similar to: call queue problems

Hi everyone, i am face with an asterisk use management interface, at the pressent, i am using AMP (asterisk Management Portal: http://coalescentsystems.ca/index.php?option=com_content&task=view&id=31&Itemid=57 ). Does anyone know a better and more documented management interface for * ? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

An application that run without flaw last week has been reported to crash with an error that says that a conversion from char to datetime has given a date outside range. This very application runs OK today in production (fortunately :-), and the difference in both SQL Servers is just that the one where the exception has started to appear has Spanish conventions. I can''t give

As we all know you cannot trust anything you receive from the internet. I am wondering what the correct RoR way is to solve this. For the model there are various validates_* methods that you can use to ensure some integrity of that part. That is good by itself. Though it would be nice if setting these would also result in enforcement in the database backend itself when supported. Here I am as

Hi all Does anybody know where this command is supposed to go? Set(MONITOR_EXEC=mv /var/spool/asterisk/monitor/^{MONITOR_FILENAME} /tmp/^{MONITOR_FILENAME}) In the queues.conf file it talks about it. So I naturally thought after I set up my monitor with monitor-format = wav monitor-type = MixMonitor That I could put a lame command in there to convert and move the file elsewhere for backup with

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,

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.

Hey Matt I wonder if it is possible that it doesn't work with AEL, does this seem ok to you? s => { Ringing(); wait(2); Answer(); Set(MONITOR_EXEC=/etc/asterisk/lameconvert.php /var/spool/asterisk/monitor/^{MONITOR_FILENAME}); Queue(MyTestQ,ni,,,18); Hangup(); } I have debug

Hi I have this in my queue extension and I see this in asterisk when I call to the queue, but no file is created in the directory any ideas? exten => s,1,Set(MONITOR_FILENAME=/var/spool/asterisk/queuecalls/QSAMPLE-${UNIQUEID}) -- Executing [s at kundservice:1] Set("SIP/0850001175-b7942770", "MONITOR_FILENAME=/var/spool/asterisk/queuecalls/QSAMPLE-1228382046.12") Regards

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

Hi All; I noticed in the queues.conf the configuration for recording the calls in the queuing, and regarding to the filename (or any other parameter), it is written that I can determine the filename using the command: Set(MONITOR_FILENAME=foo) But it should be called from the dialing plan, but really i did not understand how to call it from the dialing plan. Well, for example this is my

In my * box I've configured two queues and incoming number and whenever any one calls those number call comes to my *box and it sends call to my agents in queue. but if no agent is available it still answer the call. Is there any why when my agents are not available I don't want call to get answered. Here is my dialplan: exten => xxxx,1,GotoIfTime(*|*|20|dec?ccagents,xxxx,6) exten

Hi all, I'm new in the list, and I have a problem upgrading from asterisk 1.2 to asterisk 1.4: There is a diference from asterisk1.2 to asterisk1.4 in AMI events. When I do a call to a queue (with the same extensions.conf dial plan) with ast1.2 and ast1.4, in ast1.2 apper 3 newcallerid event in ast1.4 apper only 2. It is normal? anyone knows it? what is the reason? I

Hey All ~ 1, queue records on fairly unreliable. I would say about 40 - 60 percent of the queue calls are not being recorded and I'm not sure why. I don't seem to see any kind of pattern to the failure. I've added a sample of our queue config at the bottom. 2, cel_pgsql module seems to crash regularly. It seems every time I look at our asterisk server, the cel_pgsql module is

Hi, I've an PRI coming to my asterisk ,calls are coming fine and my agents are able to answer no prob. but I've an agreement with my telco with some incoming no if the no of calls on these no are more then 3 then send to another no. they use busy signal to divert call on another number so I'm sending the call to Congestion() if no of calls in this group are more then 3. But my

Hi, Since Carl has kindly provided us with fax support for CAPI based cards, we have been using it with much success. Today I have modified app_capiFax so that it now supports a dynamic CSID. The following example uses the DNID created by chan_capi on an AVM Fritz! card. * Receive a fax with CAPI API. * Usage : capiAnswerFax2(path_output_file.SFF|stationID) * * This function can be

Hello. I'm having Asterisk 1.6.0.x and trying to solve the issue concerning with a bad quality of voice for incoming SIP calls into the app_meetme. As I know, in my case of calls, jitter buffer is NOT executed on anyone channel. So, after reading Russell Bryant's post ( http://www.russellbryant.net/blog/2007/10/09/asterisk-jitterbuffer-support-for-applications/) I added following scheme

On systems even key systems it is customary to have an 'internal' dial tone. Since Asterisk simply ignores the 9 and keeps the tone going it is hard to tell for some 'new users' if they can make a call. My first idea was to change the generated dial tone via source. Then if the user presses 9 go to a different context where I would record about 30 seconds of the normal dial

Guys, if you define recording on queues.conf and also define a monitor_filename var on your dialplna, you can record a queue call but, isthere a way to do something with the file after the call ends? I need to move the file to some other place but I cant find where to define a command to run after a queue call finishes. Any hints?

Hi, I'm sure some else has been facing this problem. I want to record all the call coming in my queue. I want this format: YYYYMMDD-HHMMSS-AgentID-CallerId - UniqueID. I'm using the monitor feature inside the queue.conf. I can't use the agents.conf monitor features because I'm using dynamic agent (addqueuemember) The problem I'm facing is that I can

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