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Asterisk with DECT support
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markster 3ac4e25080 Various contributed VPB fixes
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@865 f38db490-d61c-443f-a65b-d21fe96a405b
2003-04-18 14:34:46 +00:00
agi mer feb 26 20:41:00 CET 2003 2003-02-26 19:41:10 +00:00
apps Make paging support apply to forwarded messages too 2003-04-15 00:19:27 +00:00
astman Sun Mar 16 07:00:01 CET 2003 2003-03-16 06:00:11 +00:00
cdr Eliminate localtime calls, various cleanups 2003-03-31 03:19:34 +00:00
channels Various contributed VPB fixes 2003-04-18 14:34:46 +00:00
codecs Fix sample frame 2003-04-17 15:42:01 +00:00
configs Add preliminary voicetronix support 2003-04-15 14:39:06 +00:00
contrib Minor voicemail patch 2003-03-21 20:59:29 +00:00
db1-ast Version 0.3.0 from FTP 2003-01-04 18:53:14 +00:00
doc Merge hint patch, add new variables, and misc. PBX cleanups 2003-03-30 22:55:42 +00:00
editline Contributed off-by-one fix for editline 2003-03-26 22:37:33 +00:00
formats Merge hint patch, add new variables, and misc. PBX cleanups 2003-03-30 22:55:42 +00:00
images Version 0.1.12 from FTP 2002-06-05 05:00:08 +00:00
include/asterisk Add "prod" function, to get audio moving 2003-04-16 02:47:03 +00:00
keys Version 0.1.10 from FTP 2001-12-25 21:12:07 +00:00
pbx Merge enhanced status changes, add SIP subscribe from Andre 2003-04-06 18:19:51 +00:00
redhat Version 0.3.0 from FTP 2002-10-21 00:45:13 +00:00
res Implement call pickup on SIP, override context if appropriate 2003-04-09 04:00:43 +00:00
sounds Add contributed sounds from John Todd and Allison Smith 2003-04-09 05:49:03 +00:00
utils Sun Mar 16 07:00:01 CET 2003 2003-03-16 06:00:11 +00:00
.cvsignore mer feb 12 14:56:57 CET 2003 2003-02-12 13:59:15 +00:00
BUGS Version 0.1.10 from FTP 2001-10-30 14:44:25 +00:00
CHANGES Add preliminary voicetronix support 2003-04-15 14:39:06 +00:00
CREDITS Various contributed VPB fixes 2003-04-18 14:34:46 +00:00
HARDWARE Version 0.3.0 from FTP 2002-12-02 03:01:17 +00:00
LICENSE Version 0.1.1 from FTP 1999-12-08 00:16:51 +00:00
Makefile Tue Mar 18 07:00:01 CET 2003 2003-03-18 06:00:18 +00:00
README Version 0.1.12 from FTP 2002-04-19 21:17:35 +00:00
README.cdr Version 0.1.10 from FTP 2001-10-18 15:27:19 +00:00
README.festival Version 0.3.0 from FTP 2003-01-08 19:45:43 +00:00
README.iax Version 0.1.10 from FTP 2002-01-07 02:01:04 +00:00
README.variables Merge hint patch, add new variables, and misc. PBX cleanups 2003-03-30 22:55:42 +00:00
SECURITY Version 0.1.10 from FTP 2001-11-10 18:09:19 +00:00
acl.c Fix ACL issue, update CREDITS 2003-04-09 13:14:50 +00:00
addmailbox Version 0.1.9 from FTP 2001-06-20 02:50:27 +00:00
alaw.c Version 0.1.10 from FTP 2001-12-09 00:41:43 +00:00
app.c Fix silly typo 2003-03-26 00:16:00 +00:00
ast_expr.y Version 0.3.0 from FTP 2002-10-25 13:24:50 +00:00
astconf.h Version 0.3.0 from FTP 2003-01-30 15:03:20 +00:00
asterisk-ng-doxygen Version 0.1.10 from FTP 2001-10-24 17:10:36 +00:00
asterisk.c Add new app 2003-04-16 13:43:11 +00:00
asterisk.h Version 0.3.0 from FTP 2003-01-30 15:03:20 +00:00
astgenkey Version 0.1.10 from FTP 2001-12-25 23:27:38 +00:00
autoservice.c Version 0.3.0 from FTP 2002-12-30 13:09:27 +00:00
callerid.c Eliminate localtime calls, various cleanups 2003-03-31 03:19:34 +00:00
cas.h Version 0.1.8 from FTP 2001-04-18 03:38:18 +00:00
cdr.c Merge hint patch, add new variables, and misc. PBX cleanups 2003-03-30 22:55:42 +00:00
channel.c Finish implementing prod code 2003-04-17 02:52:53 +00:00
chanvars.c Version 0.3.0 from FTP 2002-10-22 15:31:47 +00:00
cli.c Implement call pickup on SIP, override context if appropriate 2003-04-09 04:00:43 +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 Fix little leak thanks to James G. 2003-04-07 16:22:53 +00:00
db.c Add ast_db_freetree and ast_db_gettree 2003-04-13 04:17:45 +00:00
dsp.c Additional fix on not sending additional digits 2003-03-26 17:15:31 +00:00
ecdisa.h Version 0.1.10 from FTP 2001-11-10 20:31:39 +00:00
festival-1.4.1-diff Version 0.2.0 from FTP 2002-09-11 17:09:48 +00:00
festival-1.4.2.diff Version 0.3.0 from FTP 2003-01-08 16:57:11 +00:00
file.c Make voicemail timeout configurable 2003-04-09 02:10:29 +00:00
frame.c Add iLBC codec 2003-04-15 04:36:52 +00:00
fskmodem.c Version 0.1.10 from FTP 2001-11-10 20:31:39 +00:00
image.c Version 0.3.0 from FTP 2003-01-30 15:03:20 +00:00
indications.c dom mar 16 23:37:23 CET 2003 2003-03-16 22:37:31 +00:00
init.asterisk Version 0.3.0 from FTP 2002-10-22 15:31:47 +00:00
io.c Version 0.3.0 from FTP 2002-11-27 05:04:07 +00:00
loader.c Version 0.3.0 from FTP 2003-01-30 15:03:20 +00:00
logger.c Eliminate localtime calls, various cleanups 2003-03-31 03:19:34 +00:00
make_build_h Version 0.1.8 from FTP 2001-05-09 03:11:22 +00:00
manager.c Add new field for USER info to manager interface perms 2003-04-14 18:47:36 +00:00
md5.c Version 0.1.12 from FTP 2002-06-29 22:09:03 +00:00
pbx.c Add new app 2003-04-16 13:43:11 +00:00
privacy.c Version 0.3.0 from FTP 2003-01-17 03:46:33 +00:00
rtp.c Fix transmissions 2003-04-17 19:19:12 +00:00
safe_asterisk mar feb 18 19:15:15 CET 2003 2003-02-18 18:15:30 +00:00
sample.call Version 0.3.0 from FTP 2002-10-29 05:38:33 +00:00
sas.h Version 0.1.8 from FTP 2001-04-17 20:06:58 +00:00
say.c Eliminate localtime calls, various cleanups 2003-03-31 03:19:34 +00:00
sched.c Version 0.2.0 from FTP 2002-09-05 21:32:54 +00:00
sounds.txt Add contributed sounds from John Todd and Allison Smith 2003-04-09 05:49:03 +00:00
tdd.c Version 0.1.10 from FTP 2001-12-07 22:57:34 +00:00
term.c dom mar 16 23:37:23 CET 2003 2003-03-16 22:37:31 +00:00
translate.c dom mar 16 23:37:23 CET 2003 2003-03-16 22:37:31 +00:00
ulaw.c Version 0.1.10 from FTP 2001-12-07 22:57:34 +00:00
valgrind-RedHat-8.0.supp Wed Mar 19 07:00:01 CET 2003 2003-03-19 06:00:11 +00:00
vmail.cgi Minor voicemail patch 2003-03-21 20:59:29 +00:00
zonedata.c Version 0.3.0 from FTP 2002-10-20 19:57:17 +00:00

README

The Asterisk Open Source PBX
by Mark Spencer <markster@linux-support.net>
Copyright (C) 2001, Linux Support Services, Inc.
================================================================
* 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.asteriskpbx.com

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

  Linux Support Services, Inc. retains copyright to all of the core
Asterisk system, and therefore can grant, at its sole discression, 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.  At
this time (5/21/2001) the only component of Asterisk which is covered
under GPL and not under our Copyright is the Xing MP3 decoder.

  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 excemption 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.
  
* 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
as well.


* GETTING STARTED

First, be sure you've got supported hardware.  To use Asterisk right now,
you will need one of the following:

	* All Wildcard (tm) products from LSS (www.linux-support.net)
	* 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)

Assuming you have one of these (most likely the third) you're ready to 
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.

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

Finally, you may wish to visit the web site and join the mailing list if
you're interested in getting more information.

Mark