asterisk/README

The Asterisk Open Source PBX
by Mark Spencer <markster@digium.com>
Copyright (C) 2001-2005 Digium, Inc.
================================================================
* 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 the Asterisk is sensitive to large
jumps in time. Those who live in areas that are on Daylight Savings Time (or
equivalent) should set their system and hardware clocks to use UTC in order
to avoid any possible jumps in system time. There should be no noticeable
effects to the user, as you should still set your system to use the local
offset from UTC.

  Even for those who don't live in DST zones, this issue may manifest itself
if the administrator makes large manual time adjustments. Thus, it is good
practice to keep the time on your Asterisk server synced to a reliable
source, such as an NTP server. 

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

* 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