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Asterisk with DECT support
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markster 47c004373a Add fixup to chan_phone (bug #3825)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@5304 f38db490-d61c-443f-a65b-d21fe96a405b
2005-03-30 06:39:39 +00:00
agi Merge slimey's Solaris compatibility (with small mods) (bug #2740) 2004-12-14 23:36:30 +00:00
apps fix group and arg parsing bug 2005-03-29 17:46:37 +00:00
cdr Fix cross compiling (bug #3868) 2005-03-27 22:39:17 +00:00
channels Add fixup to chan_phone (bug #3825) 2005-03-30 06:39:39 +00:00
codecs Fix cross compiling (bug #3868) 2005-03-27 22:39:17 +00:00
configs Take out unimplemented comment from logger.conf (bug #3890) 2005-03-30 06:33:44 +00:00
contrib Add load tester (bug #3870) 2005-03-27 22:44:09 +00:00
db1-ast Add support for Solaris/x86 (bug #3064) 2005-03-17 23:12:15 +00:00
doc Fix a bunch of const stuff, merge queue changes, add experimental "hybrid" DTMF mode 2005-03-28 20:48:24 +00:00
editline Merge slimey's Solaris compatibility (with small mods) (bug #2740) 2004-12-14 23:36:30 +00:00
formats Simplify endianness and fix for unaligned reads (bug #3867) 2005-03-29 04:49:24 +00:00
images Version 0.1.12 from FTP 2002-06-05 05:00:08 +00:00
include Allow functions to be written to (bug #2278, with mods) 2005-03-29 06:16:49 +00:00
keys Add information for IAX on Free World Dialup 2004-06-02 23:19:36 +00:00
patches Apply queuelog patch and perform final test of "test patches" system 2005-03-11 08:49:01 +00:00
pbx Fix cross compiling (bug #3868) 2005-03-27 22:39:17 +00:00
redhat Update spec file 2005-03-16 16:51:45 +00:00
res Little say numer fix (bug #3884) 2005-03-29 06:21:21 +00:00
sounds Add missing sounds 2005-03-24 21:56:58 +00:00
stdtime Merge slimey's Solaris compatibility (with small mods) (bug #2740) 2004-12-14 23:36:30 +00:00
utils Fix cross compiling (bug #3868) 2005-03-27 22:39:17 +00:00
.cleancount Merge API changes for chanspy 2005-03-23 21:52:31 +00:00
.cvsignore Allow me to force a "make clean ; make install" on a cvs update (bug #3358) 2005-01-17 04:48:51 +00:00
BUGS Update Changelog/BUGS 2004-07-17 02:52:52 +00:00
CHANGES Update ChangeLog 2004-11-01 02:43:53 +00:00
CREDITS Merge slimey's Solaris compatibility (with small mods) (bug #2740) 2004-12-14 23:36:30 +00:00
HARDWARE Plane commits (a.k.a. the Delta deltas): 1) Make muted reconnect 2) Add "X" option to meetme and add ${MEETME_EXIT_CONTEXT}, 3) Allow SIP call parking with supervised transfer, 4) Only create parking entries when calls actually get parked, 5) Add "sunshine" song, 6) Update hardware documentation, 7) Don't load empty strings from history file 2004-08-03 06:31:20 +00:00
LICENSE Version 0.1.1 from FTP 1999-12-08 00:16:51 +00:00
Makefile Fix CC (bug #3895) 2005-03-30 06:31:42 +00:00
README Allow me to force a "make clean ; make install" on a cvs update (bug #3358) 2005-01-17 04:48:51 +00:00
README.fpm Add little note about hold music 2004-08-16 17:43:48 +00:00
SECURITY Update security document, work on threading with pbx.c and small SIP fixes 2004-04-02 07:24:33 +00:00
UPGRADE.txt Fix a bunch of const stuff, merge queue changes, add experimental "hybrid" DTMF mode 2005-03-28 20:48:24 +00:00
acl.c Make ACL be what SIP is going to need (bug #2358, just first part) 2005-03-09 05:48:11 +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 Simplify endianness and fix for unaligned reads (bug #3867) 2005-03-29 04:49:24 +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 Fix app bug, update skel example, add skel to makefile as option (bug #3869) 2005-03-27 22:29:57 +00:00
ast_expr.y Fix quad_t (bug #3048) 2004-12-15 16:07:35 +00:00
astconf.h Version 0.3.0 from FTP 2003-01-30 15:03:20 +00:00
asterisk.8.gz Add timestamping to console (bug #3653 with minor mods) 2005-03-11 07:24:10 +00:00
asterisk.c Fix order of priority reading / file reading (bug #3860) 2005-03-26 07:16:18 +00:00
asterisk.h Merge OEJ's channel type listing (bug #3187) with slight modifications 2004-12-31 00:04:41 +00:00
asterisk.sgml Add timestamping to console (bug #3653 with minor mods) 2005-03-11 07:24:10 +00:00
astmm.c Merge Russell's formatting patch (bug #3838) 2005-03-23 05:56:32 +00:00
autoservice.c Rework channel structure to eliminate "pvt" portion of channel (bug #3573) 2005-03-04 06:47:24 +00:00
callerid.c Add new callerpres parsing API (bug #3648) 2005-02-26 07:34:09 +00:00
cdr.c Allow functions to be written to (bug #2278, with mods) 2005-03-29 06:16:49 +00:00
channel.c Fix a bunch of const stuff, merge queue changes, add experimental "hybrid" DTMF mode 2005-03-28 20:48:24 +00:00
chanvars.c Little variable optimizations 2004-11-01 15:48:42 +00:00
cli.c Merge Russell's formatting patch (bug #3838) 2005-03-23 05:56:32 +00:00
coef_in.h Merge UK + DTMF Caller*ID stuff and fix app_test description 2004-09-19 16:17:18 +00:00
coef_out.h Version 0.1.7 from FTP 2001-03-20 20:11:20 +00:00
config.c Fix help and command line completion for "show config mappings" (Bug #3766) 2005-03-17 00:35:06 +00:00
config_old.c Add old config files (bug #3406) 2005-01-25 06:11:11 +00:00
db.c Fix "oopsie" (bug #3603) 2005-02-23 22:58:11 +00:00
dlfcn.c Fix misspellings of separate (bug #3607) 2005-02-16 02:58:18 +00:00
dns.c Simplify endianness and fix for unaligned reads (bug #3867) 2005-03-29 04:49:24 +00:00
dsp.c Rework channel structure to eliminate "pvt" portion of channel (bug #3573) 2005-03-04 06:47:24 +00:00
ecdisa.h Version 0.1.10 from FTP 2001-11-10 20:31:39 +00:00
enum.c Fix casting error (bug #3681, take 2) 2005-03-12 16:58:04 +00:00
file.c Ignore interrupted system calls (bug #3831) 2005-03-22 20:34:04 +00:00
frame.c Add README for jitter buffer (bug #3812), make src char *src a const 2005-03-21 04:30:57 +00:00
fskmodem.c Merge UK + DTMF Caller*ID stuff and fix app_test description 2004-09-19 16:17:18 +00:00
image.c Rework channel structure to eliminate "pvt" portion of channel (bug #3573) 2005-03-04 06:47:24 +00:00
indications.c Fix misspellings of separate (bug #3607) 2005-02-16 02:58:18 +00:00
io.c Make it build and run on MacOS X 2003-10-26 18:50:49 +00:00
jitterbuf.c Add PLC and jitter buffer and iax2 meta trunk with timestamps (bug #2532, #3400) 2005-03-17 21:30:19 +00:00
jitterbuf.h Add PLC and jitter buffer and iax2 meta trunk with timestamps (bug #2532, #3400) 2005-03-17 21:30:19 +00:00
loader.c Merge config updates (bug #3406) 2005-01-25 06:10:20 +00:00
logger.c No need for sizeof (bug #3887) 2005-03-28 20:57:59 +00:00
make_build_h Version 0.1.8 from FTP 2001-05-09 03:11:22 +00:00
manager.c Merge Russell's formatting patch (bug #3838) 2005-03-23 05:56:32 +00:00
md5.c Simplify endianness and fix for unaligned reads (bug #3867) 2005-03-29 04:49:24 +00:00
mkdep Fix mkdep to work with /bin/sh on solaris and friends (bug #3050) 2004-12-17 07:59:26 +00:00
mkpkgconfig Add support for Solaris/x86 (bug #3064) 2005-03-17 23:12:15 +00:00
muted.c update copyright headers for 2005 2005-01-21 07:06:25 +00:00
muted.conf.sample clean up config file sample 2004-05-17 06:39:17 +00:00
pbx.c Make sure ExecIf stuff returns properly (bug #3864) 2005-03-29 06:18:58 +00:00
plc.c Fix PLC for BSD (bug #2532) 2005-03-20 02:53:48 +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 Simplify endianness and fix for unaligned reads (bug #3867) 2005-03-29 04:49:24 +00:00
sample.call Add example of using Account in sample.call file 2004-03-26 08:04:13 +00:00
say.c Repair danish format (bug #3239) 2005-03-27 22:07:39 +00:00
sched.c Minor scheduling fixups 2004-07-20 13:43:33 +00:00
sounds.txt Fix sound files 2005-03-24 03:31:53 +00:00
srv.c REduce chattyness 2004-07-17 22:06:26 +00:00
strcompat.c Add support for Solaris/x86 (bug #3064) 2005-03-17 23:12:15 +00:00
tdd.c Fix a bunch of const stuff, merge queue changes, add experimental "hybrid" DTMF mode 2005-03-28 20:48:24 +00:00
term.c Merge Tilghman's color detection patch (bug #2495) 2004-10-01 03:11:52 +00:00
translate.c Rework channel structure to eliminate "pvt" portion of channel (bug #3573) 2005-03-04 06:47:24 +00:00
ulaw.c Version 0.1.10 from FTP 2001-12-07 22:57:34 +00:00
utils.c Add support for Solaris/x86 (bug #3064) 2005-03-17 23:12:15 +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"

Each time you update or checkout from CVS, you are strongly encouraged 
to ensure all previous object files are removed to avoid internal 
inconsistency in Asterisk. Normally, this is automatically done with 
the presence of the file .cleancount, which increments each time a 'make clean'
is required, and the file .lastclean, which contains the last .cleancount used. 

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