similar to: Monitor, MixMonitor and volume levels

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

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

Hello! I'm using Asterisk for our office telephony, but we have some problems that still we can't resolve about it. Here they are: 1) merge in/out call recording files I also tried to use a script I found on the internet, called 2wav2mp3 In extensions.conf I added the following lines ; script to be executed when monitoring has been finished MONITOR_EXEC=/usr/local/bin/2wav2mp3 exten

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, 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]

Is there an equivalent stopmonitor command if you are using MixMonitor ? StopMonitor does not seem to have an effect on MixMonitor Julian.

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

Hi! Since we upgraded to 1.6.1.11, asterisk is only outputing monitored files -in and -out. It is not mixing them in the end. queues.conf has monitor-type=MixMonitor... Would somebody help me debug why it doesn't mix the sounds?? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

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 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

A week or so ago, I explained that we need to "blank" our call recording when some sensitive information like credit cards where being discussed. With the lists help, I managed to find the pause/ unpause monitor commands. That works great. However (there is always a however), what that now means is that the length of the call does not match the length of the call recording, so adding

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. I notice that when a call that is recorded with MixMonitor is transfered to another co-worker, the recording ends. exten => 409,n,Macro(SDstartrecording,external,${DID}) the incoming call then goes to a queue... [macro-startrecording] ; ARG1 = incoming DID or CALLERID(name) ; ARG2 = outgoing dialnumber ... exten => s,n,MixMonitor(/var/ftp/${NR}/${recordfile},b,chown -R

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.

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

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 =>

I have an application that requires I be able to dial into an asterisk box, then from there dial out to another user through a PSTN. I'd like to be able to both 1) record this call and 2) let another user dial in using something like ChanSpy to listen to the conversation. I can get this working by executing an auto-dial script to connect one end of a call to an outside Asterisk box which

Dne 27.1.2016 v 13:14 A J Stiles napsal(a): > On Wednesday 27 Jan 2016, Marek ?ervenka wrote: >> hi, >> >> i have strange problem with asterisk 13 mixmonitor, recording to wav >> (centos6) >> when the system is under load, there are sometimes missing syllable >> >> there arent BIG spikes on cpus >> recordings are to ramdisk (/dev/shm) >>

Dne 27.1.2016 v 17:50 A J Stiles napsal(a): > On Wednesday 27 Jan 2016, Marek ?ervenka wrote: >> Dne 27.1.2016 v 13:14 A J Stiles napsal(a): >>> On Wednesday 27 Jan 2016, Marek ?ervenka wrote: >>>> hi, >>>> >>>> i have strange problem with asterisk 13 mixmonitor, recording to wav >>>> (centos6) >>>> when the system is

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