Asterisk.org: Developer Documentation (28 Oct 2009)

Wed Oct 28 15:49:24 2009

## Asterisk developer's documentation

# README - the general administrator introduction

    The Asterisk Open Source PBX
    by Mark Spencer <markster@digium.com>
    and the Asterisk.org developer community

    Copyright (C) 2001-2005 Digium, Inc.
    and other copyright holders.
    ================================================================

    * SECURITY
      It is imperative that you read and fully understand the contents of
    the SECURITY file before you attempt to configure and run an Asterisk
    server.

    * 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:

               http://www.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, 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 soundcard) to install and run Asterisk.

      Supported telephony hardware includes:

    	* All Wildcard (tm) products from Digium (www.digium.com)
    	* QuickNet Internet PhoneJack and LineJack (http://www.quicknet.net)
    	* any full duplex sound card supported by ALSA or OSS
            * 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, chan_misdn and
    zaphfc.

    * UPGRADING FROM VERSION 1.0

      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.
    To discover the major new features of Asterisk 1.2, please visit
    http://edvina.net/asterisk1-2/

    * 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.

      So let's proceed:

    1) Run "make"

      Assuming the build completes successfully:

    2) Run "make install"

      Each time you update or checkout from CVS, 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:

    3) "make samples"

      Doing so will overwrite any existing config files you have.

      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 "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).

      "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.

    * 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 zapata.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 approxiately 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.

      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