赤峰市会计培训班:Asterisk 1.4.0 CLI commands

Introduction
Asterisk CLI supports large variety of commands which can be used for testing, configuration and monitoring. In this tutorial we will describe all commands available at the standard Asterisk version 1.4.0. We will divide this tutorial into few sections in order to facilitate the reading.
General CLI commands
! - Execute a shell command
abort halt - Cancel a running halt
cdr status - Display the CDR status
feature show - Lists configured features
feature show channels - List status of feature channels
file convert - Convert audio file
group show channels - Display active channels with group(s)
help - Display help list, or specific help on a command
indication add - Add the given indication to the country
indication remove - Remove the given indication from the country
indication show - Display a list of all countries/indications
keys init - Initialize RSA key passcodes
keys show - Displays RSA key information
local show channels - List status of local channels
logger mute - Toggle logging output to a console
logger reload - Reopens the log files
logger rotate - Rotates and reopens the log files
logger show channels - List configured log channels
meetme - Execute a command on a conference or conferee
mixmonitor - Execute a MixMonitor command.
User Comments
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10 June 2009 19:10:37
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11 April 2009 08:30:00
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Muhammad Shahzad (muhammadshahzad27 at hotmail dot com)
07 January 2009 20:45:36
Can any one help me to set up an IVR in asterisk step by step bc i am new for asterisk
waiting 4 ur favaourable reply
syed (sdzubair76 at gmail dot com)
06 November 2008 10:25:59
i want to set a song rather than ringing sound on all extentions in my network.
help me
asif (asifaleem_atk at yahoo dot com)
22 April 2008 05:50:07
i want to know about asterisk outbound calls based on extensions.
anthony (asticha at voicevalley dot com)
05 April 2008 00:32:08
pri show span no longer works in asterisk
Connected to Asterisk 1.4.18 currently running on scp-0100 (pid = 2570)
Verbosity is at least 20
-- Remote UNIX connection
scp-0100*CLI> pri show span
No such command 'pri show' (type 'help' for help)
Sunder (sunder dot kotha at gmail dot com)
13 March 2008 11:23:48
Hi,
I am having the problem which is call was disconnecting itself after sometime.Plz give me right solution.
Regards
Sunder
Arros (namnhanco at gmail dot com)
14 February 2008 02:49:50
Hi ! I know all command of asterisk , please send me
Thanks.
David Aman (daman1 at Rochester dot rr dot com)
06 February 2008 01:04:06
Ho do you open a website using the CLI?
asif aleem (asifaleem_atk at yahoo dot com)
25 January 2008 05:25:27
i want to know how to integrate dnc list with my asterisk
GGG (gal83 at abv dot bg)
14 December 2007 22:24:12
Hi,
I have no astman on my Asterisk and I try to install but it`s does not work.
aleem (asif dot aleem at gmail dot com)
06 December 2007 11:24:42
i need the asterisk whole application source code in php
asif aleem (asifaleem_atk at yahoo dot com)
06 December 2007 11:22:58
i need the source code of asterisk whole application
Johnny (johnnycctam at hotmail dot com)
27 October 2007 10:07:44
Reinstalling Asterisk 1.4.13. However, after re-installing, I am missing the
realtime mysql status CLI commands.
Hariharan (hariharan dot s dot pillai at gmail dot com)
04 September 2007 22:44:14
Hi, I am having a problem that i have multiple sip a/c that evry body 8 members havin diffrent sip a/c but when a user dialing other sip were not working it says the a/c already in use how to goabout this is thier any soution for this i want if one line was busy it should go to next one and start dialing ok
here is my mine ambition i want to use predictvie dialing if it is possible with a sip a/c to share with several members thay all
AleE (alee dot riz at gmail dot com)
16 August 2007 20:41:40
Hi Experts,
I am having problems like distortion, call hangup and low voice.
System: Fedore Core 6
Asterisk version 1.4.0
Softphone: Eyebeam
Audio Codec: G729
Allocated bandwidth: 512kbps
around 15 Agents.
please help me out experts.
regards
Ali
baosongwei (bsw007 at sina dot com)
11 June 2007 04:37:04
one sip channel in use but not real
so how can i stop it
Patrick (patrickkadama at yahoo dot com)
29 May 2007 09:59:20
It interesting to see this happen. However, I am not able to dial from one client yet can from another. Secondly, I can't send text msgs. waiting 4 ur advice.
Luis (luis at teledata dot com dot uy)
13 April 2007 13:57:11
This chapter should include a section on CLI commands for Asterisk version 1.2.0 also.
Emmanuel (emmanuel80 at hotmail dot com)
16 March 2007 01:57:47
Hi, I'd like to join two asterisk servers with sip, i got this:
> ------
> ----
Iam missing something?
souvik (souvik_sadhu at yahoo dot com)
01 March 2007 10:05:43
HI..
I m facing one problem that...
after installation of everything, i m not able to use these zap command.
I have installed
1.libpri
2.zaptel
3.asterisk
into /usr/src/asterisk/....
Can anybody tell me what is the problem..
and how to solve this problem.
I m also not getting the dial tone( in TDM11B)
Waiting for response
souvik
Miguel (miguel dot palmer at beronet dot com)
22 February 2007 14:09:32
Hello, I work now in an interface for Asterisk Manager in HTML, but I want to use the command UpdateConfig, and i can just delete de first category, and I want too others categorys delete. Can anybody help me. The answers can be too in German or Spanish. Chao