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The Asterisk Open Source PBX
by Mark Spencer <>
Copyright (C) 2001-2005 Digium, Inc.
  It is imperative that you read and fully understand the contents of
  the SECURITY file before you attempt to configure an Asterisk server.

  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.  For more information
on the project itself, please visit the Asterisk home page at:


In addition you'll find lots of information compiled by the Asterisk
community on this Wiki:


  Asterisk is distributed under GNU General Public License and is also
available under alternative licenses negotiated directly with Digium, Inc.
If you obtained Asterisk under the GPL, then the GPL applies to all
loadable modules used on your system as well, except as defined below.

  Digium, Inc. (formerly Linux Support Services) retains copyright and/or a
sufficient license to all components of the core Asterisk system, and therefore
can grant, at its sole discretion, the ability for companies, individuals, or
organizations to create proprietary or Open Source (but non-GPL'd) modules
which may be dynamically linked at runtime with the portions of Asterisk which
fall under our copyright/license umbrella, or are distributed under more
flexible licenses than GPL.  

  If you wish to use our code in other GPL programs, don't worry -- there
is no requirement that you provide the same exception in your GPL'd
products (although if you've written a module for Asterisk we would
strongly encourage you to make the same exception that we do).

  Specific permission is also granted to OpenSSL and OpenH323 to link with

  If you have any questions, whatsoever, regarding our licensing policy,
please contact us.

  Modules that are GPL-licensed and not available under Digium's 
licensing scheme are added to the Asterisk-addons CVS module.

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

== Others ==
  Asterisk has also been 'ported' and reportedly runs properly on other
operating systems as well, including Sun Solaris, Apple's Mac OS X, 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 soundcard) to install and run Asterisk.

  Supported telephony hardware includes:

	* All Wildcard (tm) products from Digium (
	* QuickNet Internet PhoneJack and LineJack (
	* any full duplex sound card supported by ALSA or OSS
	* ISDN4Linux compatible ISDN card
        * VoiceTronix OpenLine products

Hint: CAPI compatible ISDN cards can be run using the add-on channel chan_capi.

  Second, ensure that your system contains a compatible compiler and development
libraries.  Asterisk requires either the GNU Compiler Collection (GCC) version
3.0 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 OpenSSL and zlib.
On many distributions, these files are installed by packages with names like
'libc-devel', 'openssl-devel' and 'zlib-devel' or similar.

  So let's proceed:

1) Run "make"

  Assuming the build completes successfully:

2) Run "make install"

  Each time you update or checkout from CVS, you are strongly encouraged 
to ensure all previous object files are removed to avoid internal 
inconsistency in Asterisk. Normally, this is automatically done with 
the presence of the file .cleancount, which increments each time a 'make clean'
is required, and the file .lastclean, which contains the last .cleancount used. 

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

3) "make samples"

  Doing so will overwrite any existing config files you have. If you are lacking a
soundcard you won't be able to use the DIAL command on the console, though.

  Finally, you can launch Asterisk 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 "help" at any time to get help with the system.  For help
with a specific command, type "help <command>".  To start the PBX using
your sound card, you can type "dial" to dial the PBX.  Then you can use
"answer", "hangup", and "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).

  Feel free to look over the configuration files in /etc/asterisk, where
you'll 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 zapata.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

  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 approxiately 150
SIP calls simultaneously.  To change the number of file descriptors
follow the instructions for your system below:

== PAM-based Linux System ==

  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.

== Generic UNIX System ==

  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.

  Finally, you may wish to visit the web 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