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/

David M. Lee 3ea69b7c78 security: Inhibit execution of privilege escalating functions пре 11 година
addons 09e25d9974 reject call attempts when gatekeeper is configured but not registered пре 11 година
agi 72b6ef1e90 Merged revisions 338227 via svnmerge from пре 13 година
apps b64308962d app_sms: BufferOverflow when receiving odd length 16 bit message пре 11 година
autoconf e6330e70ed Fix compile problem when old version of libvorbisfile v1.1.2 is used. пре 12 година
bridges 3484c58fcc Refactor ast_timer_ack to return an error and handle the error in timer users пре 12 година
build_tools c931c7bbec build_tools: Allow Asterisk to report git SHAs in version string. пре 12 година
cdr ce34b8de22 Fix misuses of asprintf throughout the code. пре 12 година
cel 56957ae835 Fix memory leak when CEL is successfully written to PostgreSQL database пре 12 година
channels 7c5a9e98b6 AST-2013-005: Fix crash caused by invalid SDP пре 11 година
codecs a20e459ebb codec_dahdi: Fix output of "transcoder show" CLI command. пре 12 година
configs 3ea69b7c78 security: Inhibit execution of privilege escalating functions пре 11 година
contrib d9274dbf65 Update init.d scripts to handle stderr; readd splash screen for remote consoles пре 12 година
doc c530108a3b Restore CODING-GUIDELINES to doc folder пре 12 година
formats fba1d8943d Fix error that caused truncate operations to fail пре 12 година
funcs 3ea69b7c78 security: Inhibit execution of privilege escalating functions пре 11 година
images e1298a8b96 even uglier gui with more buttons пре 16 година
include 3ea69b7c78 security: Inhibit execution of privilege escalating functions пре 11 година
main 3ea69b7c78 security: Inhibit execution of privilege escalating functions пре 11 година
pbx 256d0b7a41 Remove unnecessary channel module references. пре 12 година
phoneprov b3116a5888 res_phoneprov add snom 300, 320, 360, 370, 820, 821, 870 support пре 14 година
res 8a6ae411e3 Prevent exhaustion of system resources through exploitation of event cache пре 12 година
sounds d8c0388a13 Incremented EXTRA_SOUNDS_VERSION in sounds/Makefile to 1.4.12 for new Extra Sounds releases пре 12 година
static-http bc8f9d2651 Merged revisions 291575 via svnmerge from пре 14 година
tests 1b2a6f63b1 Migrate hashtest/hashtest2 to be unit tests. пре 12 година
utils 3f94b51bf2 Fixed extconf.c breakage introduced in r376306. пре 12 година
.cleancount 28bc09082a Fix stuck DTMF when bridge is broken. пре 12 година
BSDmakefile e998fb2c63 Merged revisions 285090 via svnmerge from пре 14 година
BUGS 22a92da9b9 These changes should really be in trunk, not the 1.10 branch. пре 13 година
CHANGES 8f3d0093f3 app_queue: Fix multiple calls to a queue member that is in only one queue. пре 12 година
COPYING 70a40d92e3 пре 19 година
CREDITS 877d64d226 Revert previous commit пре 13 година
LICENSE b35c2ced5d Merged revisions 245044 via svnmerge from пре 15 година
Makefile c931c7bbec build_tools: Allow Asterisk to report git SHAs in version string. пре 12 година
Makefile.moddir_rules 2fd850ce8d Change the internal name of the menuselect options that are used to control пре 13 година
Makefile.rules b0ff3aefa8 Simplify build system architecture optimization пре 12 година
README 53709a22b7 Doxygen Updates пре 12 година
README-SERIOUSLY.bestpractices.txt 3ea69b7c78 security: Inhibit execution of privilege escalating functions пре 11 година
README-addons.txt 358628a848 Move Asterisk-addons modules into the main Asterisk source tree. пре 15 година
UPGRADE-1.2.txt 1224d7ea74 as suggested by jtodd, document the purposes of the CHANGES and UPGRADE files пре 16 година
UPGRADE-1.4.txt 041d524be0 Convert this branch to Opsound music-on-hold. пре 15 година
UPGRADE-1.6.txt b791cd73a4 Update UPGRADE-1.6.txt stating insecure=very has been removed. пре 14 година
UPGRADE-1.8.txt c95f2c2c9f chan_sip: Note change in behavior to how directmediapermit/deny ACL works пре 12 година
UPGRADE.txt 3ea69b7c78 security: Inhibit execution of privilege escalating functions пре 11 година
Zaptel-to-DAHDI.txt 904fefe038 Merged revisions 137679 via svnmerge from пре 16 година
bootstrap.sh efde6a6b90 Make sure asterisk builds on OpenBSD пре 13 година
config.guess a6412d6fea Update config.guess and config.sub: 2012-10-10 пре 12 година
config.sub a6412d6fea Update config.guess and config.sub: 2012-10-10 пре 12 година
configure c931c7bbec build_tools: Allow Asterisk to report git SHAs in version string. пре 12 година
configure.ac c931c7bbec build_tools: Allow Asterisk to report git SHAs in version string. пре 12 година
default.exports 95510fb2a7 Merged revisions 182808 via svnmerge from пре 15 година
install-sh a13e072bec silly people that don't want to install/run autoconf :-) пре 18 година
makeopts.in c931c7bbec build_tools: Allow Asterisk to report git SHAs in version string. пре 12 година
missing a13e072bec silly people that don't want to install/run autoconf :-) пре 18 година
mkinstalldirs a13e072bec silly people that don't want to install/run autoconf :-) пре 18 година
sample.call 4739d48ca4 Merged revisions 299138 via svnmerge from пре 14 година

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

In addition you'll find lots of information compiled by the Asterisk
community on this Wiki:

https://wiki.asterisk.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.
-------------------------------------------------------------------------------

-------------------------------------------------------------------------------
--- 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.