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Mark Spencer 3e8bdfc9c9
Fix "0" auto deleting messages from voicemail (bug #3057)
21 years ago
agi Merge slimey's Solaris compatibility (with small mods) (bug #2740) 21 years ago
apps Fix "0" auto deleting messages from voicemail (bug #3057) 21 years ago
cdr Merge Olle's comment patch (bug #3097) 21 years ago
channels Merge OEJ's print groups feature (bug #3228, with changes) 21 years ago
codecs Fix Speex config issue (bug #3175) 21 years ago
configs Fix formatting etc in queues (bug #3159) 21 years ago
contrib update astxs (small tweak) 21 years ago
db1-ast Merge slimey's Solaris compatibility (with small mods) (bug #2740) 21 years ago
doc Fix formatting etc in queues (bug #3159) 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
include Merge OEJ's print groups feature (bug #3228, with changes) 21 years ago
keys
pbx List improvements from kpfleming (bugs #3166,#3140) 21 years ago
redhat
res Merge kpflemings moh_files fixes (bug #3224) 21 years ago
sounds Fix "0" auto deleting messages from voicemail (bug #3057) 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
CHANGES Update ChangeLog 21 years ago
CREDITS Merge slimey's Solaris compatibility (with small mods) (bug #2740) 21 years ago
HARDWARE
LICENSE
Makefile Show conflits on make update (bug #3191) 21 years ago
README Fixed?? 21 years ago
README.fpm
SECURITY
acl.c Fix leak of file descriptors in odd networking situations 21 years ago
aescrypt.c
aeskey.c
aesopt.h Merge slimey's Solaris compatibility (with small mods) (bug #2740) 21 years ago
aestab.c
alaw.c
app.c Make voicemail registration apps register (bug #3034) 21 years ago
ast_expr.y Fix quad_t (bug #3048) 21 years ago
astconf.h
asterisk.8.gz
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
astmm.c
autoservice.c Big diet for struct ast_channel 21 years ago
callerid.c Fix some callerid output 21 years ago
cdr.c rollback stupid code 21 years ago
channel.c Handle group memberships better with masquerade (bug #3067) 21 years ago
chanvars.c Little variable optimizations 21 years ago
cli.c Fix additional typos (bug #3162) 21 years ago
coef_in.h Merge UK + DTMF Caller*ID stuff and fix app_test description 21 years ago
coef_out.h
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
dns.c
dsp.c Merge twisted's dsp formatting fixes (bug #3218) 21 years ago
ecdisa.h
enum.c
file.c Merge anthm's native MOH patch (bug #2639) he promises me he'll rid it of ast_flag_moh... 21 years ago
frame.c fix case sensitivity issue in codecs 21 years ago
fskmodem.c Merge UK + DTMF Caller*ID stuff and fix app_test description 21 years ago
image.c
indications.c Merge russell's flag macro patch (with slight mods) (bug #3046) 21 years ago
io.c
loader.c fix logging issue 21 years ago
logger.c fix logging issue 21 years ago
make_build_h
manager.c Fix possible race... 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
muted.conf.sample
pbx.c List improvements from kpfleming (bugs #3166,#3140) 21 years ago
poll.c
privacy.c
rtp.c Ignore invalid RTP packets (bug #3030) 21 years ago
sample.call
say.c Add partial greek support (bug #3107) 21 years ago
sched.c
sounds.txt Fix "0" auto deleting messages from voicemail (bug #3057) 21 years ago
srv.c
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
utils.c Merge OEJ's print groups feature (bug #3228, with changes) 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