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Asterisk with DECT support
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markster 5e82ef61af Work properly on a reload
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@3516 f38db490-d61c-443f-a65b-d21fe96a405b
2004-07-26 19:37:11 +00:00
agi Merge major BSD mutex and symbol conflict patches (bug #1816) (link patch still pending) 2004-06-22 17:42:14 +00:00
apps Fix ADSI prog to only accept 253 (bug #2135) 2004-07-26 03:23:38 +00:00
astman Fix astman build on FreeBSD (bug #2119) 2004-07-24 03:52:53 +00:00
cdr Add Manager CDR (off by default) (bug #2127) courtesy cybershield 2004-07-25 14:56:34 +00:00
channels Work properly on a reload 2004-07-26 19:37:11 +00:00
codecs Fix YDL build (bug #2109) 2004-07-22 23:43:26 +00:00
configs Turn off DTMF generally in MGCP and make option to enable RFC2833 or in-band 2004-07-26 19:33:36 +00:00
contrib Let safe_asterisk take arguments (bug #2085) 2004-07-25 23:12:33 +00:00
db1-ast Merge remaining audit patch (save dlfcn.c) 2004-07-14 13:57:15 +00:00
doc Update dial documentation (bug #2096) 2004-07-19 19:47:21 +00:00
editline Merge remaining audit patch (save dlfcn.c) 2004-07-14 13:57:15 +00:00
formats Remove pthread.h from source. We should be using asterisk/lock.h everywhere instead (except in asterisk/lock.h). 2004-06-22 18:49:00 +00:00
images Version 0.1.12 from FTP 2002-06-05 05:00:08 +00:00
include/asterisk Oops, make that ast_features 2004-07-17 23:56:12 +00:00
keys Add information for IAX on Free World Dialup 2004-06-02 23:19:36 +00:00
pbx Merge remaining audit patch (save dlfcn.c) 2004-07-14 13:57:15 +00:00
redhat Add Asterisk manpage 2004-07-18 17:58:05 +00:00
res added support to be able to set the channel var TRANSFER_CONTEXT so when 2004-07-24 02:29:40 +00:00
sounds Voicemail fixes (bug #1982) 2004-07-17 01:34:20 +00:00
stdtime Merge remaining audit patch (save dlfcn.c) 2004-07-14 13:57:15 +00:00
utils Fix astman build on FreeBSD (bug #2119) 2004-07-24 03:52:53 +00:00
.cvsignore Add support for E1 E&M 2004-04-16 18:00:00 +00:00
BUGS Update Changelog/BUGS 2004-07-17 02:52:52 +00:00
CHANGES Update Changelog/BUGS 2004-07-17 02:52:52 +00:00
CREDITS add ww 2004-01-13 06:11:18 +00:00
HARDWARE Update HARDWARE 2004-01-12 03:16:56 +00:00
LICENSE Version 0.1.1 from FTP 1999-12-08 00:16:51 +00:00
Makefile Merge mic's minor patchlet (bug #2092) 2004-07-19 15:52:57 +00:00
README Fix 2 typos in README 2004-02-15 06:58:43 +00:00
SECURITY Update security document, work on threading with pbx.c and small SIP fixes 2004-04-02 07:24:33 +00:00
acl.c Use INET_ADDRLEN (bug #1956) (from airport!) 2004-06-30 16:56:51 +00:00
aescrypt.c Add AES support 2003-12-25 14:01:55 +00:00
aeskey.c Add AES support 2003-12-25 14:01:55 +00:00
aesopt.h Fix AES for MacOS build 2004-05-24 20:09:05 +00:00
aestab.c Add AES support 2003-12-25 14:01:55 +00:00
alaw.c Version 0.1.10 from FTP 2001-12-09 00:41:43 +00:00
app.c More strcpy / snprintf as part of rgagnon's audit (bug #2004) 2004-07-09 10:08:09 +00:00
ast_expr.y Merge source cleanups (bug #1911) 2004-06-26 03:50:14 +00:00
astconf.h Version 0.3.0 from FTP 2003-01-30 15:03:20 +00:00
asterisk.8.gz Add Asterisk manpage 2004-07-18 17:58:05 +00:00
asterisk.c Make that setgid... 2004-07-18 18:52:36 +00:00
asterisk.h Close logging stuff so system doesn't have to (bug #1855) 2004-06-17 01:13:10 +00:00
asterisk.sgml Add Asterisk manpage 2004-07-18 17:58:05 +00:00
astmm.c Remove pthread.h from source. We should be using asterisk/lock.h everywhere instead (except in asterisk/lock.h). 2004-06-22 18:49:00 +00:00
autoservice.c Remove pthread.h from source. We should be using asterisk/lock.h everywhere instead (except in asterisk/lock.h). 2004-06-22 18:49:00 +00:00
callerid.c Demand more information on callerid to prevent possible overruns 2004-07-23 00:27:26 +00:00
cdr.c More strcpy / snprintf as part of rgagnon's audit (bug #2004) 2004-07-09 10:08:09 +00:00
channel.c Swap states early in masquerade process (bug #1987) 2004-07-09 10:57:43 +00:00
chanvars.c Include fixes for portability 2003-05-16 23:33:41 +00:00
cli.c Minor formatting fix from code audit in cli.c 2004-07-14 14:53:24 +00:00
coef_in.h Version 0.1.7 from FTP 2001-03-20 20:11:20 +00:00
coef_out.h Version 0.1.7 from FTP 2001-03-20 20:11:20 +00:00
config.c More strcpy / snprintf as part of rgagnon's audit (bug #2004) 2004-07-09 10:08:09 +00:00
db.c More strcpy / snprintf as part of rgagnon's audit (bug #2004) 2004-07-09 10:08:09 +00:00
dlfcn.c Make it build and run on MacOS X 2003-10-26 19:17:28 +00:00
dns.c Misc formatting cleanups 2004-06-22 20:11:15 +00:00
dsp.c Lower default volume 2004-07-24 21:06:54 +00:00
ecdisa.h Version 0.1.10 from FTP 2001-11-10 20:31:39 +00:00
enum.c Remaining rgagnon source audit improvements (bug #2011) 2004-07-14 07:44:19 +00:00
file.c If breakon is unspecified, make it "" 2004-07-15 14:37:09 +00:00
frame.c Merge mic's minor patchlet (bug #2092) 2004-07-19 15:52:57 +00:00
fskmodem.c Version 0.1.10 from FTP 2001-11-10 20:31:39 +00:00
image.c Remove pthread.h from source. We should be using asterisk/lock.h everywhere instead (except in asterisk/lock.h). 2004-06-22 18:49:00 +00:00
indications.c Need actual offset space (bug #2076) 2004-07-18 15:58:57 +00:00
io.c Make it build and run on MacOS X 2003-10-26 18:50:49 +00:00
loader.c Remove pthread.h from source. We should be using asterisk/lock.h everywhere instead (except in asterisk/lock.h). 2004-06-22 18:49:00 +00:00
logger.c Fix logger issue at hangup 2004-07-20 15:10:18 +00:00
make_build_h Version 0.1.8 from FTP 2001-05-09 03:11:22 +00:00
manager.c Remaining rgagnon source audit improvements (bug #2011) 2004-07-14 07:44:19 +00:00
md5.c Fix AES for MacOS build 2004-05-24 20:09:05 +00:00
mkdep Make mkdep throw away stderr since people think the error messages printed are serious when they are not 2004-05-23 17:17:27 +00:00
muted.c Remaining rgagnon source audit improvements (bug #2011) 2004-07-14 07:44:19 +00:00
muted.conf.sample clean up config file sample 2004-05-17 06:39:17 +00:00
pbx.c Merge tilghman's app/appdata in newexten patch (bug #2104) 2004-07-24 02:26:07 +00:00
poll.c Make it build and run on MacOS X 2003-10-26 19:17:28 +00:00
privacy.c Remove pthread.h from source. We should be using asterisk/lock.h everywhere instead (except in asterisk/lock.h). 2004-06-22 18:49:00 +00:00
rtp.c Don't copmile in the no-RTP checksum stuff for non-Linux (bug #2089) 2004-07-19 18:09:33 +00:00
sample.call Add example of using Account in sample.call file 2004-03-26 08:04:13 +00:00
say.c datetime patches from Tilghman (bug #1905) 2004-06-23 20:19:12 +00:00
sched.c Minor scheduling fixups 2004-07-20 13:43:33 +00:00
sounds.txt Voicemail fixes (bug #1982) 2004-07-17 01:34:20 +00:00
srv.c REduce chattyness 2004-07-17 22:06:26 +00:00
tdd.c Version 0.1.10 from FTP 2001-12-07 22:57:34 +00:00
term.c Remove pthread.h from source. We should be using asterisk/lock.h everywhere instead (except in asterisk/lock.h). 2004-06-22 18:49:00 +00:00
translate.c Remaining rgagnon source audit improvements (bug #2011) 2004-07-14 07:44:19 +00:00
ulaw.c Version 0.1.10 from FTP 2001-12-07 22:57:34 +00:00
utils.c FreeBSD fix for utils (bug #1949) 2004-06-29 17:54:25 +00:00

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