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Mark Spencer 0f111e2d3a
Ignore ECHILD in app_system
21 years ago
agi Perl cleanups 22 years ago
apps Ignore ECHILD in app_system 21 years ago
astman Make astman ignore 'StatusComplete' events. 22 years ago
cdr Create individual sip reload command (bug #880) 22 years ago
channels Add missing include to chan_iax2.c 21 years ago
codecs Add G.726-32kbps Codec Transcoder (Tested with Cisco ATA-186) 21 years ago
configs Bug # 1079. indications.conf.sample changes for Greece 21 years ago
contrib Improve SIP friends support (should address bugs #1063 & #1052) 21 years ago
db1-ast Various warning cleanups 22 years ago
doc Add ${LANGUAGE} channel variable (bug #1078) 21 years ago
editline Move config.cache delete to "distclean" 22 years ago
formats Add G.726, revise Changelog 21 years ago
images Version 0.1.12 from FTP 23 years ago
include/asterisk Move ast_get_group from res_parking.c to channel.c 21 years ago
keys Version 0.1.10 from FTP 24 years ago
pbx Fix some comments in pbx.c and pbx_config.c 21 years ago
redhat Add the SuSE AMD64 support and fixes from Bug #706 22 years ago
res Move ast_get_group from res_parking.c to channel.c 21 years ago
sounds Add custom folder sounds 22 years ago
stdtime Get .depend for stdtime 22 years ago
utils Make astman ignore 'StatusComplete' events. 22 years ago
.cvsignore Add and update .cvsignore files for .depend 22 years ago
BUGS Fix BUGS document to report bug tracker 22 years ago
CHANGES Add G.726, revise Changelog 21 years ago
CREDITS add ww 22 years ago
HARDWARE Update HARDWARE 22 years ago
LICENSE Version 0.1.1 from FTP 26 years ago
Makefile Add IAX2 firmware upgrade support 21 years ago
README Fix 2 typos in README 22 years ago
SECURITY Version 0.1.10 from FTP 24 years ago
acl.c More BSD enhancements (#970) 22 years ago
aescrypt.c Add AES support 22 years ago
aeskey.c Add AES support 22 years ago
aesopt.h Add AES support 22 years ago
aestab.c Add AES support 22 years ago
alaw.c Version 0.1.10 from FTP 24 years ago
app.c Use digit/response timeouts 22 years ago
ast_expr.y Code cleanups (bug #66) 22 years ago
astconf.h Version 0.3.0 from FTP 23 years ago
asterisk.c Fix restarting when not called from the main console (bug #830 and #864) 21 years ago
asterisk.h Have a contact line in responses, merge logging patches 22 years ago
astmm.c add a vasprintf replacement. Bug #839 22 years ago
autoservice.c BSD portability enhancements (bug #234) 22 years ago
callerid.c Typo 22 years ago
cdr.c Add application to log user data to the CDRs 22 years ago
channel.c Move ast_get_group from res_parking.c to channel.c 21 years ago
chanvars.c Include fixes for portability 22 years ago
cli.c Fix 'show channel<tab>' completion, where show channel would be 22 years ago
coef_in.h Version 0.1.7 from FTP 24 years ago
coef_out.h Version 0.1.7 from FTP 24 years ago
config.c Add agent groupings, fix the "incorrect" message on first login attempt 22 years ago
db.c Make valgrind happy on db read 22 years ago
dlfcn.c Make it build and run on MacOS X 22 years ago
dns.c Make it build and run on MacOS X 22 years ago
dsp.c Fix excessive fax detection (Thanks Steve Underwood!!!) 22 years ago
ecdisa.h Version 0.1.10 from FTP 24 years ago
enum.c Minor enum improvements for iax/iax2 22 years ago
file.c Bug #1087. Fix wav49 format so it can be played. Make file functions 21 years ago
frame.c Replace MP3 with G726 in ast_codec2str 21 years ago
fskmodem.c Version 0.1.10 from FTP 24 years ago
image.c Show the names of the codecs instead of the numbers (bug #92) 22 years ago
indications.c Add queue logging and fix indications buglet 22 years ago
io.c Make it build and run on MacOS X 22 years ago
loader.c Just in case resources with the same name are loaded 21 years ago
logger.c Get rid of compiler warnings when calling ast_queue_log 21 years ago
make_build_h Version 0.1.8 from FTP 24 years ago
manager.c Insert blank after REFER (bug #997) 22 years ago
md5.c Version 0.1.12 from FTP 23 years ago
mkdep FreeBSD compatability fixes 22 years ago
pbx.c Fix Bug # 981 21 years ago
poll.c Make it build and run on MacOS X 22 years ago
privacy.c Version 0.3.0 from FTP 23 years ago
rtp.c Fix slow down in rtp.c 21 years ago
sample.call Version 0.3.0 from FTP 23 years ago
say.c Add support for Norwegian numbers (bug #1067) 21 years ago
sched.c Unlock while processing schedule queue 22 years ago
sounds.txt Update sounds.txt 22 years ago
srv.c More cleanups and OSX fixes for 10.3 22 years ago
tdd.c Version 0.1.10 from FTP 24 years ago
term.c Add "crt" to list that knows colorization (#410) 22 years ago
translate.c Fix 'show translations' 21 years ago
ulaw.c Version 0.1.10 from FTP 24 years ago

README

The Asterisk Open Source PBX
by Mark Spencer <markster@digium.com>
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 <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) 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.

* 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