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Unofficial devel of Digium's Asterisk (tm)

tag: 10.2.0-rc1

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Octocat-spinner-32 addons
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Octocat-spinner-32 apps
Octocat-spinner-32 autoconf
Octocat-spinner-32 bridges Fix crash from bridge channel hangup race condition in ConfBridge January 13, 2012
Octocat-spinner-32 build_tools
Octocat-spinner-32 cdr
Octocat-spinner-32 cel
Octocat-spinner-32 channels
Octocat-spinner-32 codecs
Octocat-spinner-32 configs
Octocat-spinner-32 contrib
Octocat-spinner-32 doc
Octocat-spinner-32 formats
Octocat-spinner-32 funcs
Octocat-spinner-32 images even uglier gui with more buttons July 01, 2008
Octocat-spinner-32 include
Octocat-spinner-32 main
Octocat-spinner-32 menuselect
Octocat-spinner-32 pbx
Octocat-spinner-32 phoneprov res_phoneprov add snom 300, 320, 360, 370, 820, 821, 870 support February 03, 2011
Octocat-spinner-32 res
Octocat-spinner-32 sounds Set core sounds version to 1.4.22. January 24, 2012
Octocat-spinner-32 static-http Merged revisions 291575 via svnmerge from October 13, 2010
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Octocat-spinner-32 utils
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Octocat-spinner-32 BUGS These changes should really be in trunk, not the 1.10 branch. July 13, 2011
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Octocat-spinner-32 README-SERIOUSLY.bestpractices.txt Merged revisions 247509 via svnmerge from February 18, 2010
Octocat-spinner-32 README-addons.txt Move Asterisk-addons modules into the main Asterisk source tree. June 30, 2009
Octocat-spinner-32 UPGRADE-1.2.txt as suggested by jtodd, document the purposes of the CHANGES and UPGRA… November 21, 2008
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Octocat-spinner-32 UPGRADE.txt
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README
===============================================================================
===                     The Asterisk(R) Open Source PBX
===
===                   by Mark Spencer <markster@digium.com>
===                  and the Asterisk.org developer community
===
===                    Copyright (C) 2001-2009 Digium, Inc.
===                       and other copyright holders.
===============================================================================

-------------------------------------------------------------------------------
--- SECURITY ------------------------------------------------------------------

  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.

  If you downloaded Asterisk as a tarball, see the security section in the PDF
version of the documentation in doc/tex/asterisk.pdf.  Alternatively, pull up
the HTML version of the documentation in doc/tex/asterisk/index.html.  The
source for the security document is available in doc/tex/security.tex.
-------------------------------------------------------------------------------

-------------------------------------------------------------------------------
--- 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.  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 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

  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 http://www.asteriskdocs.org web site.
-------------------------------------------------------------------------------

-------------------------------------------------------------------------------
--- SUPPORTED 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, Cygwin,
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 sound card) to install and run Asterisk.

  Supported telephony hardware includes:

	* All Analog and Digital Interface cards from Digium (www.digium.com)
	* QuickNet Internet PhoneJack and LineJack (http://www.quicknet.net)
	* 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

-------------------------------------------------------------------------------

-------------------------------------------------------------------------------
--- UPGRADING FROM AN EARLIER VERSION -----------------------------------------

  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.
-------------------------------------------------------------------------------

-------------------------------------------------------------------------------
--- NEW INSTALLATIONS ---------------------------------------------------------

  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 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 README 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. They are all in the configs/ directory.

2) Run "./configure"

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

3) 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.

4) Run "make"

  Assuming the build completes successfully:

5) 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:

6) "make samples"

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

  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:

*CLI>

  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 <command>".  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.
-------------------------------------------------------------------------------

-------------------------------------------------------------------------------
--- 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 dahdi.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 approximately 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 on various features. Again,
please read all the configuration samples that include documentation on
the configuration options.

  If this release of Asterisk was downloaded from a tarball, then some
additional documentation should have been included.
     * doc/tex/asterisk.pdf --- PDF version of the documentation
     * doc/tex/asterisk/index.html --- HTML version of the 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/support

  Welcome to the growing worldwide community of Asterisk users!
-------------------------------------------------------------------------------

--- Mark Spencer, and the Asterisk.org development community

-------------------------------------------------------------------------------
Asterisk is a trademark of Digium, Inc.
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