mvanbaak has committed a change to the trunk version of Asterisk which allows for it to perform like older versions of Asterisk did so that you can type the same commands on different versions of Asterisk you work with.
The diff is available from here.
The cli_aliases.conf file will contain by default:
;
; CLI Aliases configuration
;
; This module also registers a "cli show aliases" CLI command to list
; configured CLI aliases.
[general]
; Here you define what alias templates you want to use. You can also define
; multiple templates to use as well. If you do, and there is a conflict, then
; the first alias defined will win.
;
template = friendly ; By default, include friendly aliases
;template = asterisk12 ; Asterisk 1.2 style syntax
;template = asterisk14 ; Asterisk 1.4 style syntax
;template = individual_custom ; see [individual_custom] example below which
; includes a list of aliases from an external
; file
; Because the Asterisk CLI syntax follows a "module verb argument" syntax,
; sometimes we run into an issue between being consistant with this format
; in the core system, and maintaining system friendliness. In order to get
; around this we're providing some useful aliases by default.
;
[friendly]
hangup request=channel request hangup
originate=channel originate
help=core show help
pri intense debug span=pri set debug 2 span
; CLI Alias Templates
; -------------------
;
; You can define several alias templates.
; It works with context templates like all other configuration files
;
;[asterisk](!)
; To create an alias you simply set the variable name as the alias and variable
; value as the real CLI command you want executed
;
;die die die=stop now
;[asterisk16](asterisk)
; Alias for making voicemail reload actually do module reload app_voicemail.so
;voicemail reload=module reload app_voicemail.so
; This will make the CLI command "mr" behave as though it is "module reload".
;mr=module reload
;
;
; In addition, you could also include a flat file of aliases which is loaded by
; the [individual_custom] template in the [general] section.
;
;[individual_custom]
;#include "/etc/asterisk/aliases"
; Implemented CLI Alias Templates
; -------------------------------
;
; Below here we have provided you with some templates, easily allowing you to
; utilize previous Asterisk CLI commands with any version of Asterisk. In this
; way you will be able to use Asterisk 1.2 and 1.4 style CLI syntax with any
; version Asterisk going forward into the future.
;
; We have also separated out the vanilla syntax into a context template which
; allows you to keep your custom changes separate of the standard templates
; we have provided you. In this way you can clearly see your custom changes,
; and also allowing you to combine various templates as you see fit.
;
; The naming scheme we have used is recommended, but certainly is not enforced
; by Asterisk. If you wish to use the provided templates, simply define the
; context name which does not utilize the '_tpl' at the end. For example,
; if you would like to use the Asterisk 1.2 style syntax, define in the
; [general] section
[asterisk12_tpl](!)
show channeltypes=core show channeltypes
show channeltype=core show channeltype
show manager command=manager show command
show manager commands=manager show commands
show manager connected=manager show connected
show manager eventq=manager show eventq
rtp no debug=rtp set debug off
rtp rtcp debug ip=rtcp debug ip
rtp rtcp debug=rtcp debug
rtp rtcp no debug=rtcp debug off
rtp rtcp stats=rtcp stats
rtp rtcp no stats=rtcp stats off
stun no debug=stun debug off
udptl no debug=udptl debug off
show image formats=core show image formats
show file formats=core show file formats
show applications=core show applications
show functions=core show functions
show switches=core show switches
show hints=core show hints
show globals=core show globals
show function=core show function
show application=core show application
set global=core set global
show dialplan=dialplan show
show codecs=core show codecs
show audio codecs=core show audio codecs
show video codecs=core show video codecs
show image codecs=core show image codecs
show codec=core show codec
moh classes show=moh show classes
moh files show=moh show files
agi no debug=agi debug off
show agi=agi show
dump agihtml=agi dumphtml
show features=feature show
show indications=indication show
answer=console answer
hangup=console hangup
flash=console flash
dial=console dial
mute=console mute
unmute=console unmute
transfer=console transfer
send text=console send text
autoanswer=console autoanswer
oss boost=console boost
console=console active
save dialplan=dialplan save
add extension=dialplan add extension
remove extension=dialplan remove extension
add ignorepat=dialplan add ignorepat
remove ignorepat=dialplan remove ignorepat
include context=dialplan add include
dont include=dialplan remove include
extensions reload=dialplan reload
show translation=core show translation
convert=file convert
show queue=queue show
add queue member=queue add member
remove queue member=queue remove member
ael no debug=ael nodebug
sip debug=sip set debug
sip no debug=sip set debug off
show voicemail users=voicemail show users
show voicemail zones=voicemail show zones
iax2 trunk debug=iax2 set debug trunk
iax2 jb debug=iax2 set debug jb
iax2 no debug=iax2 set debug off
iax2 no trunk debug=iax2 set debug trunk off
iax2 no jb debug=iax2 set debug jb off
show agents=agent show
show agents online=agent show online
show memory allocations=memory show allocations
show memory summary=memory show summary
show version=core show version
show version files=core show file version
show profile=core show profile
clear profile=core clear profile
[asterisk12](asterisk12_tpl)
; add any additional custom commands you want below here, for example:
;die quickly=stop now
[asterisk14_tpl](!)
cdr status=cdr show status
rtp debug=rtp set debug on
rtcp debug=rtcp set debug on
rtcp stats=rtcp set stats on
stun debug=stun set debug on
udptl debug=udptl set debug on
core show globals=dialplan show globals
core set global=dialplan set global
core set chanvar=dialplan set chanvar
agi dumphtml=agi dump html
ael debug=ael set debug
funcdevstate list=devstate list
sip history=sip set history on
skinny debug=skinny set debug on
mgcp set debug=mgcp set debug on
abort shutdown=core abort shutdown
stop now=core stop now
stop gracefully=core stop gracefully
stop when convenient=core stop when convenient
restart now=core restart now
restart gracefully=core restart gracefully
restart when convenient=core restart when convenient
[asterisk14](asterisk14_tpl)
; add any additional custom commands you want below here.
Current Rating: 0/10 (0 votes) Comments (1 posted) ZyZvbVjCvddL Written by Wolf59 (October 23, 2009, 7:43 am)
Anyway these are some great tips & these posts are even better. , Similar Articles (Based on Title) RFC: internal CLI changes - November 14, 2006 Luigi has posted a request for comment on a bugtracker entry.
VoIP-info.org: Version 0.9.1 of LDAPget application module released. - December 7, 2004 Version 0.9.1 of LDAPget application module has been released. It now supports the most recent CVS.
JavaScript module for Asterisk Unveiled - June 22, 2005 Anthm has posted details of a new module for using javascript with Asterisk.
*-Dev: asterisk audio conversion module - December 20, 2005 Redice Li has posted details of an audio conversion module for Asterisk.
*-Dev: Module testing framework - December 29, 2005 Olle has posted details of a new module test branch.
New Asterisk Module Generator Tool - August 23, 2006 Justin Tunney has posted details of a web based application that will generate the framework necessary for building your own Asterisk modules.
Another module for testing: chan_console - December 28, 2007 Russell Bryant has posted details of a branch of SVN which is ready for testing.
How to write an Asterisk Module - June 23, 2008 Russell has written a two part article on writing Asterisk modules.
How-to: Write an Asterisk Module, Part 3 - July 1, 2008 Russell has posted the third in his series on writing Asterisk modules.
Digium G.729 codec module updated and new performance testing tool released - February 17, 2009 Kevin Fleming has posted details of an update to G.729.
PA648C Video Compression Module - August 19, 2009 Palmmicro Video and Audio compression chip with H.264 support.
Flite module for asterisk 1.6.x - August 31, 2009 zaf has written a flite module which does text to speech with Asterisk.
RSS News Feed Added - September 18, 2004 We have added an RSS News Feed to the SineApps News Page
Original Content (C) 2004-2010
Matt Riddell

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Back to life July 21, 2010 Average Vote: 10
Hey all - I am back online after some pretty big projects which have taken all my time. Will be updating the Asterisk news over the next few days.
Nerd Vittles: Building a Bluetooth Proximity Detection System with Asterisk December 12, 2005 Average Vote: 10
The Nerd Vittles site has an article on proximity detection using Asterisk and a TomTom GPS
Automated Testing Update July 30, 2010 Average Vote: 10
Russell Bryant has posted details of a new mailing list for automated testing of Asterisk and some information on the progress that has been made. There is no way to say how important I think this work is. It really makes a huge difference to Asterisk and the ability to use it in an enterprise environment. Really great work.
VoIP-Info: FFasterisk Video file converter August 25, 2006 Average Vote: 10
The wiki has a link to a new piece of software for converting video to the format required for Asterisk.
Code Review: SRTP support for Asterisk March 12, 2009 Average Vote: 10
Terry Wilson has moved his SRTP branch onto the Digium review board.
HumBug - Pre BETA Launch Registration July 27, 2010 Average Vote: 10
Nir Simionovich has posted details of the beta of the new call analytics service.
Interview with BKW_ December 7, 2004 Average Vote: 10
We've finally completed our interview with BKW. Hope you like! :-)
SlashDot: GSM and Asterisk Integration August 21, 2005 Average Vote: 10
There is a post up on SlashDot which talks about using cellphones with Asterisk.
Interview with Mark Spencer November 26, 2004 Average Vote: 9.9
We have managed to get an interview with Mark Spencer AKA Markster. Mark Spencer is the creator of Asterisk and by far the most active developer.
Asterisk and Kamailio realtime integration tutorial May 24, 2010 Average Vote: 9.9
Daniel-Constantin Mierla has posted a link to a tutorial on integrating Asterisk and Kamailio using realtime.
Asterisk IPv6 update February 1, 2010 Average Vote: 9.8
Olle has posted an update on IPV6 in Asterisk and a link to a blog post of his.
Proposal for T.38 transparent gateway design in Asterisk April 29, 2010 Average Vote: 9.8
Kevin Fleming has posted a proposed design for a transparent T.38 gateway for Asterisk:
Asterisk Monitoring with iPhone and iPod touch February 12, 2010 Average Vote: 9.7
For the past couple of weeks I have been working on an app that allows you to monitor and restart Asterisk servers.
Monitoring Asterisk with Munin January 7, 2010 Average Vote: 9.7
I had a few requests for these munin plugins after some discussion on one of the Asterisk lists and thought people might like them.
New Zealand Asterisk Voices March 2, 2006 Average Vote: 9.7
Chris Hodgetts has posted details of recordings of Asterisk Sounds with a New Zealand accent.
Automated Testing Update July 30, 2010 Russell Bryant has posted details of a new mailing list for automated testing of Asterisk and some information on the progress that has been made. There is no way to say how important I think this work is. It really makes a huge difference to Asterisk and the ability to use it in an enterprise environment. Really great work.
Asterisk 1.8.0-beta2 Now Available July 28, 2010 The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta2.
HumBug - Pre BETA Launch Registration July 27, 2010 Nir Simionovich has posted details of the beta of the new call analytics service.
Branch Merging Changes July 26, 2010 Russell Bryant has posted details of some changes to the way developers need to commit code to Asterisk because of the newly released 1.8 branch.
Asterisk 1.8.0-beta1 is Now Available July 26, 2010 The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta1. This release marks the beginning of the testing process for the eventual release of Asterisk 1.8.0.
Asterisk 1.6.2.10 Now Available July 26, 2010 The Asterisk Development Team has announced the release of Asterisk 1.6.2.10.
Asterisk 1.4.34 Now Available July 26, 2010 The Asterisk Development Team has announced the release of Asterisk 1.4.34.
AppleRaisin - AstDB over realtime July 23, 2010 Olle has posted a note about his awesome AppleRaisin branch which provides the ability to store AstDB in realtime. This would make for a much simpler failover and clustering situation.
QueueMetrics 1.6.1 released July 22, 2010 Lenz has posted a note to inform us that QueueMetrics version 1.6.1 has been released. This release offers a large number of bug fixes, misc improvements and new developements including hotdesking.
Asterisk 1.8 Branch Creation July 22, 2010 Russell Bryant has posted a note to inform us of the creation of the 1.8 branch of Asterisk. |