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Asterisk with DECT support
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markster a9cde19375 First pass at auto logoff support
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@1199 f38db490-d61c-443f-a65b-d21fe96a405b
2003-07-22 11:06:56 +00:00
agi Add and update .cvsignore files for .depend 2003-05-08 03:46:43 +00:00
apps Oops 2003-07-16 16:47:44 +00:00
astman Add and update .cvsignore files for .depend 2003-05-08 03:46:43 +00:00
cdr optionally log uniqueid as well 2003-07-18 05:38:11 +00:00
channels First pass at auto logoff support 2003-07-22 11:06:56 +00:00
codecs One more 2003-05-08 03:47:22 +00:00
configs First pass at auto logoff support 2003-07-22 11:06:56 +00:00
contrib Fix a typo 2003-07-14 19:10:11 +00:00
db1-ast More BSD enhancements 2003-04-27 18:13:11 +00:00
doc optionally log uniqueid as well 2003-07-18 05:38:11 +00:00
editline More BSD enhancements 2003-04-27 18:13:11 +00:00
formats Properly implement using zaptel for timing of file playback 2003-06-29 20:32:26 +00:00
images Version 0.1.12 from FTP 2002-06-05 05:00:08 +00:00
include/asterisk Add the possibility to delete all the contexts registered by a certain registrar with ast_merge_and_delete routine; make it the default behaviour when reloding extensions 2003-07-14 17:16:47 +00:00
keys Version 0.1.10 from FTP 2001-12-25 21:12:07 +00:00
pbx Add the possibility to delete all the contexts registered by a certain registrar with ast_merge_and_delete routine; make it the default behaviour when reloding extensions 2003-07-14 17:16:47 +00:00
redhat Version 0.3.0 from FTP 2002-10-21 00:45:13 +00:00
res Add a safe way to reload extensions config (don't change/delete the current extenions until extensions.conf was parsed and the new set of extensions is created) and add "extensions reload" CLI command so we could reload only extensions.conf config file without touching config files of other modules 2003-07-14 15:33:21 +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 Add and update .cvsignore files for .depend 2003-05-08 03:46:43 +00:00
BUGS Version 0.1.10 from FTP 2001-10-30 14:44:25 +00:00
CHANGES Update changelog and update agents configa 2003-07-02 20:24:06 +00:00
CREDITS Merge Vonage's MySQL Voicemail stuff 2003-05-16 20:37:02 +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 Properly implement using zaptel for timing of file playback 2003-06-29 20:32:26 +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 Add unique identifier 2003-05-30 04:41:18 +00:00
SECURITY Version 0.1.10 from FTP 2001-11-10 18:09:19 +00:00
acl.c Take out useless cast 2003-05-08 14:53:15 +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 Use digit/response timeouts 2003-06-11 12:14:38 +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 Don't delete PID file on exit of asterisk -r 2003-07-09 13:45:59 +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
astmm.c Fix typo 2003-05-09 00:26:34 +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 Add unique identifier 2003-05-30 04:41:18 +00:00
channel.c Fix race in agent/masquerade 2003-07-16 18:54:16 +00:00
chanvars.c Include fixes for portability 2003-05-16 23:33:41 +00:00
cli.c Minor cleanups 2003-06-27 23:02:52 +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 Add agent groupings, fix the "incorrect" message on first login attempt 2003-07-01 04:08:25 +00:00
db.c Add ast_db_freetree and ast_db_gettree 2003-04-13 04:17:45 +00:00
dsp.c Add the second way of signalizing hangup when busydetect detects the busy tone 2003-07-08 21:14:59 +00:00
ecdisa.h Version 0.1.10 from FTP 2001-11-10 20:31:39 +00:00
enum.c Add SRV code to SIP, cleanup ENUM and make IAX2 do the right thing on dials 2003-06-12 12:48:57 +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 Still store timing 2003-07-01 23:17:10 +00:00
frame.c Add SIP/RTP video support, video enable app_echo, start on RTCP 2003-06-28 16:40:02 +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 Fix the playtones app so that we can pass the tones as an argument ( we don't need to refer to a defined tone in indications.conf ) 2003-06-17 18:59:58 +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 Make RTP ports configurable 2003-05-16 02:50:46 +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 Extend manager originate functionality 2003-06-26 20:32:52 +00:00
md5.c Version 0.1.12 from FTP 2002-06-29 22:09:03 +00:00
mkdep Beginning of solaris portability 2003-05-06 22:27:46 +00:00
pbx.c pbx.c didn't get updated 2003-07-14 17:17:05 +00:00
privacy.c Version 0.3.0 from FTP 2003-01-17 03:46:33 +00:00
retrieve_extensions_from_mysql.pl Fix a typo 2003-07-14 19:10:11 +00:00
rtp.c Minor rtp fixup 2003-07-11 21:51:06 +00:00
safe_asterisk Add debugging to safe_asterisk 2003-05-07 22:33:55 +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 Add commonly used include headers 2003-04-23 19:09:13 +00:00
sched.c Version 0.2.0 from FTP 2002-09-05 21:32:54 +00:00
sounds.txt Add dynamic agent stuff, still missing audio files 2003-07-14 02:31:56 +00:00
srv.c Add missing srv.c and srv.h files 2003-06-12 22:14:03 +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 Fix vmail "taint" issue 2003-05-22 04:50:53 +00:00
vmdb.sql Merge Vonage changes to VM2, ready to be edited and updated :) 2003-05-16 23:52:01 +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