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Various README updates

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Russell Bryant
2008-01-19 05:26:46 +00:00
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README
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@@ -1,65 +1,81 @@
The Asterisk(R) Open Source PBX
by Mark Spencer <markster@digium.com>
and the Asterisk.org developer community
===============================================================================
=== The Asterisk(R) Open Source PBX
===
=== by Mark Spencer <markster@digium.com>
=== and the Asterisk.org developer community
===
=== Copyright (C) 2001-2008 Digium, Inc.
=== and other copyright holders.
===============================================================================
Copyright (C) 2001-2006 Digium, Inc.
and other copyright holders.
================================================================
-------------------------------------------------------------------------------
--- SECURITY ------------------------------------------------------------------
* SECURITY
It is imperative that you read and fully understand the contents of
the security information file (doc/security.txt) before you attempt
to configure and run an Asterisk server.
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 ? --------------------------------------------------------
* 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:
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
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.
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
-------------------------------------------------------------------------------
--- SUPPORTED OPERATING SYSTEMS -----------------------------------------------
== Linux ==
--- 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 ==
--- 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
-------------------------------------------------------------------------------
--- 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.
ANY special hardware, not even a sound card) to install and run Asterisk.
Supported telephony hardware includes:
* All Wildcard (tm) products from Digium (www.digium.com)
* 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 or OSS
* any full duplex sound card supported by ALSA, OSS, or PortAudio
* any ISDN card supported by mISDN on Linux (BRI)
* The Xorcom AstriBank channel bank
* VoiceTronix OpenLine products
* VoiceTronix OpenLine products
The are several drivers for ISDN BRI cards available from third party sources.
Check the voip-info.org wiki for more information on chan_capi and
zaphfc.
-------------------------------------------------------------------------------
* UPGRADING FROM AN EARLIER VERSION
-------------------------------------------------------------------------------
--- 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
@@ -67,29 +83,34 @@ 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.
To discover the major new features of Asterisk 1.2, please visit
http://edvina.net/asterisk1-2/
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
-------------------------------------------------------------------------------
--- 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 OpenSSL,
ncurses and zlib.
On many distributions, these files are installed by packages with names like
'glibc-devel', 'ncurses-devel', 'openssl-devel' and 'zlib-devel' or similar.
library headers available, and the headers and libraries for ncurses.
So let's proceed:
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.
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"
@@ -98,8 +119,8 @@ 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 modules dependencies.
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"
@@ -107,21 +128,14 @@ compiled and to check modules dependencies.
5) Run "make install"
Each time you update or checkout from the repository, 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:
6) "make samples"
Doing so will overwrite any existing config files you have.
Doing so will overwrite any existing configuration files you have installed.
Finally, you can launch Asterisk in the foreground mode (not a daemon)
with:
Finally, you can launch Asterisk in the foreground mode (not a daemon) with:
# asterisk -vvvc
@@ -134,20 +148,22 @@ 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).
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'll find a lot of information about what you can do with 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
-------------------------------------------------------------------------------
--- 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
@@ -163,7 +179,7 @@ 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
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:
@@ -174,7 +190,7 @@ configuration file read:
switchtype = dms100
channel => 25-47
the "national" switchtype would be applied to channels one through
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.
@@ -182,8 +198,10 @@ apply to channels 25 through 47.
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
-------------------------------------------------------------------------------
--- 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)
@@ -206,8 +224,10 @@ 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
-------------------------------------------------------------------------------
--- FILE DESCRIPTORS ----------------------------------------------------------
Depending on the size of your system and your configuration,
Asterisk can consume a large number of file descriptors. In UNIX,
@@ -220,11 +240,13 @@ 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
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 ==
-------------------------------------------------------------------------------
--- 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:
@@ -242,21 +264,29 @@ 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.
-------------------------------------------------------------------------------
* MORE INFORMATION
-------------------------------------------------------------------------------
--- 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
--- Mark Spencer, and the Asterisk.org development community
----
Asterisk is a trademark belonging to Digium, inc
-------------------------------------------------------------------------------
Asterisk is a trademark of Digium, Inc.