similar to: Using MixMonitor()

Hello List, i need to be able to record the call transferred from iax extension to sip extension when i call the sip extension from the IAX extension i can record the call without any issue but when i receive a call from customer in IAX and i transfer this call to SIP client the conversation between customer and IAX client is recorded but the conversation between customer and sip extension is

Dan et al; This looks like a perfect solution. However, I have one issue. If I initiate the macro manually (put it in the proper context/dialplan) it works. I see the *.wav file being created and growing in the /var/spool/asterisk/monitor directory. If I try to implement it adding the MixMonApp => *1,self/both,Macro,mixmon to the [applicationmap] in features.conf, I cannot get it to

Hello Everyone; I am looking for a solution to record calls that come into our Asterisk server. I am hoping for something that is easy to use - however, if I have to modify it to make it easier to use, I do not mind. Does anyone know of any opensource or otherwise solutions out there that I can try out? Thanks much. Glen

Hi I'm using asterisk 1.8.3.2 (with a couple of patches) I have the following scenario... SIP call comes in and gets answered by extension A (MixMonitor is executed as part of this inbound dial plan of the number being called) Extension A puts call on hold and calls extension B Extension A then does an attended transfer of incoming call to extension B I'm finding that the recording

I have a customer that needs to record all calls coming in and out. The problem I am having is when a call comes in to the operator and it is transferred to another extension. The first mixmonitor begins recording when the operator picks up but the recording stops when the call is transferred. I need to have a single recording for the complete call no matter how many times it is transferred.

Hello list, I need to record all calls. So I'm using application Monitor. Works good until someone transfers a callee to another internal extension. Example: A calls B A set B on hold A calls C A transfers B to C with SIP transfer (SIP REFER - with phone funktions and not Asterisk attended transfer). I found http://bugs.digium.com/view.php?id=0013538 . "corruptor" asked about this

I am trying to record calls on demand both inbound and outbound calls. I can record outbound calls just fine but not inbound calls or calls from an internally between extensions. I am using the latest asterisk 1.8.x certified version. On an outbound call I see: == Using SIP RTP CoS mark 5 -- Called SIP/ BVTrunk /7190000000 -- SIP/BVTrunk-00000163 is making progress passing it to

I have a Asterisk PBX System with Redhat Linux Fedora 4, Webmin version 1.400 and I am simply trying to configure into the "Extensions.conf" script an entry that will add to the "Auto-Attendant" a line that will allow a "Caller" to enter a "0" (Zero) will then ring the extension(s) of the "Operator" to speak directly with the "OPERATOR"

cdr_mysql doesn't set the userfield when it's set inside a macro called from a feature (1.4.25, addons 1.4.8). I have a feature code: autorecord => *1,self,Macro,apprecord The apprecord macro looks like: [macro-apprecord] exten => s,1,Playback(beep) exten =>

Hello, I thought to post this here before my manager starts his own coding project to give us a workaround. My situation I'm running into is as follows: 1. A call comes into our Asterisk system, it's trunked from one office to another via IAX. 2. Call enters a queue and is picked up by one of the agents. 3. That agent has to transfer the call, could be for a number of reasons the client

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]

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 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

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 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

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'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

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

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 =>

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