search for: mixmonitored

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]

Has anyone had any success using the MixMonitor() plus "command" as nothing I have tried works. I am using 1.2.1 I did google the archive but couldn't see any mention of anyone using this. What I am hoping to do is run a macro on hangup, current method I am using seems to miss some calls 5% of calls fail to mix / convert to mp3 etc. Was hoping that MixMonitor would fix this.

Hi, I am developing a call recording application on Asterisk 11.2 and have this configuration in my dialplan: [macro-ccdev2-rec] exten => s,1,MixMonitor(${ARG1},b) [outgoing-originate] exten => _X.,1,NoOp(Will send call to ${EXTEN}) exten => _X.,n,Dial(SIP/${EXTEN}@x.y.z) [outgoing-originate-rec] exten => h,1,Agi(agi://localhost/ajpbx.agi?path=uploadrec&callid=${CC_CALLID})

Hello Guys, I am trying to convert files that are .wac to mp3 after mixmonitor command is called but it doesnt execute the command, I tried the command in terminal it worked, any help please ... below is my dial plan exten=6500,n,Set(MIXMONITOR_EXEC=&& nice -n 19 /usr/local/bin/lame -b 8 -t -F -m m --bitwidth 8 --quiet "/var/spool/asterisk/monitor/${CALLFILENAME}.wav"

Hey, I've come across two interesting problems today. First, when recording long calls using Monitor(), it appears the in and out channels become out of sync. It seems like one channel happens faster or has data missing when sox mixes them together. Digging around, I found MixMonitor, which skips the whole soxmix process. I figured that removing that step could only help. Now it seems that

Is anyone can help me with this ? I'm really desperate. Thx in ad. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ikka - Mitra Kreasindo Sent: Wednesday, September 14, 2011 5:02 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Mixmonitor command parameter problem on

Hi all, I am setting up a script to record all the call. There are two app for recording. "Monitor" and "Mixmonitor", one mixing the audio on the fly and one mixing it at the end but also allow a option not to mixing the audio at all. If mixing the audio on the fly is not that taxing on the CPU, I would like to use 'mixmonitor' app. My question is, what is penalty on

Hello Asterisk list, Hope you are all doing well! We are using the MixMonitor application to record the calls and under some situations the call can be spied using ChanSpy with whisper enabled. Sometimes the spying channel is a person who can interact in the call, and some other times it is a sound file playing a message. The problem is that for some reason the MixMonitor does not record whatever

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

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

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 =>

Hi I'm using asterisk 1.8 on CentOS 5 I'm initiating call recordings with MixMonitor and trying to pause them with the features.conf. Whenever I try to pause the recording the call dies. Is PauseMonitor incompatible with MixMonitor? Here are some key log excerpts features reload == Parsing '/etc/asterisk/features.conf': == Found == Registered Feature

Hello, ChanSpy can not be used on a Channel that is being recorded with MixMonitor. How can I verify if a channel which I want to spy on, is currently not being recorded ?! Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120202/7954fe9e/attachment.htm>

hi, I am facing some issues while using MixMonitor. My extensions logic is attached below: exten => s,1,MixMonitor(${CALLERID(number)}-${TIMESTAMP}-${UNIQUEID}.gsm,b) in this extensions TIMESTAMP is not working in Asterisk 1.4. can any help me why TIMESTAMP is not working in Asterisk 1.4. regards, Asif

Hello, We are recording our calls to queues by putting the appropriate options in our "queue.conf". This is all working properly. We would now like to set the MixMonitor option to adjust the caller volume (which is very quiet). With the regular MixMonitor application, we would just add the "v4" option to make it much louder. I don't see a way to set this option when

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 =>

Hi, I am facing some issues while using MixMonitor and StopMonitor. My extensions logic is attached below: exten => s,1,MixMonitor(${CALLERID(num)}_${TIMESTAMP}.gsm,b) exten => s,2,Dial(SIP/101,13) exten => s,3,StopMonitor() exten => s,4,NoOp(Dial Status: ${DIALSTATUS}) exten => s,5,Goto(sss-${DIALSTATUS},1) exten => sss-NOANSWER,1,VoiceMail(777 at salesvoice) exten =>

Hi, sorry to bump this one but I still have this problem. The file is always created but is always zero size. This is the dial plan that records the call: exten = _0[1-8]X.,1,Set(CALLFILENAME=/var/spool/asterisk/callrecordings/its/${STRFTIME(${EPOCH},,%Y/%m/%d)}/Outbound-${UNIQUEID}) exten = _0[1-8]X.,2,MixMonitor(${CALLFILENAME}.gsm,b) The dial plan then calls a macro that makes the call. I?ve

Hello, I'm wondering if anyone else has been observing problems lately with 1.4.18 and higher releases of asterisk not properly recording calls. When using MixMonitor, the resulting file is only a few bytes long. I think this is because asterisk is doing Native bridging even though MixMonitor should block that. Did something change around the release of 1.4.18 that would have changed

Can't find an answer to this, but maybe I've not looked hard enough ... Does MixMonitor work without transcoding? ie. if I have a g729 stream passing through and I'm recording it with e.g. MixMonitor(/dump/filename.g729,b) and specify g729 in the filename, does MixMonitor transcode both legs of the stream to a format it can then "mix" then transcode it back to g729 to