README: Convert to README.md.
Convert the README file to markdown format, remove the old README. This causes websites like github to display the README in a much nicer format with live links. The raw file is still very readable from plain text editors and terminals. Change-Id: I7d13131764a9a9026e5f8a6ddb245a01bbd788e7
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Corey Farrell
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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) 2001-2016 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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# The Asterisk(R) Open Source PBX
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```text
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By Mark Spencer <markster@digium.com> and the Asterisk.org developer community.
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Copyright (C) 2001-2016 Digium, Inc. and other copyright holders.
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```
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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 document before you attempt to configure and run
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@ -19,80 +13,62 @@ an Asterisk server.
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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. 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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vast amount of support for traditional PSTN telephony, as well.
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https://www.asterisk.org
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The official Asterisk wiki can be found at:
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https://wiki.asterisk.org
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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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https://www.voip-info.org/wiki-Asterisk
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For more information on the project itself, please visit the Asterisk
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[home page] and the official [wiki]. In addition you'll find lots
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of information compiled by the Asterisk community at [voip-info.org].
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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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the [asteriskdocs.org] web site.
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-------------------------------------------------------------------------------
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--- SUPPORTED OPERATING SYSTEMS -----------------------------------------------
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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, Cygwin,
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and the BSD variants.
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-------------------------------------------------------------------------------
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-------------------------------------------------------------------------------
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--- GETTING STARTED -----------------------------------------------------------
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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 sound card) to install and run Asterisk.
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Supported telephony hardware includes:
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Supported telephony hardware includes:
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* All Analog and Digital Interface cards from [Digium]
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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, OSS, or PortAudio
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* any ISDN card supported by mISDN on Linux
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* The Xorcom Astribank channel bank
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* VoiceTronix OpenLine products
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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, OSS, or PortAudio
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* any ISDN card supported by mISDN on Linux
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* The Xorcom Astribank channel bank
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* VoiceTronix OpenLine products
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-------------------------------------------------------------------------------
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-------------------------------------------------------------------------------
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--- UPGRADING FROM AN EARLIER VERSION -----------------------------------------
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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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read the [UPGRADE.txt] file in the source directory. There are some files
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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. 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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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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### 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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@ -101,55 +77,54 @@ 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 ncurses.
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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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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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So, let's proceed:
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1. Read this file.
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1) Read this README file.
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There are more documents than this one in the doc/ directory. You may also
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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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guides in the [configs] directory.
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2) Run "./configure"
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2. Run `./configure`
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Execute the configure script to guess values for system-dependent
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variables used during compilation.
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3) Run "make menuselect" [optional]
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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 compiled and to
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check dependencies for various optional modules.
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4) Run "make"
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4. Run `make`
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Assuming the build completes successfully:
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Assuming the build completes successfully:
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5) Run "make install"
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5. Run `make install`
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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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6. Run `make samples`
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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) with:
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# asterisk -vvvc
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7. Finally, you can launch Asterisk in the foreground mode (not a daemon) with:
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```
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# asterisk -vvvc
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```
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You'll see a bunch of verbose messages fly by your screen as Asterisk
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initializes (that's the "very very verbose" mode). When it's ready, if
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you specified the "c" then you'll get a command line console, that looks
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like this:
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*CLI>
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```
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*CLI>
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```
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You can type "core show help" at any time to get help with the system. For help
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with a specific command, type "core show help <command>". To start the PBX using
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your sound card, you can type "console dial" to dial the PBX. Then you can use
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@ -162,12 +137,10 @@ then it won't work right (not yet).
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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 you
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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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-------------------------------------------------------------------------------
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--- ABOUT CONFIGURATION FILES -------------------------------------------------
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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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@ -179,20 +152,21 @@ they're used only to help make the configuration file easier to
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understand, and do not affect how it is actually parsed.
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Entries of the form 'variable=value' set the value of some parameter in
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asterisk. For example, in dahdi.conf, one might specify:
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asterisk. For example, in [chan_dahdi.conf], one might specify:
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```
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switchtype=national
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```
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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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```
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switchtype = national
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channel => 1-4
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channel => 10-12
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switchtype = dms100
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channel => 25-47
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```
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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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@ -202,16 +176,14 @@ 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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-------------------------------------------------------------------------------
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--- SPECIAL NOTE ON TIME ------------------------------------------------------
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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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(or other similar commands) may cause SIP registrations and other
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internal processes to fail. If your system cannot keep accurate time
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by itself use NTP (http://www.ntp.org/) to keep the system clock
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by itself use [NTP] to keep the system clock
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synchronized to "real time". NTP is designed to keep the system clock
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synchronized by speeding up or slowing down the system clock until it
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is synchronized to "real time" rather than by jumping the time and
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@ -228,10 +200,8 @@ 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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-------------------------------------------------------------------------------
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--- FILE DESCRIPTORS ----------------------------------------------------------
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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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@ -247,50 +217,56 @@ 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 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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-------------------------------------------------------------------------------
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--- PAM-based Linux System ----------------------------------------------------
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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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`/etc/security/limits.conf`. Add these lines to the bottom of the file:
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```text
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root soft nofile 4096
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root hard nofile 8196
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asterisk soft nofile 4096
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asterisk hard nofile 8196
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```
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(adjust the numbers to taste). You may need to reboot the system for
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these changes to take effect.
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== Generic UNIX System ==
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#### GENERIC UNIX SYSTEM
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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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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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-------------------------------------------------------------------------------
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--- MORE INFORMATION ----------------------------------------------------------
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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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See the [doc] directory for more documentation on various features.
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Again, please read all the configuration samples that include documentation
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on 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 [support] site and join the [mailing
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list] if you're interested in getting more information.
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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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Welcome to the growing worldwide community of Asterisk users!
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```
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Mark Spencer, and the Asterisk.org development community
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```
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https://www.asterisk.org/support
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---
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Welcome to the growing worldwide community of Asterisk users!
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-------------------------------------------------------------------------------
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--- Mark Spencer, and the Asterisk.org development community
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-------------------------------------------------------------------------------
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Asterisk is a trademark of Digium, Inc.
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[home page]: https://www.asterisk.org
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[support]: https://www.asterisk.org/support
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[wiki]: https://wiki.asterisk.org/
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[mailing list]: http://lists.digium.com/mailman/listinfo/asterisk-users
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[chan_dahdi.conf]: configs/samples/chan_dahdi.conf.sample
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[voip-info.org]: http://www.voip-info.org/wiki-Asterisk
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[asteriskdocs.org]: http://www.asteriskdocs.org
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[NTP]: http://www.ntp.org/
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[Digium]: https://www.digium.com/
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[UPGRADE.txt]: UPGRADE.txt
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[CHANGES]: CHANGES
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[configs]: configs
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[doc]: doc
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