Skip to content
Mirror of the official Asterisk ( Project repository. No pull requests here please. Use Gerrit:
C C++ Python Shell M4 Makefile Other
Branch: master
Clone or download
Friendly Automation Gerrit Code Review
Latest commit e2119e8 Jan 28, 2020
Type Name Latest commit message Commit time
Failed to load latest commit information.
addons configure: Add check for MySQL client bool and my_bool type usage. Dec 16, 2019
agi Remove as much trailing whitespace as possible. Dec 22, 2017
apps Merge "app_voicemail: Remove MessageExists and MESSAGE_EXISTS()" Jan 22, 2020
autoconf Check for unbound version >= 1.5 Sep 25, 2018
bridges confbridge: Add support for specifying maximum sample rate. Dec 16, 2019
build_tools Create --disable-binary-modules option. Aug 27, 2018
cdr cdr_pgsql cel_pgsql res_config_pgsql: compatibility with PostgreSQL 12 Oct 14, 2019
cel cdr_pgsql cel_pgsql res_config_pgsql: compatibility with PostgreSQL 12 Oct 14, 2019
channels Merge "chan_sip: Always process updated SDP on media source change" Jan 28, 2020
codecs various files - fix some alerts raised by lgtm code analysis Nov 18, 2019
configs http: Add ability to disable /httpstatus URI Jan 22, 2020
contrib Merge "queue_log: Add alembic script for generate db table for queue_… Jan 20, 2020
doc Merge "http: Add ability to disable /httpstatus URI" Jan 23, 2020
formats various files - fix some alerts raised by lgtm code analysis Nov 18, 2019
funcs func_curl: Add 'followlocation' option to CURLOPT() Jan 13, 2020
images even uglier gui with more buttons Jul 1, 2008
include Merge "feat: AudioSocket channel, application, and ARI support." Jan 15, 2020
main Merge "http: Add ability to disable /httpstatus URI" Jan 23, 2020
menuselect menuselect: Fix curses build on Gentoo Linux Aug 9, 2019
pbx pbx_dundi: added IPv4/IPv6 dual bind support to DUNDi Jun 5, 2019
phoneprov res_phoneprov add snom 300, 320, 360, 370, 820, 821, 870 support Feb 3, 2011
res res_pjsip_messaging: Ensure MESSAGE_SEND_STATUS is set properly Jan 27, 2020
rest-api-templates res_ari_events: Add module reference when a WebSocket is open. Oct 24, 2019
rest-api Merge "feat: AudioSocket channel, application, and ARI support." Jan 15, 2020
sounds BuildSystem: When no download utility is available, display the expla… Mar 17, 2018
static-http Remove as much trailing whitespace as possible. Dec 22, 2017
tests CI: Update to do a "make full" Jan 8, 2020
third-party pjproject_bundled: Replace earlier reverts with official fixes. Oct 10, 2019
utils Fixes for GCC 9 May 10, 2019
.cleancount Remove obsolete struct ast_channel note. Jun 29, 2012
.gitignore CI: Initial commit for moving CI into source repo Jul 11, 2018
.gitreview gitreview: Reorder and add padding. Feb 25, 2018
BSDmakefile Merged revisions 285090 via svnmerge from Sep 6, 2010
BUGS Add UPGRADE-1.10.txt file from UPGRADE.txt. Jul 13, 2011
CHANGES Update CHANGES and UPGRADE.txt for 17.0.0 Jul 29, 2019
COPYING remove extraneous svn:executable properties Nov 29, 2005
CREDITS Fix some invalid Unicode characters Dec 21, 2017
LICENSE LICENSE: Clarify language in Asterisk's LICENSE to allow for linking … Aug 28, 2014
Makefile Build: Separate header install/uninstall Jul 16, 2019
Makefile.moddir_rules Makefile.moddir_rules: Pass PJPROJECT_BUNDLED to download_externals Mar 12, 2019
Makefile.rules core/buildsystem: check the actual compiler being version Apr 22, 2019 Speling correetions. Oct 16, 2019
README-addons.txt README*: Remove trailing whitespace Aug 22, 2015 Update year Jul 5, 2019
UPGRADE.txt Update CHANGES and UPGRADE.txt for 17.0.0 Jul 29, 2019
Zaptel-to-DAHDI.txt Merged revisions 137679 via svnmerge from Aug 14, 2008 BuildSystem: Enable autotools in Solaris 11. Jun 20, 2018
config.guess build: Update config.guess and config.sub Dec 11, 2018
config.sub build: Update config.guess and config.sub Dec 11, 2018
configure configure: Add check for MySQL client bool and my_bool type usage. Dec 16, 2019 configure: Add check for MySQL client bool and my_bool type usage. Dec 16, 2019
default.exports Add _IO_stdin_used in version-script to fix SIGBUSes on Sparc. Aug 22, 2013
install-sh Remove as much trailing whitespace as possible. Dec 22, 2017 Enable bundling of jansson, require 2.11. Jul 20, 2018
missing silly people that don't want to install/run autoconf :-) May 8, 2006
mkinstalldirs silly people that don't want to install/run autoconf :-) May 8, 2006 Remove as much trailing whitespace as possible. Dec 22, 2017

The Asterisk(R) Open Source PBX

        By Mark Spencer <> and the developer community.
        Copyright (C) 2001-2019 Digium, Inc. and other copyright holders.


It is imperative that you read and fully understand the contents of the security information document before you attempt to configure and run an Asterisk server.

See Important Security Considerations for more information.


Asterisk is an Open Source PBX and telephony toolkit. It is, in a sense, middleware between Internet and telephony channels on the bottom, and Internet and telephony applications at the top. However, Asterisk supports more telephony interfaces than just Internet telephony. Asterisk also has a vast amount of support for traditional PSTN telephony, as well.

For more information on the project itself, please visit the Asterisk home page and the official wiki. In addition you'll find lots of information compiled by the Asterisk community at

There is a book on Asterisk published by O'Reilly under the Creative Commons License. It is available in book stores as well as in a downloadable version on the web site.



The Asterisk Open Source PBX is developed and tested primarily on the GNU/Linux operating system, and is supported on every major GNU/Linux distribution.


Asterisk has also been 'ported' and reportedly runs properly on other operating systems as well, including Sun Solaris, Apple's Mac OS X, Cygwin, and the BSD variants.


First, be sure you've got supported hardware (but note that you don't need ANY special hardware, not even a sound card) to install and run Asterisk.

Supported telephony hardware includes:

  • All Analog and Digital Interface cards from Digium
  • QuickNet Internet PhoneJack and LineJack (
  • any full duplex sound card supported by ALSA, OSS, or PortAudio
  • any ISDN card supported by mISDN on Linux
  • The Xorcom Astribank channel bank
  • VoiceTronix OpenLine products


If you are updating from a previous version of Asterisk, make sure you read the UPGRADE.txt file in the source directory. There are some files and configuration options that you will have to change, even though we made every effort possible to maintain backwards compatibility.

In order to discover new features to use, please check the configuration examples in the configs directory of the source code distribution. For a list of new features in this version of Asterisk, see the CHANGES file.


Ensure that your system contains a compatible compiler and development libraries. Asterisk requires either the GNU Compiler Collection (GCC) version 4.1 or higher, or a compiler that supports the C99 specification and some of the gcc language extensions. In addition, your system needs to have the C library headers available, and the headers and libraries for ncurses.

There are many modules that have additional dependencies. To see what libraries are being looked for, see ./configure --help, or run make menuselect to view the dependencies for specific modules.

On many distributions, these dependencies are installed by packages with names like 'glibc-devel', 'ncurses-devel', 'openssl-devel' and 'zlib-devel' or similar.

So, let's proceed:

  1. Read this file.

There are more documents than this one in the doc directory. You may also want to check the configuration files that contain examples and reference guides in the configs directory.

  1. Run ./configure

Execute the configure script to guess values for system-dependent variables used during compilation.

  1. Run make menuselect [optional]

This is needed if you want to select the modules that will be compiled and to check dependencies for various optional modules.

  1. Run make

Assuming the build completes successfully:

  1. Run make install

If this is your first time working with Asterisk, you may wish to install the sample PBX, with demonstration extensions, etc. If so, run:

  1. Run make samples

Doing so will overwrite any existing configuration files you have installed.

  1. Finally, you can launch Asterisk in the foreground mode (not a daemon) with:
        # asterisk -vvvc

You'll see a bunch of verbose messages fly by your screen as Asterisk initializes (that's the "very very verbose" mode). When it's ready, if you specified the "c" then you'll get a command line console, that looks like this:


You can type "core show help" at any time to get help with the system. For help with a specific command, type "core show help ". To start the PBX using your sound card, you can type "console dial" to dial the PBX. Then you can use "console answer", "console hangup", and "console dial" to simulate the actions of a telephone. Remember that if you don't have a full duplex sound card (and Asterisk will tell you somewhere in its verbose messages if you do/don't) then it won't work right (not yet).

"man asterisk" at the Unix/Linux command prompt will give you detailed information on how to start and stop Asterisk, as well as all the command line options for starting Asterisk.

Feel free to look over the configuration files in /etc/asterisk, where you will find a lot of information about what you can do with Asterisk.


All Asterisk configuration files share a common format. Comments are delimited by ';' (since '#' of course, being a DTMF digit, may occur in many places). A configuration file is divided into sections whose names appear in []'s. Each section typically contains two types of statements, those of the form 'variable = value', and those of the form 'object => parameters'. Internally the use of '=' and '=>' is exactly the same, so they're used only to help make the configuration file easier to understand, and do not affect how it is actually parsed.

Entries of the form 'variable=value' set the value of some parameter in asterisk. For example, in chan_dahdi.conf, one might specify:


In order to indicate to Asterisk that the switch they are connecting to is of the type "national". In general, the parameter will apply to instantiations which occur below its specification. For example, if the configuration file read:

	switchtype = national
	channel => 1-4
	channel => 10-12
	switchtype = dms100
	channel => 25-47

The "national" switchtype would be applied to channels one through four and channels 10 through 12, whereas the "dms100" switchtype would apply to channels 25 through 47.

The "object => parameters" instantiates an object with the given parameters. For example, the line "channel => 25-47" creates objects for the channels 25 through 47 of the card, obtaining the settings from the variables specified above.


Those using SIP phones should be aware that Asterisk is sensitive to large jumps in time. Manually changing the system time using date(1) (or other similar commands) may cause SIP registrations and other internal processes to fail. If your system cannot keep accurate time by itself use NTP to keep the system clock synchronized to "real time". NTP is designed to keep the system clock synchronized by speeding up or slowing down the system clock until it is synchronized to "real time" rather than by jumping the time and causing discontinuities. Most Linux distributions include precompiled versions of NTP. Beware of some time synchronization methods that get the correct real time periodically and then manually set the system clock.

Apparent time changes due to daylight savings time are just that, apparent. The use of daylight savings time in a Linux system is purely a user interface issue and does not affect the operation of the Linux kernel or Asterisk. The system clock on Linux kernels operates on UTC. UTC does not use daylight savings time.

Also note that this issue is separate from the clocking of TDM channels, and is known to at least affect SIP registrations.


Depending on the size of your system and your configuration, Asterisk can consume a large number of file descriptors. In UNIX, file descriptors are used for more than just files on disk. File descriptors are also used for handling network communication (e.g. SIP, IAX2, or H.323 calls) and hardware access (e.g. analog and digital trunk hardware). Asterisk accesses many on-disk files for everything from configuration information to voicemail storage.

Most systems limit the number of file descriptors that Asterisk can have open at one time. This can limit the number of simultaneous calls that your system can handle. For example, if the limit is set at 1024 (a common default value) Asterisk can handle approximately 150 SIP calls simultaneously. To change the number of file descriptors follow the instructions for your system below:


If your system uses PAM (Pluggable Authentication Modules) edit /etc/security/limits.conf. Add these lines to the bottom of the file:

root            soft    nofile          4096
root            hard    nofile          8196
asterisk        soft    nofile          4096
asterisk        hard    nofile          8196

(adjust the numbers to taste). You may need to reboot the system for these changes to take effect.


If there are no instructions specifically adapted to your system above you can try adding the command ulimit -n 8192 to the script that starts Asterisk.


See the doc directory for more documentation on various features. Again, please read all the configuration samples that include documentation on the configuration options.

Finally, you may wish to visit the support site and join the mailing list if you're interested in getting more information.

Welcome to the growing worldwide community of Asterisk users!

        Mark Spencer, and the development community

Asterisk is a trademark of Digium, Inc.

You can’t perform that action at this time.