI script can i play an audio file (or create it with some tts) to the

Hello list, what is the correct syntax ? exten => s,n,Queue(${queuename},,,,${timeout},cleanpickup.agi^${CHANNEL}) [Sep 13 10:23:58] WARNING[23551]: res_agi.c:886 launch_script: Failed to execute '/var/lib/asterisk/agi-bin/cleanpickup.agi^SIP/329909007906-0000017a': File does not exist. Kind regards, Jonas. -------------- next part -------------- An HTML attachment was

All, I am trying to set a monitor file from the queue.conf as specified on http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf In order to avoid the default MONITOR_FILENAME format wich is: "agent-xxxxx-uniqueid.wav" for example "agent-10017-1266438575-26.wav" As you may now, when using the queue command you are not able to know which agent will take the call,

Under 1.4.5 and 1.4.6, I've seen a few instances where the channel name in queue log entries is replaced by a snippet of a manager event: --START-- 1183582823|1183582823.104763|queuename|SIP/XXXX|REMOVEMEMBER| 1183582828|1183582793.104744|queuename| Context: macro-dialout Extension: s Priority: 3 Application: GotoIf AppData: 0?blockclid Uniqueid: 1183582822.104759

Hi everyone, For some reason MixMonitor doesn't record when it should; It actually shows the MixMonitor line just fine on the CLI. How can MixMonitor be debugged for things like privilege issues or filename issues? **I had this working at one point and then stopped working. Not sure what I changed. System Info: Asterisk 1.4.21.2 Queuemetrics 1.6.3.0 [queuedial] ; this piece of dialplan is

Hi, I use QUEUE_MEMBER_LIST and QUEUE_MEMBER_COUNT to automate Agent logoff at specified time. This works fine in normal case but today i noticed than two agents where still logged-in, but with a status of 'Invalid', and the script wasn't able to remove them since while QUEUE_MEMBER_LIST() correctly listed them, QUEUE_MEMBER_COUNT() gave 0 instead of 2. If this is expected

I'm running Asterisk 1.6.0.21 and Aastra 57i phones. I'm having an issue with the agent phones ringing when someone is in our queue. The first phone will ring 3 to 4 times then the call will roll over to the next phone as expected. However, any phone after the first one will only ring once and the wait between that phone and the next will be as if it if ringing 3 to 4 times. My

I'm curious if anyone has implemented the following: Need to setup an on-call queue, that activates after 5PM and de-activates at 8AM, also that activates/deactivates on demand(I'm thinking a feature code here). The "agents" need to log in via cell phones, and when calls come in from outside to the asterisk system, it'll need to call the cell phone agents that are active.

I am just studying Asterisk now and have a question. Is it possible to force anyone who enters a queue into voice mail after they have been in the queue for 30 seconds? /************************************************************** Ken Alker ken@impulse.net ham radio: KA6SDU Impulse Internet Services http://www.impulse.net Santa Barbara, San Luis Obispo,