similar to: automixmon output file location and exec command options

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

Hello, I have a recording started in the dialplan with the MixMonitor application. I want to be able to stop it during a call and maybe restart it. I tried using the value defined in [featuremap] but it starts another MixMonitor application even if there already one instead of stopping it. Any idea on how I can stop the MixMonitor application while it is running? [featuremap] automixmon =>

MixMonitor seems to work: -- User hit '*3' to record call. filename: auto-1250792853-24-22 == Begin MixMonitor Recording SIP/snom2-084c4ec8 /var/spool/asterisk/monitor/auto-1250792853-24-22.wav exists now. Recording a call without mixing fails. > User hit '*1' to record call. filename: wav,auto-1250793354-24-22,m TOUCH_MONITOR_OUTPUT is set to

Hi all, I am having the following trouble with recording calls: When calls come into the support line did number, the call starts to record on the first queue, but appears to hang up when the call actually connects to the engineer (ie I see "got hangup request" on the cli and then mixmonitor ends.) I am guessing this has to do with the announce file that is played to the engineer

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 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

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

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

Hi folks, I was using the following featuremap: blindxfer => *1 disconnect => *9 atxfer => *2 parkcall => *7 automixmon => *0 and everything worked. Then the need arouse to use some features like automixmon during a conference, but MeetMet has the * key bound to the (admin) menu. Thus, in order to enable features like automon and transfers even during a conference, I

Hi, I have a simple Queue(named 1) and one Member(SIP/1119) logged into it. Now when a caller is placed into Queue and gets connected with Member, I want to record the call. It does record the call when I use MixMonitor() before placing the caller into Queue, but not when MixMonitor() is used in macro which is called upon Member answering the call. Following is my dialplan... [mixmonitortest]

Greetings, The software G.729 codec module from Digium has been updated for all platforms. There are x86_32 and x86_64 versions optimized for specific processors available for both Asterisk 1.6 and 1.4 for the following platforms. * Linux * Solaris 10 * FreeBSD 7.0 * FreeBSD 6.1 Changes: * For Asterisk trunk / 1.6, builds have been updated for CLI API changes. * All non-Linux

Greetings, The software G.729 codec module from Digium has been updated for all platforms. There are x86_32 and x86_64 versions optimized for specific processors available for both Asterisk 1.6 and 1.4 for the following platforms. * Linux * Solaris 10 * FreeBSD 7.0 * FreeBSD 6.1 Changes: * For Asterisk trunk / 1.6, builds have been updated for CLI API changes. * All non-Linux

Greetings, all. I would like to record calls that are entered into queues and I'm not quite sure how to do it. Here's how I'm currently set up: - Call comes in and is placed into Queue #1 (which rings all phones for 15 sec). - If call drops out of this queue, it is placed into Queue #2 (which plays MoH until the call is picked up). I've tinkered with MixMonitor and I have my

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,

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

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 list! I'm trying to configure Asterisk to record incoming calls, if the called press *3. I added in features.conf: automixmon => *3 then, in my dialplan: exten => 1,n,Dial(SIP/00493511111111,20,RcxX) Well, if I **CALL** a number I'm able to record the call, but if I'll be called, and press *3 nothing happens... In the console I can't see anything, too. Could you

On 1 July 2016 at 17:41, Joshua Colp <jcolp at digium.com> wrote: > > >> exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>) >> same => n,Dial(PJSIP/phone123, 30) >> > > Your exten line has no priority, is that how it is in your dialplan? > Actually no, I stole that line from an earlier email to this list. Mine has a priority.

Vinicius Fontes wrote: > Hey guys, I don't know if this is the right place to ask this. I was > thinking about reporting a bug, but maybe it's better to sort out if > this is really a bug or just me being lame. > > I want to record *every* call in my Asterisk box, so I use the > MixMonitor() application like this is my extensions.conf: > > exten =>