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Mark Spencer cc1da2eb5b
Qualify that SIP INFO stuff is real (bug #1558)
21 years ago
agi Clean agi-sphinx-test (bug #1547) 21 years ago
apps Make sure we don't accidently send weird delivery times on DISA 21 years ago
astman gethostbyname isn't reentrant, who knew... 21 years ago
cdr Create individual sip reload command (bug #880) 22 years ago
channels Qualify that SIP INFO stuff is real (bug #1558) 21 years ago
codecs Fix iLBC with valgrind, add iLBC format from bkw_ 21 years ago
configs Add "insecure=very" where we don't authenticate peers who are registered 21 years ago
contrib update astxs to default /usr/src/asterisk 21 years ago
db1-ast Various warning cleanups 22 years ago
doc Documentation fixes (bug #1554). 21 years ago
editline Move config.cache delete to "distclean" 22 years ago
formats format_ilbc.c comment fix from bkw 21 years ago
images Version 0.1.12 from FTP 23 years ago
include/asterisk Add include/asterisk/utils.h file. Which includes the function 21 years ago
keys Version 0.1.10 from FTP 24 years ago
pbx Internationalize say_date_time, fix small pbx_config seglet (bug #1537) 21 years ago
redhat Add the SuSE AMD64 support and fixes from Bug #706 22 years ago
res Unbuffered music on hold 21 years ago
sounds Add SayPhonetic and SayAlpha applications (bug #793) 21 years ago
stdtime Get .depend for stdtime 22 years ago
utils gethostbyname isn't reentrant, who knew... 21 years ago
.cvsignore Add support for E1 E&M 21 years ago
BUGS Fix BUGS document to report bug tracker 22 years ago
CHANGES Merge queue changes from Bug #214 21 years ago
CREDITS add ww 22 years ago
HARDWARE Update HARDWARE 22 years ago
LICENSE Version 0.1.1 from FTP 26 years ago
Makefile Add SayPhonetic and SayAlpha applications (bug #793) 21 years ago
README Fix 2 typos in README 22 years ago
SECURITY Update security document, work on threading with pbx.c and small SIP fixes 21 years ago
acl.c Optimize inaddrcmp (a little) by making it inline 21 years ago
aescrypt.c Add AES support 22 years ago
aeskey.c Add AES support 22 years ago
aesopt.h Qualify that SIP INFO stuff is real (bug #1558) 21 years ago
aestab.c Add AES support 22 years ago
alaw.c Version 0.1.10 from FTP 24 years ago
app.c Change strlen calls to ast_strlen_zero in voicemail checking stuff because it is called all the time 21 years ago
ast_expr.y More expression fixes (bug #1548 again) 21 years ago
astconf.h Version 0.3.0 from FTP 23 years ago
asterisk.c More strlen_zero checks (bug #1549) 21 years ago
asterisk.h Have a contact line in responses, merge logging patches 22 years ago
astmm.c add a vasprintf replacement. Bug #839 22 years ago
autoservice.c Fix bug 1217. Change pthread_t initializers to AST_PTHREADT_NULL and 21 years ago
callerid.c Change strlen calls to ast_strlen_zero in callerid.c 21 years ago
cdr.c Log that we are unregistering cdr module (bug 1460) 21 years ago
channel.c Code formatting cleanup in channel.c 21 years ago
chanvars.c Include fixes for portability 22 years ago
cli.c Free some cli memory 21 years ago
coef_in.h Version 0.1.7 from FTP 24 years ago
coef_out.h Version 0.1.7 from FTP 24 years ago
config.c More strlen_zero checks (bug #1549) 21 years ago
db.c Make valgrind happy on db read 22 years ago
dlfcn.c Make it build and run on MacOS X 22 years ago
dns.c Unify all the res_ninit patches 21 years ago
dsp.c When creating a new DSP, initialize the progress zone just in case 21 years ago
ecdisa.h Version 0.1.10 from FTP 24 years ago
enum.c More strlen_zero checks (bug #1549) 21 years ago
file.c Fix format unregister 21 years ago
frame.c ast_frdup optimization: only call strlen once and save the result 21 years ago
fskmodem.c Version 0.1.10 from FTP 24 years ago
image.c image unregister typo 21 years ago
indications.c Get rid of all that old needlock garbage now that we're using recursive mutexes 21 years ago
io.c Make it build and run on MacOS X 22 years ago
loader.c Loader fixes 21 years ago
logger.c Typo in logger.c (bug 1466) 21 years ago
make_build_h Version 0.1.8 from FTP 24 years ago
manager.c Allow "fast" asynchronous manager initiation of events (bug #772) 21 years ago
md5.c OpenBSD portability enhancements (bug 1002) 21 years ago
mkdep FreeBSD compatability fixes 22 years ago
pbx.c Change strlen calls to ast_strlen_zero in pbx.c 21 years ago
poll.c Make it build and run on MacOS X 22 years ago
privacy.c Version 0.3.0 from FTP 23 years ago
rtp.c Add DTX support (mark bit) (bug #1234) 21 years ago
sample.call Add example of using Account in sample.call file 21 years ago
say.c More strlen_zero checks (bug #1549) 21 years ago
sched.c Unlock while processing schedule queue 22 years ago
sounds.txt Add SayPhonetic and SayAlpha applications (bug #793) 21 years ago
srv.c More strlen_zero checks (bug #1549) 21 years ago
tdd.c Version 0.1.10 from FTP 24 years ago
term.c Merge Tilghman's color patches for the asterisk prompt (bug #1535) 21 years ago
translate.c Log when we unload a translator (bug 1460) 21 years ago
ulaw.c Version 0.1.10 from FTP 24 years ago

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