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  • gineta, Wed 14 of May, 2008 [03:58 UTC]: any here not fine the configuration of firewall juniper -screem for VOIP asterisk????
  • Anoop Prabhakaran, Tue 13 of May, 2008 [12:16 UTC]: I am developing Asterisk IVR, Whenever i make a internation call to the IVR system, the DTMF is not getting detected properly, this happens only for the first time, second call onwards system works fine. why this is happening
  • joe, Mon 12 of May, 2008 [04:27 UTC]: Is there an opensource browser based softphone, or a system like Busta where everything is not manages through their website?
  • Nick Barnes, Fri 09 of May, 2008 [11:36 UTC]: Christopher - yesterday I tried an Asterisk install on a CentOS 5.1 box with stock GUI and it all worked fine. Sorry I can't help.
  • aero, Fri 09 of May, 2008 [08:20 UTC]: can someone help me out on this, i tried to play some sound files on my asterisk box and this is the error message i got. WARNING[4429]: format_wav.c:169 check_header: Unexpected freqency 22050 May 8 11:17:39 WARNING[4433]: codec_gsm.c:194 gsmtolin_fra
  • Christopher Faust, Thu 08 of May, 2008 [14:15 UTC]: I beleive that I may have to change something in the xserver configuration. Please advise
  • Christopher Faust, Thu 08 of May, 2008 [14:14 UTC]: Everything was perfect. In the bios I have increased the memory allocated Still receive input not supported on my display.
  • Christopher Faust, Thu 08 of May, 2008 [14:13 UTC]: This would not be my main box. I am doing some testing to see if I can install zaptel and asterisk 1.4 on a full centos 5.1 box with development software Its bizzare, because before I went through the asterisk and zaptel installation everything was perfe
  • Nick Barnes, Thu 08 of May, 2008 [13:44 UTC]: Christopher - I can't see any way in which an Asterisk installation would muck your GUI, but remember that it is advised not to use a GUI on an Asterisk box anyway.
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MixMonitor

MixMonitor

Record A Call Natively

Description

 MixMonitor(<file>.<ext>[|<options>[|<command>]])

Records The audio on the current channel to the specified file.

Valid Options:
 b - Only save audio to the file while the channel is bridged. *does not include conferences*
 a - Append to the file instead of overwriting it.
 v(<x>) - Adjust the heard volume by a factor of <x> -4/4.
 V(<x>) - Adjust the spoken volume by a factor of <x> -4/4.
 W(<x>) - Adjust the overall volume by a factor of <x> -4/4.

<command> will be executed when the recording is over. Any strings matching ^{X} will be unescaped to ${X} and all variables will be evaluated at that time. (NOTE by Netoguy: My experience shows that variables are actually evaluated when MixMonitor is Called and NOT when the recoring is completed and the command is run.)
The variable MixMonitor_FILENAME will be present as well. Where <command> is a system (Linux shell) command, see Asterisk cmd System for example values. Note do NOT include the dialplan command System(blah), just blah. If you don't specify a full path, the file will be stored in the "monitor" subdir of the path specified with astspooldir in asterisk.conf (so default will be /var/spool/asterisk/monitor).

Note that no environment variables are given to <command> — you must pass these on via command-line arguments.

The audio file is closed and processing of <command> is started *after* the 'h' extension priorities have been run.


Version

New in Asterisk 1.2

Details

This application is similar to the Monitor application only it's designed to record 1 audio and mix them natively as the call is in progress to avoid the need to spawn external processes which lead to harmful cpu usage spikes.

Benefits:
  • One call can record to mutiple files at the same time.
  • Allows for recording a call to a single g729 file
  • An append mode allows an agent to record all their calls in 1 file
  • A bridge flag allows recording to only take place when the channel is bridged.
  • The volume for either side of the channel may be adjusted seperatly.
  • a cli interface makes it possible to start and stop the monitoring at will from a manager session or the cli prompt.

See also



Asterisk | Applications | Functions | Variables | Expressions | Asterisk FAQ

Created by pupeno, Last modification by Keith Smith on Wed 30 of Jan, 2008 [22:44 UTC]

Comments Filter

Mixmonitor fails to record after attended transfer

by Martin Brett on Saturday 22 of March, 2008 [00:02:19 UTC]
For those who arrive attempting to work out why some inbound calls arent being recorded I would draw your attention to a known issue namely that due to an optimisation in chan_local Mixmonitor applied to calls subjected to an attended transfer will suspend recording at point of attended transfer.
Here's hoping that this will be rectified soon

Re: incomplete recording

by Manuel Cuya on Sunday 09 of March, 2008 [03:28:07 UTC]
I have applied MixMonitor to an outside prefix rule. Everything seems to be working properly, however when I check the sound files that are recorded, I notice they are not complete. Either the file size or sound duration seem arbitrary. While recording, there is nothing weird on the console, it says "Begin MixMonitor Recording (...)" and "End MixMonitor Recording (...)". I find the sound files broken somewhere between these two messages.

I am using .wav format and W(-2) option.

Any suggestions?


i have the same problem, anybody can help us ???

Re: Can you show me a sample...?!

by Geni kim on Wednesday 01 of August, 2007 [05:24:23 UTC]
I know it!!

Thank U~~?!

Re: Can you show me a sample...?!

by Geni kim on Tuesday 31 of July, 2007 [23:53:50 UTC]
I know it!!

Thank U~~?!

Can you show me a sample...?!

by Geni kim on Monday 30 of July, 2007 [01:20:06 UTC]
First..Sorry...

I can't understand Mixmonitor....

Please telle me Mixmonitor and easy.....^^


And I have a book that Asterisk 1.4

The book tell me a Using the mixmonitor application.

But there are no example to mixmonitor, and this web pages too!

So....Please Show me a Mixmonitor application examples!

Good luck to you~

incomplete recordings

by walter on Thursday 11 of May, 2006 [16:56:24 UTC]
make sure you are using a version of asterisk >= 1.2.7 so you have the fix

reference for this is bug is: http://bugs.digium.com/view.php?id=6457

Re: incomplete recording

by Gary Richardson on Thursday 02 of March, 2006 [23:49:41 UTC]
I'm experiencing the same problem with wav files cutting short. It seems that I can stop the recording by being loud. I've tried gsm and wav format.

automixmon?

by Jonathan Addleman on Wednesday 01 of March, 2006 [01:51:56 UTC]
Is there any way to do automon in a mixmonitor-like way?

MixMonitor casues asterisk 1.2.3 to terminate

by Bill Neely on Saturday 28 of January, 2006 [22:08:38 UTC]
I am testing MixMonitor on Asterisk 1.2.3 on FreeBSD 5.4

The following batch of extensions:

exten => 299,1,Answer()
exten => 299,n,MixMonitor(manuel.gsm)
exten => 299,n,SayDigits(12345670)
exten => 299,n,StopMonitor()
exten => 299,n,Hangup()

runs ok until hangup. At that point Asterisk terminates. (That is the process terminates with a signal 11)

incomplete recording

by Diogo Baeta on Tuesday 06 of December, 2005 [00:57:25 UTC]
Hi all,

I have applied MixMonitor to an outside prefix rule. Everything seems to be working properly, however when I check the sound files that are recorded, I notice they are not complete. Either the file size or sound duration seem arbitrary. While recording, there is nothing weird on the console, it says "Begin MixMonitor Recording (...)" and "End MixMonitor Recording (...)". I find the sound files broken somewhere between these two messages.

I am using .wav format and W(-2) option.

Any suggestions?

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