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/

Kevin P. Fleming e96888e393 15 éve
agi 54727e1851 19 éve
apps a10b9aee0b properly reset the 'next' pointer when duplicating the cdr (issue #5921) 19 éve
astman 54727e1851 19 éve
cdr 54727e1851 19 éve
channels 8babceca67 properly handle signed integer input 18 éve
codecs 54727e1851 19 éve
configs 54727e1851 19 éve
contrib 54727e1851 19 éve
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editline 54727e1851 19 éve
formats 54727e1851 19 éve
images 54727e1851 19 éve
include 54727e1851 19 éve
keys 54727e1851 19 éve
pbx 0e78c0b1b0 fix memory leak with DelayedRetry (issue #6157) 19 éve
redhat 54727e1851 19 éve
res a59115133b check array bounds when parsing arguments to AGI (issue #5868) 19 éve
sounds e96888e393 15 éve
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CREDITS e96888e393 15 éve
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LICENSE 54727e1851 19 éve
Makefile 54727e1851 19 éve
README 54727e1851 19 éve
README.opsound e96888e393 15 éve
SECURITY 54727e1851 19 éve
acl.c 54727e1851 19 éve
aescrypt.c 54727e1851 19 éve
aeskey.c 54727e1851 19 éve
aesopt.h 54727e1851 19 éve
aestab.c 54727e1851 19 éve
alaw.c 54727e1851 19 éve
app.c 54727e1851 19 éve
ast_expr.y 54727e1851 19 éve
astconf.h 54727e1851 19 éve
asterisk.8.gz 54727e1851 19 éve
asterisk.c 54727e1851 19 éve
asterisk.h 54727e1851 19 éve
asterisk.sgml 54727e1851 19 éve
astmm.c 54727e1851 19 éve
autoservice.c 54727e1851 19 éve
callerid.c 54727e1851 19 éve
cdr.c 54727e1851 19 éve
channel.c 54727e1851 19 éve
chanvars.c 54727e1851 19 éve
cli.c 54727e1851 19 éve
coef_in.h 54727e1851 19 éve
coef_out.h 54727e1851 19 éve
config.c 54727e1851 19 éve
db.c 54727e1851 19 éve
dlfcn.c 54727e1851 19 éve
dns.c 54727e1851 19 éve
dsp.c 54727e1851 19 éve
ecdisa.h 54727e1851 19 éve
enum.c 54727e1851 19 éve
file.c 54727e1851 19 éve
frame.c 54727e1851 19 éve
fskmodem.c 54727e1851 19 éve
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indications.c 54727e1851 19 éve
io.c 54727e1851 19 éve
loader.c 54727e1851 19 éve
logger.c 54727e1851 19 éve
make_build_h 2fc1e33737 Version 0.1.8 from FTP 23 éve
manager.c 54727e1851 19 éve
md5.c 54727e1851 19 éve
mkdep dad6895322 Make mkdep throw away stderr since people think the error messages printed are serious when they are not 20 éve
muted.c 54727e1851 19 éve
muted.conf.sample 54727e1851 19 éve
pbx.c 54727e1851 19 éve
poll.c 54727e1851 19 éve
privacy.c 54727e1851 19 éve
rtp.c 54727e1851 19 éve
sample.call 54727e1851 19 éve
say.c 54727e1851 19 éve
sched.c 54727e1851 19 éve
sounds.txt 54727e1851 19 éve
srv.c 54727e1851 19 éve
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README

The Asterisk Open Source PBX
by Mark Spencer
Copyright (C) 2001-2004 Digium
================================================================
* SECURITY
It is imperative that you read and fully understand the contents of
the SECURITY file before you attempt to configure 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 lot's of information compiled by the Asterisk
community on this Wiki:

http://www.voip-info.org/wiki-Asterisk

* LICENSING
Asterisk is distributed under GNU General Public License. The GPL also
must apply to all loadable modules as well, except as defined below.

Digium, Inc. (formerly Linux Support Services) retains copyright to all
of the core Asterisk system, and therefore can grant, at its sole discretion,
the ability for companies, individuals, or organizations to create proprietary
or Open Source (but non-GPL'd) modules which may be dynamically linked at
runtime with the portions of Asterisk which fall under our copyright
umbrella, or are distributed under more flexible licenses than GPL.


If you wish to use our code in other GPL programs, don't worry -- there
is no requirement that you provide the same exemption in your GPL'd
products (although if you've written a module for Asterisk we would
strongly encourage you to make the same exemption that we do).

Specific permission is also granted to OpenSSL and OpenH323 to link to
Asterisk.

If you have any questions, whatsoever, regarding our licensing policy,
please contact us.

Modules that are GPL-licensed and not available under Digium's
licensing scheme are added to the Asterisk-addons CVS module.

* REQUIRED COMPONENTS

== Linux ==
Currently, the Asterisk Open Source PBX is only known to run on the
Linux OS, although it may be portable to other UNIX-like operating systems
(like FreeBSD) as well.


* GETTING STARTED

First, be sure you've got supported hardware (but note that you don't need ANY hardware, not even a soundcard) to install and run Asterisk. Supported are:

* All Wildcard (tm) products from Digium (www.digium.com)
* QuickNet Internet PhoneJack and LineJack (http://www.quicknet.net)
* Full Duplex Sound Card supported by Linux
* Adtran Atlas 800 Plus
* ISDN4Linux compatible ISDN card
* Tormenta Dual T1 card (www.bsdtelephony.com.mx)

Hint: CAPI compatible ISDN cards can be run using the add-on channel chan_capi.

So let's proceed:

1) Run "make"
2) 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:

"make samples"

Doing so will overwrite any existing config files you have. If you are lacking a soundcard you won't be able to use the DIAL command on the console, though.

Finally, you can launch Asterisk 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 ". 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) than it
won't work right (not yet).

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 tormenta.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

Then, 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 tormenta card, obtaining the settings
from the variables specified above.

* SPECIAL NOTE ON TIME

Those using SIP phones should be aware the 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.

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/index.php?menu=support

Welcome to the growing worldwide community of Asterisk users!

Mark Spencer