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	Update README, CHANGES, and Makefile. Direct users to http://wiki.asterisk.org for documentation or to the AST.txt and AST.pdf included in the tarball. (closes issue #18443) Reported by: bas Patches: changes.diff uploaded by lathama (license 1028) readme.diff uploaded by lathama (license 1028) Tested by: lathama bas git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@305560 65c4cc65-6c06-0410-ace0-fbb531ad65f3
		
			
				
	
	
		
			291 lines
		
	
	
		
			13 KiB
		
	
	
	
		
			Plaintext
		
	
	
	
	
	
			
		
		
	
	
			291 lines
		
	
	
		
			13 KiB
		
	
	
	
		
			Plaintext
		
	
	
	
	
	
| ===============================================================================
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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) 2001-2009 Digium, Inc.
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| ===                       and other copyright holders.
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| ===============================================================================
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| 
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| -------------------------------------------------------------------------------
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| --- SECURITY ------------------------------------------------------------------
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| 
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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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| an Asterisk server.
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| 
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|   If you downloaded Asterisk as a tarball, see the security section in the PDF
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| or text version of the documentation in doc/AST.pdf or doc/AST.txt.
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| -------------------------------------------------------------------------------
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| 
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| -------------------------------------------------------------------------------
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| --- WHAT IS ASTERISK ? --------------------------------------------------------
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| 
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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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| 
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|            http://www.asterisk.org
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| 
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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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| 
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|            http://www.voip-info.org/wiki-Asterisk
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| 
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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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| 
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| -------------------------------------------------------------------------------
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| --- SUPPORTED OPERATING SYSTEMS -----------------------------------------------
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| 
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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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| 
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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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| -------------------------------------------------------------------------------
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| --- GETTING STARTED -----------------------------------------------------------
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| 
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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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| 
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|   Supported telephony hardware includes:
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| 
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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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| 
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| -------------------------------------------------------------------------------
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| --- UPGRADING FROM AN EARLIER VERSION -----------------------------------------
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| 
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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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| 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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| 
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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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| 
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| -------------------------------------------------------------------------------
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| --- NEW INSTALLATIONS ---------------------------------------------------------
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| 
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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 ncurses.
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| 
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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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| 
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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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| 
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|   So, let's proceed:
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| 
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| 1) Read this README file.
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| 
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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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| 
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| 2) Run "./configure"
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| 
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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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| 
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| 3) Run "make menuselect" [optional]
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| 
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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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| 
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| 4) Run "make"
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| 
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|   Assuming the build completes successfully:
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| 
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| 5) Run "make install"
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| 
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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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| 
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| 6) "make samples"
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| 
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|   Doing so will overwrite any existing configuration files you have installed.
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| 
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|   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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| 
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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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| "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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| 
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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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| 
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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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| -------------------------------------------------------------------------------
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| --- ABOUT CONFIGURATION FILES -------------------------------------------------
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| 
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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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| many places).  A configuration file is divided into sections whose names
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| appear in []'s.  Each section typically contains two types of statements,
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| those of the form 'variable = value', and those of the form 'object =>
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| parameters'.  Internally the use of '=' and '=>' is exactly the same, so 
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| 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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| 
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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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| 
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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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| apply to channels 25 through 47.
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|   
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|   The "object => parameters" instantiates an object with the given
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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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| -------------------------------------------------------------------------------
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| --- SPECIAL NOTE ON TIME ------------------------------------------------------
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|   
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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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| 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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| causing discontinuities. Most Linux distributions include precompiled
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| versions of NTP.  Beware of some time synchronization methods that get
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| the correct real time periodically and then manually set the system
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| clock.
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| 
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|   Apparent time changes due to daylight savings time are just that,
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| apparent.  The use of daylight savings time in a Linux system is
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| purely a user interface issue and does not affect the operation of the
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| Linux kernel or Asterisk.  The system clock on Linux kernels operates
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| on UTC.  UTC does not use daylight savings time.
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| 
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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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| -------------------------------------------------------------------------------
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| --- FILE DESCRIPTORS ----------------------------------------------------------
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| 
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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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| file descriptors are used for more than just files on disk.  File
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| descriptors are also used for handling network communication
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| (e.g. SIP, IAX2, or H.323 calls) and hardware access (e.g. analog and
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| digital trunk hardware).  Asterisk accesses many on-disk files for
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| everything from configuration information to voicemail storage.
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| 
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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 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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| -------------------------------------------------------------------------------
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| --- PAM-based Linux System ----------------------------------------------------
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| 
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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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| 
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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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| 
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| == Generic UNIX System ==
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| 
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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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| 
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| -------------------------------------------------------------------------------
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| --- MORE INFORMATION ----------------------------------------------------------
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| 
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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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| 
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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/AST.pdf --- PDF version of the documentation
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|      * doc/AST.txt --- Text version of the documentation
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| 
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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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| 
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|    http://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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| 
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| --- Mark Spencer, and the Asterisk.org development community
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| 
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| -------------------------------------------------------------------------------
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| Asterisk is a trademark of Digium, Inc.
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