diff --git a/README b/README.md similarity index 56% rename from README rename to README.md index a1ccf2c802c60ac5e18389ea9c9b14d2807ef224..4ed9b3eec2aa6138ba5d86cfe4857a7cff233cfb 100644 --- a/README +++ b/README.md @@ -1,15 +1,9 @@ -=============================================================================== -=== The Asterisk(R) Open Source PBX -=== -=== by Mark Spencer <markster@digium.com> -=== and the Asterisk.org developer community -=== -=== Copyright (C) 2001-2016 Digium, Inc. -=== and other copyright holders. -=============================================================================== - -------------------------------------------------------------------------------- ---- SECURITY ------------------------------------------------------------------ +# The Asterisk(R) Open Source PBX +```text + By Mark Spencer <markster@digium.com> and the Asterisk.org developer community. + Copyright (C) 2001-2016 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 @@ -19,80 +13,62 @@ an Asterisk server. 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. 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: +vast amount of support for traditional PSTN telephony, as well. - https://www.asterisk.org - - The official Asterisk wiki can be found at: - - https://wiki.asterisk.org - - In addition you'll find lots of information compiled by the Asterisk -community on this Wiki: - - https://www.voip-info.org/wiki-Asterisk + For more information on the project itself, please visit the Asterisk +[home page] and the official [wiki]. In addition you'll find lots +of information compiled by the Asterisk community at [voip-info.org]. 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. -------------------------------------------------------------------------------- +the [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, Cygwin, 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 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 - -------------------------------------------------------------------------------- +Supported telephony hardware includes: +* All Analog and Digital Interface cards from [Digium] +* 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 ----------------------------------------- +### 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 +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. -------------------------------------------------------------------------------- +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 @@ -101,55 +77,54 @@ 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. +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: +So, let's proceed: +1. Read this file. -1) Read this README file. - - There are more documents than this one in the doc/ directory. You may also + 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. +guides in the [configs] directory. -2) Run "./configure" +2. Run `./configure` Execute the configure script to guess values for system-dependent variables used during compilation. -3) Run "make menuselect" [optional] +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" +4. Run `make` - Assuming the build completes successfully: +Assuming the build completes successfully: -5) Run "make install" +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" +6. Run `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 - +7. 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> - +``` + *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 @@ -162,12 +137,10 @@ then it won't work right (not yet). 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 + 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 @@ -179,20 +152,21 @@ 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: - +asterisk. For example, in [chan_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 @@ -202,16 +176,14 @@ 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) (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 +by itself use [NTP] 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 @@ -228,10 +200,8 @@ 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, @@ -247,50 +217,56 @@ 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 ---------------------------------------------------- +#### 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: - +`/etc/security/limits.conf`. Add these lines to the bottom of the file: +```text 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 == +#### 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 +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. + 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 [support] site and join the [mailing +list] if you're interested in getting more information. - Finally, you may wish to visit the web site and join the mailing list if -you're interested in getting more information. +Welcome to the growing worldwide community of Asterisk users! +``` + Mark Spencer, and the Asterisk.org development community +``` - https://www.asterisk.org/support +--- - Welcome to the growing worldwide community of Asterisk users! -------------------------------------------------------------------------------- +Asterisk is a trademark of Digium, Inc. ---- Mark Spencer, and the Asterisk.org development community +[home page]: https://www.asterisk.org +[support]: https://www.asterisk.org/support +[wiki]: https://wiki.asterisk.org/ +[mailing list]: http://lists.digium.com/mailman/listinfo/asterisk-users +[chan_dahdi.conf]: configs/samples/chan_dahdi.conf.sample +[voip-info.org]: http://www.voip-info.org/wiki-Asterisk +[asteriskdocs.org]: http://www.asteriskdocs.org +[NTP]: http://www.ntp.org/ +[Digium]: https://www.digium.com/ +[UPGRADE.txt]: UPGRADE.txt +[CHANGES]: CHANGES +[configs]: configs +[doc]: doc -------------------------------------------------------------------------------- -Asterisk is a trademark of Digium, Inc.