similar to: MixMonitor Files Always Empty

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

Hi All I have a problem with mixmonitor in 13.4.0 doing the following: 1. Caller phones in 2. Reception picks up 3. Talks to caller 4. Does attended transfer, talks to manager to screen the caller wanting to speak to him 5. Complete the transfer by putting down her handset so the caller can speak to the manager 6. Caller talks to the manager The problem is that mixmonitor only records

Hi all Asterisk 1.8.11.0 on Centos 6.5 My VOIP phones are using G729 to a G729 trunk from a vendor (Centracom, South Africa). Unlicensed G729 codec version on server. 75% of my .gsm files from MixMonitor are coming up corrupted about 3 minutes into the recording. The server has been up for 7 months beforehand with no problems with recordings to .gsm format files. I noted

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 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! 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

On 1/8/18 9:38 AM, Bertrand LUPART - Linkeo.com wrote: > Hello Carlos, > > >> We have a server that records all calls so we set Mixmonitor with the b option to only record calls that are actually bridged. I notice that we have lost of 44 byte files in /var/spool/asterisk/monitor which correspond to calls that were not answered. If a call is not answered I assume it was never

Hi, I'm noticing MixMonitor records 5 seconds aprox less of a call. The recording is iniciated via Queue and ends at the hungup. (gsm format), when I listen to the audio file, has 5 seconds missing at the end of the call. Any idea?? thanks ASt.1.6.0.1 -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have started using the call recording facilities in Asterisk 1.2 recently, and having worked out some of the foibles regarding call forwarding etc etc, I think I have a mostly working system. I do still seem to have a problem with recording volume though. It seems that all SIP call legs are recorded at "normal" volume, but all my Zap (ISDN) and IAX (via Provider -> ISDN) calls

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]

Hi, I'm noticing MixMonitor records 5 seconds aprox less of a call. The recording is iniciated via Queue and ends at the hungup. (gsm format), when I listen to the audio file, has 5 seconds missing at the end of the call. Any idea?? thanks ASt.1.6.0.1 -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm noticing MixMonitor records 5 seconds aprox less of a call. The recording is iniciated via Queue and ends at the hungup. (gsm format), when I listen to the audio file, has 5 seconds missing at the end of the call. Any idea?? thanks ASt.1.6.0.1 -------------- next part -------------- An HTML attachment was scrubbed... URL:

We have a server that records all calls so we set Mixmonitor with the b option to only record calls that are actually bridged. I notice that we have lost of 44 byte files in /var/spool/asterisk/monitor which correspond to calls that were not answered. If a call is not answered I assume it was never bridged so why would Asterisk create a file? Is there a way to avoid getting those empty

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, which file extensios are supported in mixmonitor application? https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Application_MixMonitor can i record to Opus? -- --------------------------------------- Marek Cervenka =======================================

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

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

I'm using asterisk 1.8 as an answering machine. I'd like to hear the calls it answers aloud in case I want to pick up and interrupt the call. There are a few articles describing, for example, three-way calling a monitor phone set to auto-answer, but I couldn't find anything that described how to just send the audio to a local speaker. I am currently using MixMonitor to append the

Hello, I'm trying to set up call recording for a queue. Right now the recording appears to work correctly, but when I call and chatter for a minute or so, at the end of the call I end up with a very small file (less than 100 bytes), which contains about .06 seconds of silence. If I talk for another minute, this file will get up to 200 bytes or so. In my queue configuration, I have: [testq]