diff options
author | Corey Farrell <git@cfware.com> | 2017-11-21 10:16:24 -0500 |
---|---|---|
committer | Corey Farrell <git@cfware.com> | 2017-11-21 10:17:28 -0500 |
commit | c6e1e6e9682964baf8f97fb4bbea4b655a1b8b0a (patch) | |
tree | 4fdc03012a3c57f7607a1aecfb23161675879796 /README | |
parent | 0fd8db7ec24abfd84a4d9a87b7e677b06bf83e39 (diff) |
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
Diffstat (limited to 'README')
-rw-r--r-- | README | 296 |
1 files changed, 0 insertions, 296 deletions
diff --git a/README b/README deleted file mode 100644 index 871ae962a..000000000 --- a/README +++ /dev/null @@ -1,296 +0,0 @@ -=============================================================================== -=== 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: - - 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 - - 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. - - https://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. |