login | register
Wed 01 of Aug, 2007 [09:13 UTC]

voip-info.org

Search with Google
Search this site with Google. Results may not include recent changes.

Web www.voip-info.org
Shoutbox
  • Aykut, Wed 01 of Aug, 2007 [07:53 UTC]: Hi all, does anybody know about Thomson ST2030 SIP phone. I have upgraded it to latest version (1.56) but "Hold" and "Conf" features are not working after the upgrade ?? Do you know any solution or do you have Ver. 1.52 ?? Where can I find it?
  • Edward J Brown, Tue 31 of Jul, 2007 [23:33 UTC]: Has anybody experienced Choppy voice quality when using a Linksys SPA942 in an Asterisk Conference bridge? It works fine with my polycom and Cisco, but sucks with my Linksys.
  • www.astawerks.com, Fri 27 of Jul, 2007 [18:00 UTC]: does anyone use asterisk on top of clark connect? does it work good?
  • simon, Fri 27 of Jul, 2007 [14:16 UTC]: Hi All, Has anyone here managed to get the Cisco79x1 to successfully fail over to the backup proxy. I have 2 asterisk servers , handsets all register and function, except that backup proxy function doesn't work. Any working example would be very apprecia
  • Matthew Richmond, Thu 26 of Jul, 2007 [03:40 UTC]: using the page() application to page across our building...often the meetme conferences don't disconnect after the caller hangs up. Anyone else having this problem. (using Polycom phones)
  • Matthew Richmond, Wed 25 of Jul, 2007 [02:58 UTC]: thanks Nicholas Blasgen! I haven't worked with AGI before, but there's always a first! Thanks again!
  • Nicholas Blasgen, Tue 24 of Jul, 2007 [19:18 UTC]: Matthew Richmond, AGI will handle all that for you.
  • sam, Mon 23 of Jul, 2007 [16:39 UTC]: need help - certain voicemail extension will stop working and recording voicemail on asterisk - anyone know why and how to fix it? Thanks
  • john haji, Mon 23 of Jul, 2007 [14:55 UTC]: free calls to pakistan
  • bong, Sat 21 of Jul, 2007 [19:09 UTC]: hi good day to all can anyone help me how to configured the nortel sip to the signaling server and how to activate in mobile w/ sip compatible without mcs
Server Stats
  • Execution time: 0.37s
  • Memory usage: 2.22MB
  • Database queries: 30
  • GZIP: Disabled
  • Server load: 3.18

Asterisk cmd Monitor

Synopsis

Record a telephone conversation to a sound file

Description

  • Monitor(ext,basename)
  • Monitor(ext,basename,flags) — New feature added to CVS 2004-06-03

The Monitor command starts recording a channel. The channel's input and output voice packets are saved to separate sound files. You may change filenames during a recording by using the ChangeMonitor command. Recording continues until either the StopMonitor command is executed or the channel hangs up.
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).

A more detailed description on recording with Asterisk can be found at Asterisk cmd Record.

Command Parameters

  • ext: The sound file format to save in, which will be also used as the filename extension. Default: wav

  • basename: The base filename to use when saving the sound files. If not supplied, the default basename is constructed on the channel name plus a number, for example, IAX2[foo@bar]-3. The channel's input voice packets will be saved to basename-in.ext and the output voice packets will be saved to basename-out.ext. The default location for saved files is the /var/spool/asterisk/monitor directory.

  • flags: If flags contains the letter m, then when recording finishes, Asterisk will execute a unix program to combine the two sound files into a single sound file. By default, Asterisk will execute soxmix and then delete the original two sound files. Note that sox/soxmix may not necessarily understand the sound format (e.g. alaw) and can't therefore mix the in and out files down to one single file. You may specify a different mixing method by setting the MONITOR_EXEC channel variable to the path of the unix program you wish executed, then call Monitor to begin recording. At the completion of recording, the specified unix program will be executed with three command-line parameters: the two sound files and the filename where the program should save the combined sound file. In this situation, earlier versions of Asterisk will not delete the two original sound files; it's up to your program to do that if you need/wish to. The "m" flag is settable through the manager interface. Also see b - Don't begin recording unless a call is bridged to another channel.

Example 1

When a call is sent to extension 2060, recording of the call will begin, and the caller is sent to conference number 1 with the MeetMe command.

 exten => 2060,1,Answer
 exten => 2060,2,Wait(1)
 exten => 2060,3,Monitor(wav,myfilename)
 exten => 2060,4,Meetme(1,ps)

For an extensive example of using Monitor, see:


Example 2: 512 Simultaneous Calls with Digital Recording

How to use a RAM disk to eliminate the I/O bottleneck associated with digitally recording calls via the Monitor application:


See also


Asterisk | Configuration | Asterisk config extensi
ons.conf|The Dialplan - extensions.conf | Dialplan Commands
Created by oej, Last modification by Cyril Rbt on Thu 31 of May, 2007 [17:40 UTC]

Comments Filter

Only one user in the channel

by Cyril Rbt on Thursday 31 of May, 2007 [17:41:24 UTC]
Apparently, Monitoring skips when a user is alone on a channel (waiting for something, for example). If there are no
commands from Asterisk for more than 5 seconds, it won't be monitored. When listening to the recorded file, 5 seconds
could have been 10 minutes during the call.

Monitor Status?

by Richard Lavigne on Friday 18 of August, 2006 [20:40:21 UTC]
Is there a way to get a status of a channel to tell if it's being monitored/recorded, and to what filename?

via the manager API, "Action: Status" shows all channels as "Unmonitored", regardless if they're being recorded or not. (Does that "unmonitored" refer to something completely unrelated?)

option b may cause audio delay

by walter on Tuesday 04 of April, 2006 [15:33:27 UTC]
Please check before you use option b with monitor as may cause audio delay of 1 sec or greater from one side of the call.
Edit

MP3 Option

by Anonymous on Thursday 20 of January, 2005 [11:43:28 UTC]
Anyone like to set an extra-Option to make soxmix to mux into a mp3-file?

Please update this page with new information, just login and click on the "Edit" or "Add Comment" button above. Get a free login here: Register Thanks! - support@voip-info.org

Page Changes | Comments

Sponsored by:

Terms of Service Privacy Policy
© 2003-2007 Arte Marketing, Inc.

Powered by bitweaver