similar to: asterisk v1.6 monitor_exec

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

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

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 planning to upgrade an old public/internal development infrastructure and will use CentOS 5.4 x86_64 as basis. One of the critical server is Subversion (as an Apache httpd module). We currently use Subversion v1.4 on the server and v1.6 on our clients. While I was very happy with the v1.4 server over the years, we now sometimes have weird issues which seem related to compatibility

Hi all, I am trying to isntall the v1.6 version of Asterisk on my Intrepid system, but I get an error after I have typed make: [CC] manager.c -> manager.o manager.c: In function ?action_getvar?: manager.c:1732: error: ?SENTINEL? undeclared (first use in this function) manager.c:1732: error: (Each undeclared identifier is reported only once manager.c:1732: error: for each function it appears

Optimize opus decode (float only) use case using ARM NE10. Mainly effects opus_ifft and ctl_mdct_backward and related functions. Work based on previous Encode optimization using ARM NE10 library. TBD: Add commit id of upstream Encode NE10 optimization patch so that users have reference of how to enable this optimization Signed-off-by: Viswanath Puttagunta <viswanath.puttagunta at

Optimize opus decode (float only) use case using ARM NE10. Mainly effects opus_ifft and ctl_mdct_backward and related functions. Work based on previous Encode optimization using ARM NE10 library. TBD: Add commit id of upstream Encode NE10 optimization patch so that users have reference of how to enable this optimization Signed-off-by: Viswanath Puttagunta <viswanath.puttagunta at

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 am using Asterisk 11.17.1 with my program that uses AMI Originate calls to generate a bunch of calls for a callcenter. The PBX configuration is handled by FreePBX 2.11. I want to understand the dialplan behavior in order to figure out why the CDR(recordingfile) is blank on the CDR records despite the dialplan setting it. My program generates the calls by setting Channel=Local/NUMBERTODIAL at

I am trying to record a call into a stereo mp3 in Asterisk 1.4, but I can't seem to get it to work correct. Could someone point me to what I need to do? I have attached what I believe are the relevant parts. [globals] ; script to be executed when monitoring has been finished MONITOR_EXEC=/usr/local/bin/2wav2mp3 ; uncomment this line if you are using Ogg Vorbis

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

I want to set up a conference call to be recorded automatically, so I'd like the recording to start when the second caller joins the conference (one caller already there). The recording would continue until the last user hangs up. How can you determine how many are already in the conference bridge? [conferences] exten => 66,1,Answer exten => 66,n,Wait(1) exten =>

Was using this monitor line to get soxmix to mix test-in.wav and test- out.wav into test.wav. exten => 1200,1,Monitor(wav|/tmp/test|m) When I start the conference, the * console shows this: monitor executing ( nice -n 19 soxmix "//tmp/test-in.wav" "//tmp/test- out.wav" "//tmp/test.wav" && rm -f "//tmp/test-"* ) & /tmp shows test-in.wav,

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 using ConfBridge within Asterisk 1.6.20 and want to record the conference, so I'd like to start the recording when the second user joins, so in the example below, for example, how can I get the current user count in ConfBridge 3000? [conferences] ;authenticated conference (ext C-O-N-F = 2663) exten => 2663,1,Answer same => n,Wait(1) same => n,Authenticate(143382)

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