123456789101112131415161718192021222324252627282930313233343536373839404142434445464748495051525354555657585960616263646566676869707172737475767778798081828384858687888990919293949596979899100101102103104105106107108109110111112113114115116117118119120121122123124125126127128129130131132133134135136137138139140141142143144145146147148149150151152153154155156157158159160161162163164165166167168169170171172173174175176177178179180181182183184185186187188189190191192193194195196197198199200201202203204205206207208209210211212213214215216217218219220221222223224225226227228229230231232233234235236237238239240241242243244245246247248249250251252253254255256257258259260261262263264265266267268269270271272273274275276277278279280281282283284285286287288289290291292293294295296297 |
- ===============================================================================
- === 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:
- http://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:
- 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.
- -------------------------------------------------------------------------------
- -------------------------------------------------------------------------------
- --- 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.
- http://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.
|