The Asterisk Open Source PBX

by Mark Spencer <markster@digium.com>

Copyright (C) 2001-2005 Digium, Inc.

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* SECURITY

  It is imperative that you read and fully understand the contents of

  the SECURITY file before you attempt to configure an Asterisk server.



* WHAT IS ASTERISK

  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:



           http://www.asterisk.org



In addition you'll find lots of information compiled by the Asterisk

community on this Wiki:



           http://www.voip-info.org/wiki-Asterisk



* LICENSING

  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

Asterisk.



  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.

  

* OPERATING SYSTEMS



== 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

distribution.



== 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.



* GETTING STARTED



  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 (www.digium.com)

	* QuickNet Internet PhoneJack and LineJack (http://www.quicknet.net)

	* 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:



*CLI>



  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.



* ABOUT CONFIGURATION FILES



  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:



	switchtype=national



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.



* SPECIAL NOTE ON TIME

  

  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 (http://www.ntp.org/) 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.



* FILE DESCRIPTORS



  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.



* MORE INFORMATION



  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.



   http://www.asterisk.org/index.php?menu=support



  Welcome to the growing worldwide community of Asterisk users!



Mark Spencer