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Asterisk with DECT support
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markster 92f0707533 Don't take digits in-band when in overlap dial mode
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@2668 f38db490-d61c-443f-a65b-d21fe96a405b
2004-04-09 16:33:37 +00:00
agi Add support for Costa Rica to DSP 2004-03-28 02:53:04 +00:00
apps Add ability to do PINs with dynamic conf (bug 1026) 2004-04-08 19:38:26 +00:00
astman Make astman ignore 'StatusComplete' events. 2004-02-15 07:17:37 +00:00
cdr Create individual sip reload command (bug #880) 2004-01-23 15:43:31 +00:00
channels Don't take digits in-band when in overlap dial mode 2004-04-09 16:33:37 +00:00
codecs Minor samples updates 2004-04-04 16:33:56 +00:00
configs Add Matt's option to call an external application when a voicemail is 2004-04-06 21:52:42 +00:00
contrib Make primary key context and voicemailbox and make the file easier on 2004-03-20 07:51:56 +00:00
db1-ast Various warning cleanups 2003-11-23 22:14:32 +00:00
doc Add ${LANGUAGE} channel variable (bug #1078) 2004-02-22 05:25:58 +00:00
editline Move config.cache delete to "distclean" 2004-02-02 07:01:25 +00:00
formats Make ADPCM frames standard 2004-03-11 19:57:10 +00:00
images Version 0.1.12 from FTP 2002-06-05 05:00:08 +00:00
include/asterisk Get rid of all that old needlock garbage now that we're using recursive mutexes 2004-04-06 22:17:32 +00:00
keys Version 0.1.10 from FTP 2001-12-25 21:12:07 +00:00
pbx Make outgoing call spool only consider # a comment if at the beginning of a line or following whitespace (bug 1346) 2004-04-05 18:03:40 +00:00
redhat Add the SuSE AMD64 support and fixes from Bug #706 2004-01-08 16:52:11 +00:00
res Fix double parking crash (bug #1302) 2004-04-07 04:11:00 +00:00
sounds Add sound files for dynamic conf stuff (bug 1026) 2004-04-08 19:38:47 +00:00
stdtime Get .depend for stdtime 2003-11-05 06:19:41 +00:00
utils Make astman ignore 'StatusComplete' events. 2004-02-15 07:17:37 +00:00
.cvsignore Add and update .cvsignore files for .depend 2003-05-08 03:46:43 +00:00
BUGS Fix BUGS document to report bug tracker 2004-01-12 03:03:23 +00:00
CHANGES Merge queue changes from Bug #214 2004-03-13 06:00:41 +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 Install to iaxy.bin, not iaxy.conf 2004-03-23 17:00:13 +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 Fix ACL for bizarre routing cases 2004-03-20 19:08:57 +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 BSD portability fix 2004-03-02 22:53:08 +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 Get rid of all that old needlock garbage now that we're using recursive mutexes 2004-04-06 22:17:32 +00:00
ast_expr.y Code cleanups (bug #66) 2003-09-19 01:20:23 +00:00
astconf.h Version 0.3.0 from FTP 2003-01-30 15:03:20 +00:00
asterisk.c Fix command completion and remove compile time warnings 2004-04-07 18:26:24 +00:00
asterisk.h Have a contact line in responses, merge logging patches 2003-11-26 22:00:07 +00:00
astmm.c add a vasprintf replacement. Bug #839 2004-01-14 06:35:01 +00:00
autoservice.c Fix bug 1217. Change pthread_t initializers to AST_PTHREADT_NULL and 2004-03-15 07:51:22 +00:00
callerid.c Recognize + as a valid part of a phone number 2004-04-04 22:14:10 +00:00
cdr.c Add application to log user data to the CDRs 2004-01-11 03:23:05 +00:00
channel.c Remove double unlock from ast_read 2004-04-07 14:43:18 +00:00
chanvars.c Include fixes for portability 2003-05-16 23:33:41 +00:00
cli.c Free some cli memory 2004-04-08 05:28:53 +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 function ast_false, like ast_true 2004-03-02 07:57:06 +00:00
db.c Make valgrind happy on db read 2003-12-02 15:12:56 +00:00
dlfcn.c Make it build and run on MacOS X 2003-10-26 19:17:28 +00:00
dns.c Copy old extension unless it's ASYNCGOTO (bug #1141) 2004-03-04 05:57:47 +00:00
dsp.c Get rid of all that old needlock garbage now that we're using recursive mutexes 2004-04-06 22:17:32 +00:00
ecdisa.h Version 0.1.10 from FTP 2001-11-10 20:31:39 +00:00
enum.c Make DNS callbacks return -1 on error, so invalid records are ignored 2004-03-03 03:12:59 +00:00
file.c Get rid of all that old needlock garbage now that we're using recursive mutexes 2004-04-06 22:17:32 +00:00
frame.c Handle 4 byte packets too 2004-03-14 21:01:13 +00:00
fskmodem.c Version 0.1.10 from FTP 2001-11-10 20:31:39 +00:00
image.c Show the names of the codecs instead of the numbers (bug #92) 2003-08-16 05:10:35 +00:00
indications.c Get rid of all that old needlock garbage now that we're using recursive mutexes 2004-04-06 22:17:32 +00:00
io.c Make it build and run on MacOS X 2003-10-26 18:50:49 +00:00
loader.c Just in case resources with the same name are loaded 2004-02-19 01:04:21 +00:00
logger.c Indicate the need to reload without actually reloading... 2004-03-30 16:16:05 +00:00
make_build_h Version 0.1.8 from FTP 2001-05-09 03:11:22 +00:00
manager.c Allow manager.conf to use port and portno, but log that portno is deprecated. 2004-04-05 19:40:30 +00:00
md5.c Change md5 to use uint32_t instead of uint32 2004-03-30 23:55:41 +00:00
mkdep FreeBSD compatability fixes 2003-08-19 06:06:50 +00:00
pbx.c Get rid of all that old needlock garbage now that we're using recursive mutexes 2004-04-06 22:17:32 +00:00
poll.c Make it build and run on MacOS X 2003-10-26 19:17:28 +00:00
privacy.c Version 0.3.0 from FTP 2003-01-17 03:46:33 +00:00
rtp.c Use caller's negotiated values when possible. 2004-03-22 21:52:22 +00:00
sample.call Add example of using Account in sample.call file 2004-03-26 08:04:13 +00:00
say.c Add support for Norwegian numbers (bug #1067) 2004-02-20 17:43:46 +00:00
sched.c Unlock while processing schedule queue 2003-11-21 22:05:08 +00:00
sounds.txt Fix sounds.txt to have extensions for recently included sound files 2004-04-09 00:15:34 +00:00
srv.c Make DNS callbacks return -1 on error, so invalid records are ignored 2004-03-03 03:12:59 +00:00
tdd.c Version 0.1.10 from FTP 2001-12-07 22:57:34 +00:00
term.c Add "crt" to list that knows colorization (#410) 2003-10-22 03:04:45 +00:00
translate.c Don't translate time for packets with no delivery 2004-04-05 20:47:44 +00:00
ulaw.c Version 0.1.10 from FTP 2001-12-07 22:57:34 +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