similar to: Channel name in queue log replaced by a manager event?

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

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}" !=

Hi, I'm putting together a system to manage agents with Realtime, and without chan_agent. In 1.2.13, there's a handy (although marked as deprecated in apps/Makefile) PGSQL application to let me do this: macro queue-addremove(queuename,penalty) { switch(${MACRO_EXTEN:0:1}) { case I: // Login PGSQL(Connect connid host=XXX user=XXX password=XXX

I am experiencing a change in behaviour of my Queues in 1.4.9 vs 1.4.8. I do not pass the 'n' option to any call to Queue() in my dialplan. Yet since I upgraded to 1.4.9, I have occasionally seen this on my console: -- Nobody picked up in 20000 ms -- Exiting on time-out cycle That log message "Exiting on time-out cycle" is exclusive to the logic in app_queue meant to

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

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

Hi all, I've been working on this for days and can't find a solution. I need to use AddQueueMember for my agent logins to my Queues -- but a number of my agents are outside the main server, which is connected to my asterisk network over IAX2. I can't just do a AddQueueMember(queuename) because it puts in a complicated member calleridnum like: IAX2/peername:65723/23 Which won't

>From voip-info.org: Queue(queuename|options|optionalurl|announceoverride|timeout) 'optionalurl' allows you to send a URL to devices that support it. Does anyone have details on the "devices" that support the optionalurl method of the Queue application? I am wondering if there is a softphone that supports this. The only thing that seems to happen is the

Hi I've a queue which has generic zap extensions (of my legacy PBX which is connected to asterisk via cross over on span 4 ) logged in ..The legacy pbx extensions are able to logon to queue perfect.. but Whenever a call comes in queue the status of that extension in "queue show <queuename>" always shows as "NOT IN USE" instead of ringing or In use as shown in a SIP

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

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,

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

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

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.

On Wed, Jan 28, 2015 at 12:23 PM, Ishfaq Malik <ish at pack-net.co.uk> wrote: > 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

Hi, following situation: 1) You create a raw printer (called "testprinter") under cups and leave the description-field (info-field) empty. cups will create an entry in /etc/printcap of the form testprinter|testprinter:rm=linux100:rp=testprinter: 2) You start samba with a standard [printers] and [print$] share. samba will expose the printqueue to the clients. 3) You upload a windows

I just installed 1.4.23.1 with the queue realtime logger backport. Here are my configs: musiconhold.conf [default] mode=files directory=/var/lib/asterisk/moh-native random=yes queues.conf [7703] wrapuptime=0 timeout=15 strategy=rrmemory retry=5 queue-youarenext=queue-youarenext queue-thereare=queue-thereare queue-thankyou=queue-thankyou queue-callswaiting=queue-callswaiting

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

I have a couple of real quick questions for ya'll. I am trying to setup * in a test scenario to emulate my company's 20 agent ACD. 1) First, I am curious if anyone has any configs that they can throw up on the wiki or just send via e-mail. I have a bit of it pieced together and am able to make some very basic queued calls, but I am sure that there are much smarter people out there

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,