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Mark Spencer e707a89e63
Allow globals in extensions.conf to reference previous variables
22 years ago
agi Add and update .cvsignore files for .depend 22 years ago
apps Implement remaining queue strategies, ADSI fixes, and queue config updates 22 years ago
astman Add and update .cvsignore files for .depend 22 years ago
cdr Fix the INSERT for the disposition in the string 22 years ago
channels Honor context in agent callback login 22 years ago
codecs One more 22 years ago
configs Implement remaining queue strategies, ADSI fixes, and queue config updates 22 years ago
contrib Fix couple of bugs 22 years ago
db1-ast More BSD enhancements 22 years ago
doc Change the CREATE TABLE so that desposition would be a vartext (ANSWER, BUSY, etc) 22 years ago
editline More BSD enhancements 22 years ago
formats Properly implement using zaptel for timing of file playback 22 years ago
images Version 0.1.12 from FTP 23 years ago
include/asterisk Add the possibility to delete all the contexts registered by a certain registrar with ast_merge_and_delete routine; make it the default behaviour when reloding extensions 22 years ago
keys Version 0.1.10 from FTP 24 years ago
pbx Allow globals in extensions.conf to reference previous variables 22 years ago
redhat Version 0.3.0 from FTP 23 years ago
res Minor fixes 22 years ago
sounds Add sounds and descriptions 22 years ago
utils Sun Mar 16 07:00:01 CET 2003 22 years ago
.cvsignore Add and update .cvsignore files for .depend 22 years ago
BUGS Version 0.1.10 from FTP 24 years ago
CHANGES Implement remaining queue strategies, ADSI fixes, and queue config updates 22 years ago
CREDITS Merge Vonage's MySQL Voicemail stuff 22 years ago
HARDWARE Version 0.3.0 from FTP 23 years ago
LICENSE Version 0.1.1 from FTP 26 years ago
Makefile Move the PROC lines so that it works when VIA ppl change that variable 22 years ago
README Version 0.1.12 from FTP 23 years ago
README.cdr Version 0.1.10 from FTP 24 years ago
README.festival Version 0.3.0 from FTP 23 years ago
README.iax Version 0.1.10 from FTP 24 years ago
README.variables Add unique identifier 22 years ago
SECURITY Version 0.1.10 from FTP 24 years ago
acl.c Take out useless cast 22 years ago
addmailbox Version 0.1.9 from FTP 24 years ago
alaw.c Version 0.1.10 from FTP 24 years ago
app.c Use digit/response timeouts 22 years ago
ast_expr.y Version 0.3.0 from FTP 23 years ago
astconf.h Version 0.3.0 from FTP 23 years ago
asterisk-ng-doxygen Version 0.1.10 from FTP 24 years ago
asterisk.c Don't delete PID file on exit of asterisk -r 22 years ago
asterisk.h Version 0.3.0 from FTP 23 years ago
astgenkey Version 0.1.10 from FTP 24 years ago
astmm.c Fix typo 22 years ago
autoservice.c Version 0.3.0 from FTP 23 years ago
callerid.c Eliminate localtime calls, various cleanups 22 years ago
cas.h Version 0.1.8 from FTP 24 years ago
cdr.c Add unique identifier 22 years ago
channel.c Fix some substantial locking issues 22 years ago
chanvars.c Include fixes for portability 22 years ago
cli.c Minor cleanups 22 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 Add agent groupings, fix the "incorrect" message on first login attempt 22 years ago
db.c Add ast_db_freetree and ast_db_gettree 22 years ago
dsp.c Add the second way of signalizing hangup when busydetect detects the busy tone 22 years ago
ecdisa.h Version 0.1.10 from FTP 24 years ago
enum.c Add SRV code to SIP, cleanup ENUM and make IAX2 do the right thing on dials 22 years ago
festival-1.4.1-diff Version 0.2.0 from FTP 23 years ago
festival-1.4.2.diff Version 0.3.0 from FTP 23 years ago
file.c Still store timing 22 years ago
frame.c Add SIP/RTP video support, video enable app_echo, start on RTCP 22 years ago
fskmodem.c Version 0.1.10 from FTP 24 years ago
image.c Version 0.3.0 from FTP 23 years ago
indications.c Fix the playtones app so that we can pass the tones as an argument ( we don't need to refer to a defined tone in indications.conf ) 22 years ago
init.asterisk Version 0.3.0 from FTP 23 years ago
io.c Version 0.3.0 from FTP 23 years ago
loader.c Make RTP ports configurable 22 years ago
logger.c Eliminate localtime calls, various cleanups 22 years ago
make_build_h Version 0.1.8 from FTP 24 years ago
manager.c Extend manager originate functionality 22 years ago
md5.c Version 0.1.12 from FTP 23 years ago
mkdep Beginning of solaris portability 22 years ago
pbx.c Allow globals in extensions.conf to reference previous variables 22 years ago
privacy.c Version 0.3.0 from FTP 23 years ago
retrieve_extensions_from_mysql.pl Fix couple of bugs 22 years ago
rtp.c Minor rtp fixup 22 years ago
safe_asterisk Add debugging to safe_asterisk 22 years ago
sample.call Version 0.3.0 from FTP 23 years ago
sas.h Version 0.1.8 from FTP 24 years ago
say.c Add commonly used include headers 22 years ago
sched.c Merge / correct MM's patches 22 years ago
sounds.txt Fix sounds descriptions, add more info to chan_agent 22 years ago
srv.c Add missing srv.c and srv.h files 22 years ago
tdd.c Version 0.1.10 from FTP 24 years ago
term.c dom mar 16 23:37:23 CET 2003 22 years ago
translate.c dom mar 16 23:37:23 CET 2003 22 years ago
ulaw.c Version 0.1.10 from FTP 24 years ago
valgrind-RedHat-8.0.supp Wed Mar 19 07:00:01 CET 2003 22 years ago
vmail.cgi Fix vmail "taint" issue 22 years ago
vmdb.sql Merge Vonage changes to VM2, ready to be edited and updated :) 22 years ago
zonedata.c Version 0.3.0 from FTP 23 years ago

README

The Asterisk Open Source PBX
by Mark Spencer <markster@linux-support.net>
Copyright (C) 2001, Linux Support Services, Inc.
================================================================
* 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.asteriskpbx.com

* LICENSING
  Asterisk is distributed under GNU General Public License.  The GPL also
must apply to all loadable modules as well, except as defined below.

  Linux Support Services, Inc. retains copyright to all of the core
Asterisk system, and therefore can grant, at its sole discression, 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.  At
this time (5/21/2001) the only component of Asterisk which is covered
under GPL and not under our Copyright is the Xing MP3 decoder.

  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 excemption 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.
  
* 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
as well.


* GETTING STARTED

First, be sure you've got supported hardware.  To use Asterisk right now,
you will need one of the following:

	* All Wildcard (tm) products from LSS (www.linux-support.net)
	* 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)

Assuming you have one of these (most likely the third) you're ready to 
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.

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

Finally, you may wish to visit the web site and join the mailing list if
you're interested in getting more information.

Mark