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Mark Spencer 47f8883942
Add test application (TestClient/TestServer), thanks Russell!
21 years ago
agi Merge major BSD mutex and symbol conflict patches (bug #1816) (link patch still pending) 21 years ago
apps Add test application (TestClient/TestServer), thanks Russell! 21 years ago
astman Fix astman build on FreeBSD (bug #2119) 21 years ago
cdr Build on older versions of TDS (bug #2194) 21 years ago
channels Set DTMF modes by peer/user properly (bug #2303) 21 years ago
codecs Minor Makefile cleanups 21 years ago
configs Add new "route" mode to work around UNIDEN bugs (bug #2308) 21 years ago
contrib fix a bug in some small changes to astxs utility 21 years ago
db1-ast Merge remaining audit patch (save dlfcn.c) 21 years ago
doc Add README.mp3 to docs to clear up confusion about working and 21 years ago
editline Merge remaining audit patch (save dlfcn.c) 21 years ago
formats fixed a use count bug 21 years ago
images Version 0.1.12 from FTP 23 years ago
include/asterisk Don't use "class" keyword in music on hold (bug #2316) 21 years ago
keys Add information for IAX on Free World Dialup 21 years ago
pbx Save CID and switches in "save dialplan" command (bug #2279) 21 years ago
redhat Add Asterisk manpage 21 years ago
res make bug 2272 (which is actually caused by human error) less likely to happen 21 years ago
sounds Merge MOG's first serious patch (new message patch) (bug #2071) 21 years ago
stdtime Merge remaining audit patch (save dlfcn.c) 21 years ago
utils Fix astman build on FreeBSD (bug #2119) 21 years ago
.cvsignore Add support for E1 E&M 21 years ago
BUGS Update Changelog/BUGS 21 years ago
CHANGES Make Asterisk cause codes match those of Q.931 (bug #1999) 21 years ago
CREDITS Add CREDITS credit for www.freeplaymusic.com - Hold Music 21 years ago
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 21 years ago
LICENSE Version 0.1.1 from FTP 26 years ago
Makefile Don't install mpg123 unless it was built 21 years ago
README Fixed?? 21 years ago
README.fpm Add little note about hold music 21 years ago
SECURITY Update security document, work on threading with pbx.c and small SIP fixes 21 years ago
acl.c Use INET_ADDRLEN (bug #1956) (from airport!) 21 years ago
aescrypt.c Add AES support 22 years ago
aeskey.c Add AES support 22 years ago
aesopt.h Fix AES for MacOS build 21 years ago
aestab.c Add AES support 22 years ago
alaw.c Version 0.1.10 from FTP 24 years ago
app.c More strcpy / snprintf as part of rgagnon's audit (bug #2004) 21 years ago
ast_expr.y Merge source cleanups (bug #1911) 21 years ago
astconf.h Version 0.3.0 from FTP 23 years ago
asterisk.8.gz Add Asterisk manpage 21 years ago
asterisk.c Warn if unable to open an overridden config file (but #2285) 21 years ago
asterisk.h Close logging stuff so system doesn't have to (bug #1855) 21 years ago
asterisk.sgml Add Asterisk manpage 21 years ago
astmm.c Remove pthread.h from source. We should be using asterisk/lock.h everywhere instead (except in asterisk/lock.h). 21 years ago
autoservice.c Merge BSD stack size work (bug #2067) 21 years ago
callerid.c Demand more information on callerid to prevent possible overruns 21 years ago
cdr.c More strcpy / snprintf as part of rgagnon's audit (bug #2004) 21 years ago
channel.c Fix generator for VAD as well as for automatically syncing to incoming signal if present (bug #2312) 21 years ago
chanvars.c Include fixes for portability 22 years ago
cli.c Fix a couple minor command line completion issues 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 Allow on/off (bug #2233) 21 years ago
db.c More strcpy / snprintf as part of rgagnon's audit (bug #2004) 21 years ago
dlfcn.c Make it build and run on MacOS X 22 years ago
dns.c Misc formatting cleanups 21 years ago
dsp.c Fix divide by zero (bugs #2268 and 2259) 21 years ago
ecdisa.h Version 0.1.10 from FTP 24 years ago
enum.c Remaining rgagnon source audit improvements (bug #2011) 21 years ago
file.c If breakon is unspecified, make it "" 21 years ago
frame.c Merge mic's minor patchlet (bug #2092) 21 years ago
fskmodem.c Version 0.1.10 from FTP 24 years ago
image.c Remove pthread.h from source. We should be using asterisk/lock.h everywhere instead (except in asterisk/lock.h). 21 years ago
indications.c Need actual offset space (bug #2076) 21 years ago
io.c Make it build and run on MacOS X 22 years ago
loader.c Remove pthread.h from source. We should be using asterisk/lock.h everywhere instead (except in asterisk/lock.h). 21 years ago
logger.c Fix logger issue at hangup 21 years ago
make_build_h Version 0.1.8 from FTP 24 years ago
manager.c Fix missing \r\n 21 years ago
md5.c Fix AES for MacOS build 21 years ago
mkdep Make mkdep throw away stderr since people think the error messages printed are serious when they are not 21 years ago
muted.c 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 21 years ago
muted.conf.sample clean up config file sample 21 years ago
pbx.c Merge "show applications" from corydon76 (bug #2291) 21 years ago
poll.c Make it build and run on MacOS X 22 years ago
privacy.c Remove pthread.h from source. We should be using asterisk/lock.h everywhere instead (except in asterisk/lock.h). 21 years ago
rtp.c Repair offer/answer model (bug #2293), initial CNG work for new frametype 21 years ago
sample.call Add example of using Account in sample.call file 21 years ago
say.c Make the polish speech not such an insane coding style 21 years ago
sched.c Minor scheduling fixups 21 years ago
sounds.txt Merge MOG's first serious patch (new message patch) (bug #2071) 21 years ago
srv.c REduce chattyness 21 years ago
tdd.c Version 0.1.10 from FTP 24 years ago
term.c Remove pthread.h from source. We should be using asterisk/lock.h everywhere instead (except in asterisk/lock.h). 21 years ago
translate.c Rename newp to newpvt (bug #2190), change hold music. 21 years ago
ulaw.c Version 0.1.10 from FTP 24 years ago
utils.c Merge "show applications" from corydon76 (bug #2291) 21 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