Various README updates
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README
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README
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The Asterisk(R) Open Source PBX
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by Mark Spencer <markster@digium.com>
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and the Asterisk.org developer community
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===============================================================================
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=== The Asterisk(R) Open Source PBX
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===
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=== by Mark Spencer <markster@digium.com>
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=== and the Asterisk.org developer community
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===
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=== Copyright (C) 2001-2008 Digium, Inc.
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=== and other copyright holders.
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===============================================================================
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Copyright (C) 2001-2006 Digium, Inc.
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and other copyright holders.
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================================================================
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-------------------------------------------------------------------------------
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--- SECURITY ------------------------------------------------------------------
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* SECURITY
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It is imperative that you read and fully understand the contents of
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the security information file (doc/security.txt) before you attempt
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to configure and run an Asterisk server.
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the security information document before you attempt to configure and run
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an Asterisk server.
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If you downloaded Asterisk as a tarball, see the security section in the PDF
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version of the documentation in doc/tex/asterisk.pdf. Alternatively, pull up
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the HTML version of the documentation in doc/tex/asterisk/index.html. The
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source for the security document is available in doc/tex/security.tex.
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-------------------------------------------------------------------------------
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-------------------------------------------------------------------------------
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--- WHAT IS ASTERISK ? --------------------------------------------------------
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* WHAT IS ASTERISK ?
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Asterisk is an Open Source PBX and telephony toolkit. It is, in a
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sense, middleware between Internet and telephony channels on the bottom,
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and Internet and telephony applications at the top. For more information
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on the project itself, please visit the Asterisk home page at:
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and Internet and telephony applications at the top. However, Asterisk supports
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more telephony interfaces than just Internet telephony. Asterisk also has a
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vast amount of support for traditional PSTN telephony, as well. For more
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information on the project itself, please visit the Asterisk home page at:
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http://www.asterisk.org
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In addition you'll find lots of information compiled by the Asterisk
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In addition you'll find lots of information compiled by the Asterisk
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community on this Wiki:
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http://www.voip-info.org/wiki-Asterisk
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There is a book on Asterisk published by O'Reilly under the
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Creative Commons License. It is available in book stores as well
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as in a downloadable version on the http://www.asteriskdocs.org
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web site.
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There is a book on Asterisk published by O'Reilly under the Creative Commons
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License. It is available in book stores as well as in a downloadable version on
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the http://www.asteriskdocs.org web site.
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-------------------------------------------------------------------------------
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* SUPPORTED OPERATING SYSTEMS
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-------------------------------------------------------------------------------
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--- SUPPORTED OPERATING SYSTEMS -----------------------------------------------
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== Linux ==
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--- Linux
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The Asterisk Open Source PBX is developed and tested primarily on the
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GNU/Linux operating system, and is supported on every major GNU/Linux
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distribution.
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== Others ==
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--- Others
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Asterisk has also been 'ported' and reportedly runs properly on other
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operating systems as well, including Sun Solaris, Apple's Mac OS X, and
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the BSD variants.
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-------------------------------------------------------------------------------
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* GETTING STARTED
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-------------------------------------------------------------------------------
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--- GETTING STARTED -----------------------------------------------------------
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First, be sure you've got supported hardware (but note that you don't need
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ANY special hardware, not even a soundcard) to install and run Asterisk.
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ANY special hardware, not even a sound card) to install and run Asterisk.
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Supported telephony hardware includes:
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* All Wildcard (tm) products from Digium (www.digium.com)
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* All Analog and Digital Interface cards from Digium (www.digium.com)
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* QuickNet Internet PhoneJack and LineJack (http://www.quicknet.net)
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* any full duplex sound card supported by ALSA or OSS
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* any full duplex sound card supported by ALSA, OSS, or PortAudio
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* any ISDN card supported by mISDN on Linux (BRI)
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* The Xorcom AstriBank channel bank
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* VoiceTronix OpenLine products
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* VoiceTronix OpenLine products
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The are several drivers for ISDN BRI cards available from third party sources.
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Check the voip-info.org wiki for more information on chan_capi and
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zaphfc.
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-------------------------------------------------------------------------------
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* UPGRADING FROM AN EARLIER VERSION
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-------------------------------------------------------------------------------
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--- UPGRADING FROM AN EARLIER VERSION -----------------------------------------
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If you are updating from a previous version of Asterisk, make sure you
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read the UPGRADE.txt file in the source directory. There are some files
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@ -67,29 +83,34 @@ and configuration options that you will have to change, even though we
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made every effort possible to maintain backwards compatibility.
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In order to discover new features to use, please check the configuration
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examples in the /configs directory of the source code distribution.
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To discover the major new features of Asterisk 1.2, please visit
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http://edvina.net/asterisk1-2/
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examples in the /configs directory of the source code distribution. For a
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list of new features in this version of Asterisk, see the CHANGES file.
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-------------------------------------------------------------------------------
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* NEW INSTALLATIONS
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-------------------------------------------------------------------------------
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--- NEW INSTALLATIONS ---------------------------------------------------------
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Ensure that your system contains a compatible compiler and development
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libraries. Asterisk requires either the GNU Compiler Collection (GCC) version
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3.0 or higher, or a compiler that supports the C99 specification and some of
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the gcc language extensions. In addition, your system needs to have the C
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library headers available, and the headers and libraries for OpenSSL,
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ncurses and zlib.
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On many distributions, these files are installed by packages with names like
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'glibc-devel', 'ncurses-devel', 'openssl-devel' and 'zlib-devel' or similar.
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library headers available, and the headers and libraries for ncurses.
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So let's proceed:
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There are many modules that have additional dependencies. To see what
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libraries are being looked for, see ./configure --help, or run
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"make menuselect" to view the dependencies for specific modules.
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On many distributions, these dependencies are installed by packages with names
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like 'glibc-devel', 'ncurses-devel', 'openssl-devel' and 'zlib-devel'
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or similar.
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So, let's proceed:
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1) Read this README file.
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There are more documents than this one in the doc/ directory.
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You may also want to check the configuration files that contain
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examples and reference guides. They are all in the configs/
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directory.
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There are more documents than this one in the doc/ directory. You may also
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want to check the configuration files that contain examples and reference
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guides. They are all in the configs/ directory.
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2) Run "./configure"
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@ -98,8 +119,8 @@ variables used during compilation.
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3) Run "make menuselect" [optional]
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This is needed if you want to select the modules that will be
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compiled and to check modules dependencies.
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This is needed if you want to select the modules that will be compiled and to
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check dependencies for various optional modules.
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4) Run "make"
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@ -107,21 +128,14 @@ compiled and to check modules dependencies.
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5) Run "make install"
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Each time you update or checkout from the repository, you are strongly
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encouraged to ensure all previous object files are removed to avoid internal
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inconsistency in Asterisk. Normally, this is automatically done with
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the presence of the file .cleancount, which increments each time a 'make clean'
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is required, and the file .lastclean, which contains the last .cleancount used.
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If this is your first time working with Asterisk, you may wish to install
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the sample PBX, with demonstration extensions, etc. If so, run:
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6) "make samples"
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Doing so will overwrite any existing config files you have.
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Doing so will overwrite any existing configuration files you have installed.
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Finally, you can launch Asterisk in the foreground mode (not a daemon)
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with:
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Finally, you can launch Asterisk in the foreground mode (not a daemon) with:
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# asterisk -vvvc
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@ -134,20 +148,22 @@ like this:
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You can type "help" at any time to get help with the system. For help
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with a specific command, type "help <command>". To start the PBX using
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your sound card, you can type "dial" to dial the PBX. Then you can use
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"answer", "hangup", and "dial" to simulate the actions of a telephone.
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Remember that if you don't have a full duplex sound card (and Asterisk
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will tell you somewhere in its verbose messages if you do/don't) then it
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won't work right (not yet).
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your sound card, you can type "console dial" to dial the PBX. Then you can use
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"console answer", "console hangup", and "console dial" to simulate the actions
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of a telephone. Remember that if you don't have a full duplex sound card
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(and Asterisk will tell you somewhere in its verbose messages if you do/don't)
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then it won't work right (not yet).
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"man asterisk" at the Unix/Linux command prompt will give you detailed
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information on how to start and stop Asterisk, as well as all the command
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line options for starting Asterisk.
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Feel free to look over the configuration files in /etc/asterisk, where
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you'll find a lot of information about what you can do with Asterisk.
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Feel free to look over the configuration files in /etc/asterisk, where you
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will find a lot of information about what you can do with Asterisk.
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-------------------------------------------------------------------------------
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* ABOUT CONFIGURATION FILES
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-------------------------------------------------------------------------------
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--- ABOUT CONFIGURATION FILES -------------------------------------------------
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All Asterisk configuration files share a common format. Comments are
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delimited by ';' (since '#' of course, being a DTMF digit, may occur in
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@ -163,7 +179,7 @@ asterisk. For example, in zapata.conf, one might specify:
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switchtype=national
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in order to indicate to Asterisk that the switch they are connecting to is
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In order to indicate to Asterisk that the switch they are connecting to is
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of the type "national". In general, the parameter will apply to
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instantiations which occur below its specification. For example, if the
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configuration file read:
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@ -174,7 +190,7 @@ configuration file read:
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switchtype = dms100
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channel => 25-47
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the "national" switchtype would be applied to channels one through
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The "national" switchtype would be applied to channels one through
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four and channels 10 through 12, whereas the "dms100" switchtype would
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apply to channels 25 through 47.
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@ -182,8 +198,10 @@ apply to channels 25 through 47.
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parameters. For example, the line "channel => 25-47" creates objects for
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the channels 25 through 47 of the card, obtaining the settings
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from the variables specified above.
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-------------------------------------------------------------------------------
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* SPECIAL NOTE ON TIME
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-------------------------------------------------------------------------------
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--- SPECIAL NOTE ON TIME ------------------------------------------------------
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Those using SIP phones should be aware that Asterisk is sensitive to
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large jumps in time. Manually changing the system time using date(1)
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@ -206,8 +224,10 @@ on UTC. UTC does not use daylight savings time.
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Also note that this issue is separate from the clocking of TDM
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channels, and is known to at least affect SIP registrations.
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-------------------------------------------------------------------------------
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* FILE DESCRIPTORS
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-------------------------------------------------------------------------------
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--- FILE DESCRIPTORS ----------------------------------------------------------
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Depending on the size of your system and your configuration,
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Asterisk can consume a large number of file descriptors. In UNIX,
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@ -220,11 +240,13 @@ everything from configuration information to voicemail storage.
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Most systems limit the number of file descriptors that Asterisk can
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have open at one time. This can limit the number of simultaneous
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calls that your system can handle. For example, if the limit is set
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at 1024 (a common default value) Asterisk can handle approxiately 150
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at 1024 (a common default value) Asterisk can handle approximately 150
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SIP calls simultaneously. To change the number of file descriptors
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follow the instructions for your system below:
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-------------------------------------------------------------------------------
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== PAM-based Linux System ==
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-------------------------------------------------------------------------------
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--- PAM-based Linux System ----------------------------------------------------
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If your system uses PAM (Pluggable Authentication Modules) edit
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/etc/security/limits.conf. Add these lines to the bottom of the file:
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@ -242,21 +264,29 @@ these changes to take effect.
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If there are no instructions specifically adapted to your system
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above you can try adding the command "ulimit -n 8192" to the script
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that starts Asterisk.
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-------------------------------------------------------------------------------
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* MORE INFORMATION
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-------------------------------------------------------------------------------
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--- MORE INFORMATION ----------------------------------------------------------
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See the doc directory for more documentation on various features. Again,
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please read all the configuration samples that include documentation on
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the configuration options.
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If this release of Asterisk was downloaded from a tarball, then some
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additional documentation should have been included.
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* doc/tex/asterisk.pdf --- PDF version of the documentation
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* doc/tex/asterisk/index.html --- HTML version of the documentation
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Finally, you may wish to visit the web site and join the mailing list if
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you're interested in getting more information.
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http://www.asterisk.org/support
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Welcome to the growing worldwide community of Asterisk users!
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-------------------------------------------------------------------------------
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Mark Spencer
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--- Mark Spencer, and the Asterisk.org development community
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----
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Asterisk is a trademark belonging to Digium, inc
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-------------------------------------------------------------------------------
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Asterisk is a trademark of Digium, Inc.
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