Various README updates

git-svn-id: http://svn.digium.com/svn/asterisk/trunk@99166 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 102 insertions, 72 deletions

diff --git a/README b/README
index 176bfca44..eed22e5c1 100644
--- a/README
+++ b/README
@@ -1,65 +1,81 @@
-The Asterisk(R) Open Source PBX
-by Mark Spencer <markster@digium.com>
-and the Asterisk.org developer community
+===============================================================================
+===                     The Asterisk(R) Open Source PBX
+===
+===                   by Mark Spencer <markster@digium.com>
+===                  and the Asterisk.org developer community
+===
+===                    Copyright (C) 2001-2008 Digium, Inc.
+===                       and other copyright holders.
+===============================================================================
+
+-------------------------------------------------------------------------------
+--- SECURITY ------------------------------------------------------------------
 
-Copyright (C) 2001-2006 Digium, Inc.
-and other copyright holders.
-================================================================
-
-* SECURITY
   It is imperative that you read and fully understand the contents of
-the security information file (doc/security.txt) before you attempt 
-to configure and run an Asterisk server.
+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 ? --------------------------------------------------------
 
-* 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.  For more information
-on the project itself, please visit the Asterisk home page at:
+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:
 
            http://www.asterisk.org
 
-In addition you'll find lots of information compiled by the Asterisk
+  In addition you'll find lots of information compiled by the Asterisk
 community on this Wiki:
 
            http://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.
+  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
+-------------------------------------------------------------------------------
+--- 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, 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 soundcard) to install and run Asterisk.
+ANY special hardware, not even a sound card) to install and run Asterisk.
 
   Supported telephony hardware includes:
 
-	* All Wildcard (tm) products from Digium (www.digium.com)
+	* 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 or OSS
+	* any full duplex sound card supported by ALSA, OSS, or PortAudio
 	* any ISDN card supported by mISDN on Linux (BRI)
 	* The Xorcom AstriBank channel bank
-        * VoiceTronix OpenLine products
+	* VoiceTronix OpenLine products
 
-The are several drivers for ISDN BRI cards available from third party sources.
-Check the voip-info.org wiki for more information on chan_capi and 
-zaphfc.
+-------------------------------------------------------------------------------
 
-* 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
@@ -67,29 +83,34 @@ 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. 
-To discover the major new features of Asterisk 1.2, please visit 
-http://edvina.net/asterisk1-2/
+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
 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 OpenSSL,
-ncurses and zlib.
-On many distributions, these files are installed by packages with names like
-'glibc-devel', 'ncurses-devel', 'openssl-devel' and 'zlib-devel' or similar.
+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.
 
-  So let's proceed:
+  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.
+  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"
 
@@ -98,8 +119,8 @@ 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 modules dependencies.
+  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"
 
@@ -107,21 +128,14 @@ compiled and to check modules dependencies.
 
 5) Run "make install"
 
-  Each time you update or checkout from the repository, you are strongly
-encouraged to ensure all previous object files are removed to avoid internal 
-inconsistency in Asterisk. Normally, this is automatically done with 
-the presence of the file .cleancount, which increments each time a 'make clean'
-is required, and the file .lastclean, which contains the last .cleancount used. 
-
   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 config files you have.
+  Doing so will overwrite any existing configuration files you have installed.
 
-  Finally, you can launch Asterisk in the foreground mode (not a daemon)
-with:
+  Finally, you can launch Asterisk in the foreground mode (not a daemon) with:
 
 # asterisk -vvvc
 
@@ -134,20 +148,22 @@ like this:
 
   You can type "help" at any time to get help with the system.  For help
 with a specific command, type "help <command>".  To start the PBX using
-your sound card, you can type "dial" to dial the PBX.  Then you can use
-"answer", "hangup", and "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).
+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'll find a lot of information about what you can do with 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
+-------------------------------------------------------------------------------
+--- 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
@@ -163,7 +179,7 @@ asterisk.  For example, in zapata.conf, one might specify:
 
 	switchtype=national
 
-in order to indicate to Asterisk that the switch they are connecting to is
+  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:
@@ -174,7 +190,7 @@ configuration file read:
 	switchtype = dms100
 	channel => 25-47
 
-the "national" switchtype would be applied to channels one through
+  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.
   
@@ -182,8 +198,10 @@ 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)
@@ -206,8 +224,10 @@ 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,
@@ -220,11 +240,13 @@ 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 approxiately 150
+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:
@@ -242,21 +264,29 @@ these changes to take effect.
   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
+-------------------------------------------------------------------------------
+--- 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.
 
    http://www.asterisk.org/support
 
   Welcome to the growing worldwide community of Asterisk users!
+-------------------------------------------------------------------------------
 
-Mark Spencer
+--- Mark Spencer, and the Asterisk.org development community
 
-----
-Asterisk is a trademark belonging to Digium, inc
+-------------------------------------------------------------------------------
+Asterisk is a trademark of Digium, Inc.