similar to: set monitor_filename

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,

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 playing around with the QUEUE_WAITING_COUNT function but it always seems to return zero? I've tried everything. I suspect that this feature is not implemented in 1.2.7 which I am running.. Does anyone know in which version this function was added? Regards, Jan

Hello, Raw stats: Version:1.8.3.2 OS:Centos 5.6 Special setup: postgre database I am having a few queue issues with Asterisk specifically relating to breaking out from queues while on hold. The intent is that while someone is on hold they can press a key (lets say *) to break from the queue and go elsewhere (in this case to leave a message). However In all of my testing I am unable to get

Hi, Can somebody please shed some light on how to use the QUEUE_VARIABLES() function? The built-in help says ---cut--- Return Queue information in variables [Description] Makes the following queue variables available. QUEUEMAX maxmimum number of calls allowed QUEUESTRATEGY the strategy of the queue QUEUECALLS number of calls currently in the queue QUEUEHOLDTIME current average hold time

Hi, There are 3 new settings (setinterfacevar, setqueueentryvar, setqueuevar) and membermacro settings in 1.6 queues.conf. What is the potential use of these settings? The variables set are useful, but there is no indication of the purpose they could be used? Any one with some light on potential use case of these new features? raj

2009/12/4 Olivier <oza-4h07 at myamail.com> > Hello, > > Has someone successfully used this QUEUE_VARIABLES() function (in > 1.6.2-rc7) ? > I tried to use it as I'm using SIPPEER() but without success. > > A previous question about it remainded unanswered ( > http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/224466). > > Regards > How can

Is there an Asterisk Application/Function/Variable that returns the current number of callers in a given queue? Thanks, Doug

Hi all, I have configured asterisk and a2billing.for inbound i have also configured did and its forwarded to sip extensions. But i want to enable queues with inbound numbers(DID).But i could not find a way to do this in a2billing. I want enable that if some did comes to asterisk/a2billing it should be forwarded to queues not sip extensions and their i want to enable hunting so if one

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

We have 40 nortel ip2002 phones connected to asterisk 1.6, the problem I have is that the phone looses the connections with the server and then drops calls, we can reconnect but the customers don't like it. Anyone has the same problem? /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Sveav?gen 56C, 111 34 Stockholm, Sweden Dir: +46-(0)707548074, vxl:

Hi Im using a sip phone SPA921, and the one that calls me can hear me but I cant hear them, when I make the call I can hear them. Im running asterisk 1.6 behind a firewall, I have port 10000-20000 for rtp and 5060 for sip forward to my asterisk. Any tips? Regards /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Sveav?gen 56C, 111 34 Stockholm, Sweden Dir:

Hi When i try to transfer calls from my ip2002 phone in asterisk 1.6, I can make the transfer and it rings on the extension I transfer to, but when I accept the call, asterisk dumps. How can I get it to work? And how do I save the dump error? Regards /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir:

Hi I want to be able to use one provider if I dial 0 before the number and another if I dial 1 before, how can I do that in asterisk 1.6? /ralf Ralf Tr?skman, IT AdLibris AB, Box 3667, 103 59 Stockholm. Bes?ksadress: Sveav?gen 56C, 111 34, Stockholm - Obs ny address! Dir: +46-(0)8-5460 60 91, mob: +46-(0)70-7548074, vxl: +46-(0)8-5460 60 00, fax: +46-(0)8-5460 60 99 ralf at

Hi Is there a user client that a group, like customer service can use? We have today an avaya IP-office with phonemanager pro and I want something equal to phonemanager pro, where you can logon to ques and see how many calls is in that queue and so on. Regards /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir:

Hi I have a grandstream gxp-2000 and trying it on an asterisk 1.6. When I call internally between extensions I can hear the other person in the gxp2000, but when I call externally from the gxp I can't hear the person on the other end, but he can hear me. How do you configure the grandstream 2000 to work on asterisk 1.6? Regards /ralf ________________________________________________ Ralf

Hi I have asterisk 1.6 and running queues with realtime mysql. I am trying to set another musiconhold then "default" but I cant get it to work, I have an musiconhold entry in my queue_table, but don't know what to put in there and where to put the file. Regards /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm,

Hi [Feb 17 07:59:45] WARNING[21539]: channel.c:3477 ast_request: No channel type registered for 'USTM' [Feb 17 07:59:45] WARNING[21539]: app_dial.c:1502 dial_exec_full: Unable to create channel of type 'USTM' (cause 66 - Channel not implemented) == Everyone is busy/congested at this time (1:0/0/1) I get this after I restart my asterisk 1.6, it all worked yesterday. I have the

Hi To dial an outside line i have to dial 0. I want to have that when we dial local numbers, that is we are in the 08 area, I don't want to have to dial 08, how to set this up in asterisk 1.6? Regards /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460

Hi Our sip provider has two servers that sends calls to our asterisk 1.6. When server 1 sends call everything is working, but when server 2 sends call I get [Jan 13 14:56:23] NOTICE[16680]: chan_sip.c:16869 handle_request_invite: Call from '' to extension '0840303390' rejected because extension not found. And the provider get an "404 not found" error on their side. What