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README: Convert to README.md.

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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
This commit is contained in:
Corey Farrell
2017-11-17 22:33:04 -05:00
parent 3abf8dfeae
commit 90f9885f73
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README → README.md
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@@ -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:
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
* 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 -----------------------------------------
### 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!
-------------------------------------------------------------------------------
--- Mark Spencer, and the Asterisk.org development community
-------------------------------------------------------------------------------
Asterisk is a trademark of Digium, Inc.
[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