git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@12492 65c4cc65-6c06-0410-ace0-fbb531ad65f3
243 lines
9.5 KiB
Text
243 lines
9.5 KiB
Text
The Asterisk Open Source PBX
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by Mark Spencer <markster@digium.com>
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and the Asterisk.org developer community
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Copyright (C) 2001-2005 Digium, Inc.
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and other copyright holders.
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================================================================
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* SECURITY
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It is imperative that you read and fully understand the contents of
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the SECURITY file before you attempt to configure and run an Asterisk
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server.
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* WHAT IS ASTERISK ?
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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. For more information
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on the project itself, please visit the Asterisk home page at:
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http://www.asterisk.org
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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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http://www.voip-info.org/wiki-Asterisk
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There is a book on Asterisk published by O'Reilly under the
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Creative Commons License. It is available in book stores as well
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as in a downloadable version on the http://www.asteriskdocs.org
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web site.
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* SUPPORTED OPERATING SYSTEMS
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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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== 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, and
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the BSD variants.
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* GETTING STARTED
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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 soundcard) to install and run Asterisk.
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Supported telephony hardware includes:
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* All Wildcard (tm) products 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 or OSS
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* any ISDN card supported by mISDN on Linux (BRI)
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* The Xorcom AstriBank channel bank
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* VoiceTronix OpenLine products
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The are several drivers for ISDN BRI cards available from third party sources.
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Check the voip-info.org wiki for more information on chan_capi and
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zaphfc.
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* UPGRADING FROM AN EARLIER VERSION
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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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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.
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To discover the major new features of Asterisk 1.2, please visit
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http://edvina.net/asterisk1-2/
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* NEW INSTALLATIONS
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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 OpenSSL,
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ncurses and zlib.
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On many distributions, these files are installed by packages with names like
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'glibc-devel', 'ncurses-devel', 'openssl-devel' and 'zlib-devel' or similar.
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So let's proceed:
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1) Run "make"
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Assuming the build completes successfully:
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2) Run "make install"
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Each time you update or checkout from the repository, you are strongly
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encouraged to ensure all previous object files are removed to avoid internal
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inconsistency in Asterisk. Normally, this is automatically done with
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the presence of the file .cleancount, which increments each time a 'make clean'
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is required, and the file .lastclean, which contains the last .cleancount used.
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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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3) "make samples"
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Doing so will overwrite any existing config files you have.
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Finally, you can launch Asterisk in the foreground mode (not a daemon)
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with:
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# asterisk -vvvc
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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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*CLI>
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You can type "help" at any time to get help with the system. For help
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with a specific command, type "help <command>". To start the PBX using
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your sound card, you can type "dial" to dial the PBX. Then you can use
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"answer", "hangup", and "dial" to simulate the actions of a telephone.
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Remember that if you don't have a full duplex sound card (and Asterisk
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will tell you somewhere in its verbose messages if you do/don't) then it
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won't work right (not yet).
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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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Feel free to look over the configuration files in /etc/asterisk, where
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you'll find a lot of information about what you can do with Asterisk.
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* ABOUT CONFIGURATION FILES
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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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Entries of the form 'variable=value' set the value of some parameter in
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asterisk. For example, in zapata.conf, one might specify:
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switchtype=national
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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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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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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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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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* SPECIAL NOTE ON TIME
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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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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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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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* FILE DESCRIPTORS
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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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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 approxiately 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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== PAM-based Linux System ==
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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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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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(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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== Generic UNIX System ==
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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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* MORE INFORMATION
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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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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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http://www.asterisk.org/support
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Welcome to the growing worldwide community of Asterisk users!
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Mark Spencer
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