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Mark Spencer 5a69a332e5
Fix to be sure we have a valid fd on a peer
21 years ago
agi Merge slimey's Solaris compatibility (with small mods) (bug #2740) 21 years ago
apps Allow softhangup to work on partial matches if requested (bug #3261) 21 years ago
cdr Fix types for character (bug #3255, take 3) 21 years ago
channels Fix to be sure we have a valid fd on a peer 21 years ago
codecs Handle speex subdirectories peroperly (bug #3283) 21 years ago
configs Make ODBC storage as int configurable to be string or int (bug #3255) 21 years ago
contrib add rawplayer applet to contrib/utils 21 years ago
db1-ast Merge slimey's Solaris compatibility (with small mods) (bug #2740) 21 years ago
doc Update IAX readme (bug #3310) 21 years ago
editline Merge slimey's Solaris compatibility (with small mods) (bug #2740) 21 years ago
formats Mildly resurect G.723.1 file format 21 years ago
images Version 0.1.12 from FTP 23 years ago
include Allow multiple bindaddrs so asterisk uses the same interface for tx as rx 21 years ago
keys Add information for IAX on Free World Dialup 21 years ago
pbx More flagification, courtesy drumkilla (bug #3280) 21 years ago
redhat Add Asterisk manpage 21 years ago
res Fix one touch record (bug #3263, take two) 21 years ago
sounds Add beep error sound 21 years ago
stdtime Merge slimey's Solaris compatibility (with small mods) (bug #2740) 21 years ago
utils Merge Olle's comment patch (bug #3097) 21 years ago
.cvsignore Remove comment about update.out 21 years ago
BUGS Update Changelog/BUGS 21 years ago
CHANGES Update ChangeLog 21 years ago
CREDITS Merge slimey's Solaris compatibility (with small mods) (bug #2740) 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
Makefile Fix voicemail symlinks (bug #3024) 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 Allow multiple bindaddrs so asterisk uses the same interface for tx as rx 21 years ago
aescrypt.c Add AES support 22 years ago
aeskey.c Add AES support 22 years ago
aesopt.h Merge slimey's Solaris compatibility (with small mods) (bug #2740) 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 flagification, courtesy drumkilla (bug #3280) 21 years ago
ast_expr.y Fix quad_t (bug #3048) 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 Merge OEJ's channel type listing (bug #3187) with slight modifications 21 years ago
asterisk.h Merge OEJ's channel type listing (bug #3187) with slight modifications 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 Big diet for struct ast_channel 21 years ago
callerid.c Fix some callerid output 21 years ago
cdr.c More flagification, courtesy drumkilla (bug #3280) 21 years ago
channel.c More flagification, courtesy drumkilla (bug #3280) 21 years ago
chanvars.c Little variable optimizations 21 years ago
cli.c Fix command completion issue (bug #3257) 21 years ago
coef_in.h Merge UK + DTMF Caller*ID stuff and fix app_test description 21 years ago
coef_out.h Version 0.1.7 from FTP 24 years ago
config.c tiny tweak to allow pvt config engines to use __ast_load 21 years ago
db.c Merge tilghman's "showkey" patch (bug #2986) 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 typo in tone detect (bug #3315) 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 Fix voicemail symlinks (bug #3024) 21 years ago
frame.c Merge experimental codec preferences for IAX2 (bug #2971) 21 years ago
fskmodem.c Merge UK + DTMF Caller*ID stuff and fix app_test description 21 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 Merge russell's flag macro patch (with slight mods) (bug #3046) 21 years ago
io.c Make it build and run on MacOS X 22 years ago
loader.c Revert loader change now that we do it at runtime 21 years ago
logger.c fix logging issue 21 years ago
make_build_h Version 0.1.8 from FTP 24 years ago
manager.c Allow connection notifications on manager interface to be hidden (bug #3085) 21 years ago
md5.c Merge slimey's Solaris compatibility (with small mods) (bug #2740) 21 years ago
mkdep Fix mkdep to work with /bin/sh on solaris and friends (bug #3050) 21 years ago
mkpkgconfig Merge slimey's Solaris compatibility (with small mods) (bug #2740) 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 Restore functionality of "show dialplan" with no arguments 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 Support CNG transmission when on hold (bug #2904) 21 years ago
sample.call Add example of using Account in sample.call file 21 years ago
say.c Add partial greek support (bug #3107) 21 years ago
sched.c Minor scheduling fixups 21 years ago
sounds.txt Merge Moc's announcement patch (bug #3219) 21 years ago
srv.c REduce chattyness 21 years ago
strcompat.c Merge slimey's Solaris compatibility (with small mods) (bug #2740) 21 years ago
tdd.c Make sure malloc worked before accessing the mem in tdd.c 21 years ago
term.c Merge Tilghman's color detection patch (bug #2495) 21 years ago
translate.c Minor translation performance improvement (bug #2987, not that patch though) 21 years ago
ulaw.c Version 0.1.10 from FTP 24 years ago
utils.c Include missing file (bug #3277) 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