A git mirror of http://svn.asterisk.org/svn/asterisk . May lag a few hours behind. Mirrors /branches (and /trunk ). Includes tags for /tags . Does not include /team . See also it's web interface: http://svnview.digium.com/svn/asterisk . http://asterisk.org/

Matthew Jordan e5ce6db4d1 clang compiler warnings: Fix various warnings for tests 7 years ago
addons 4d1ad37eae Fix typo's (retrieve, specified, address). 8 years ago
agi 800a7d7a52 Title update 10 years ago
apps fc26140240 apps/app_queue: Prevent possible crash when evaluating queue penalty rules 7 years ago
autoconf 10d8edf277 Add missing file from previous commit. 8 years ago
bridges dcb80bd0fd chan_pjsip/res_pjsip/bridge_softmix/core: Improve translation path choices. 7 years ago
build_tools 50c246c214 Tell menuselect that MALLOC_DEBUG conflicts with DEBUG_CHAOS. 7 years ago
cdr bc8f2f2537 AMI: Add documentation for the missing Cdr/CEL events. 8 years ago
cel b1f5a69645 cel_pgsl: Add support for GMT timestamps 7 years ago
channels dcb80bd0fd chan_pjsip/res_pjsip/bridge_softmix/core: Improve translation path choices. 7 years ago
codecs 288a32f926 clang compiler warnings: Fix a variety of "unused" warnings 7 years ago
configs 7008c6c0ca chan_sip: make progressinband default to no 7 years ago
contrib 56fe3dceb8 res_pjsip: Add an 'auto' option for DTMF Mode 7 years ago
doc 5d243883a4 docs: Escape unescaped minus sign in asterisk.8 manpage. 8 years ago
formats f7524710d5 clang compiler warnings: Fix -Wself-assign 7 years ago
funcs 62c4eef263 clang compiler warnings: Fix autological comparisons 7 years ago
images e1298a8b96 even uglier gui with more buttons 14 years ago
include dcb80bd0fd chan_pjsip/res_pjsip/bridge_softmix/core: Improve translation path choices. 7 years ago
main dcb80bd0fd chan_pjsip/res_pjsip/bridge_softmix/core: Improve translation path choices. 7 years ago
menuselect faaad16627 Menuselect: Fix incorrect enabling on failed deps 8 years ago
pbx a6da3fe169 clang compiler warnings: Fix sometimes-uninitialized warning in pbx_config 7 years ago
phoneprov b3116a5888 res_phoneprov add snom 300, 320, 360, 370, 820, 821, 870 support 12 years ago
res 7678fd6051 res/res_pjsip_t38: Add missing initialization of t38faxmaxdatagram 7 years ago
rest-api 8476197a7c res/ari: Fix model validation for ChannelHold event 7 years ago
rest-api-templates 5fc15939c2 ARI: Improve wiki documentation 8 years ago
sounds 9baf14c16d Sounds/BuildSystem: Modifications to include new releases and Japanese language. 8 years ago
static-http 32b2da6e1b Add licens/copyright header 10 years ago
tests e5ce6db4d1 clang compiler warnings: Fix various warnings for tests 7 years ago
utils d7940e9bb6 clang compiler warnings: Remove large chunks of unused code from extconf 7 years ago
.cleancount dcbb781867 Remove obsolete struct ast_channel note. 10 years ago
BSDmakefile e998fb2c63 Merged revisions 285090 via svnmerge from 12 years ago
BUGS e53ea0e259 Add UPGRADE-1.10.txt file from UPGRADE.txt. 11 years ago
CHANGES 7008c6c0ca chan_sip: make progressinband default to no 7 years ago
COPYING 70a40d92e3 17 years ago
CREDITS 0d85116afc res_pjsip: Add PJSIP CLI commands 9 years ago
LICENSE 4c7311bf48 LICENSE: Clarify language in Asterisk's LICENSE to allow for linking to UniMRCP 8 years ago
Makefile a7af4687eb Add support for the clang compiler; update RAII_VAR to use BlocksRuntime 8 years ago
Makefile.moddir_rules 3b5ae179ab Doxygen Updates - Title update 10 years ago
Makefile.rules c7b09e120e clang compiler warnings: Ignore -Wunused-command-line-argument 7 years ago
README 409e0bfece Start work on documentation janitor project with a little commit. This adds a link to the Asterisk wiki at https://wiki.asterisk.org to the README file. 10 years ago
README-SERIOUSLY.bestpractices.txt 001142ce53 security: Inhibit execution of privilege escalating functions 9 years ago
README-addons.txt 358628a848 Move Asterisk-addons modules into the main Asterisk source tree. 13 years ago
UPGRADE-1.2.txt 1224d7ea74 as suggested by jtodd, document the purposes of the CHANGES and UPGRADE files 14 years ago
UPGRADE-1.4.txt 041d524be0 Convert this branch to Opsound music-on-hold. 13 years ago
UPGRADE-1.6.txt b791cd73a4 Update UPGRADE-1.6.txt stating insecure=very has been removed. 12 years ago
UPGRADE-1.8.txt 6d1c84e3db chan_sip: Note change in behavior to how directmediapermit/deny ACL works 10 years ago
UPGRADE-10.txt 4166b565ba Add comments about the BUILD_NATIVE change 10 years ago
UPGRADE-11.txt bb3f17f2f1 Fix UPGRADE.txt Due To Merging From Branch 11 9 years ago
UPGRADE-12.txt 12c495a65a Added note to UPGRADE.txt about the default value of live_dangerously changing 9 years ago
UPGRADE-13.txt 89cc48e597 Added ConfBridge AMI event note to UPGRADE.txt. 8 years ago
UPGRADE.txt 82fa2c3b70 Logger: Convert 'struct ast_callid' to unsigned int. 7 years ago
Zaptel-to-DAHDI.txt 904fefe038 Merged revisions 137679 via svnmerge from 14 years ago
bootstrap.sh 926291b6c3 Make sure asterisk builds on OpenBSD 11 years ago
config.guess 39c1f63772 Update config.guess and config.sub 8 years ago
config.sub 39c1f63772 Update config.guess and config.sub 8 years ago
configure c9e46da27f dns: Add core DNS API + unit tests and res_resolver_unbound module + unit tests. 7 years ago
configure.ac c9e46da27f dns: Add core DNS API + unit tests and res_resolver_unbound module + unit tests. 7 years ago
default.exports 9cd50e4a5b Add _IO_stdin_used in version-script to fix SIGBUSes on Sparc. 9 years ago
install-sh a13e072bec silly people that don't want to install/run autoconf :-) 16 years ago
makeopts.in c9e46da27f dns: Add core DNS API + unit tests and res_resolver_unbound module + unit tests. 7 years ago
missing a13e072bec silly people that don't want to install/run autoconf :-) 16 years ago
mkinstalldirs a13e072bec silly people that don't want to install/run autoconf :-) 16 years ago
sample.call 4739d48ca4 Merged revisions 299138 via svnmerge from 12 years ago

README

===============================================================================
=== The Asterisk(R) Open Source PBX
===
=== by Mark Spencer
=== 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 ". 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.