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Mark Spencer f0774e7a6c
Take out DOS returns
22 years ago
agi Add DESTDIR support (bug #200) 23 years ago
apps Don't close audio[0] if you're not careful 22 years ago
astman Various warning cleanups 23 years ago
cdr prevent deadlock if no config file 22 years ago
channels re-do the monitor fix (check for io before killing ourself) 22 years ago
codecs Various warning cleanups 23 years ago
configs Add auth debug option 22 years ago
contrib Fix SQL description for longer channels 22 years ago
db1-ast Various warning cleanups 23 years ago
doc Update README for timestamp 23 years ago
editline Various warning cleanups 23 years ago
formats Remove really broke MP3 stuff in favor of G.726 in the near future 23 years ago
images
include/asterisk Add AES support 22 years ago
keys
pbx scanf != sscanf 22 years ago
redhat
res Hangup calling channel when transferring peer 23 years ago
sounds Add voicemail prepending feature plus forwarding to many extensions if you specify exten1*exten2*.....# 22 years ago
stdtime Get .depend for stdtime 23 years ago
utils Various warning cleanups 23 years ago
.cvsignore
BUGS
CHANGES Update Changelog and README 23 years ago
CREDITS Trustingly add Thorston's deadlock patch 23 years ago
HARDWARE
LICENSE
Makefile Add AES support 22 years ago
README
README.cdr
README.festival
README.iax
README.messages-expire Add message expiry from cron (bug #388) 23 years ago
README.mysql Remove MySQL support from default Asterisk in accordance with new MySQL library licensing 23 years ago
README.variables Update README for timestamp 23 years ago
SECURITY
acl.c Route fixes for OpenBSD (bug #415) 23 years ago
addmailbox Update addmailbox script (bug #404) 23 years ago
aescrypt.c Add AES support 22 years ago
aeskey.c Add AES support 22 years ago
aesopt.h Add AES support 22 years ago
aestab.c Add AES support 22 years ago
alaw.c
app.c
ast_expr.y Code cleanups (bug #66) 23 years ago
astconf.h
asterisk-ng-doxygen
asterisk.c Fix typo 22 years ago
asterisk.h Have a contact line in responses, merge logging patches 23 years ago
astgenkey
astmm.c Fix astmm for new build process 23 years ago
autoservice.c BSD portability enhancements (bug #234) 23 years ago
callerid.c Typo 23 years ago
cdr.c Don't complain that wait4 is unkown and make sure that we won't segfault if chan->cdr is NULL 23 years ago
channel.c Gotta actually free the frame... 23 years ago
chanvars.c
cli.c Revert bad patch in 187 22 years ago
coef_in.h
coef_out.h
config.c
db.c Make valgrind happy on db read 23 years ago
dlfcn.c Make it build and run on MacOS X 23 years ago
dns.c Make it build and run on MacOS X 23 years ago
dsp.c Use new code by default 22 years ago
ecdisa.h
enum.c Minor enum improvements for iax/iax2 23 years ago
festival-1.4.1-diff
festival-1.4.2.diff
festival-1.4.3.diff Lose the 1.4.1 diff, add a 1.4.3 diff 23 years ago
file.c Add voicemail prepending feature plus forwarding to many extensions if you specify exten1*exten2*.....# 22 years ago
frame.c Remove really broke MP3 stuff in favor of G.726 in the near future 23 years ago
fskmodem.c
image.c Show the names of the codecs instead of the numbers (bug #92) 23 years ago
indications.c
init.asterisk
io.c Make it build and run on MacOS X 23 years ago
loader.c Don't allow to issue a 'reload' command if the previous one didn't finish yet 23 years ago
logger.c Fix minor typo 22 years ago
make_build_h
manager.c Add variable/account code to manager create 23 years ago
md5.c
messages-expire.pl Add message expiry from cron (bug #388) 23 years ago
mkdep FreeBSD compatability fixes 23 years ago
pbx.c Add ${TIMESTAMP} (bug #607) and don't ever ast_log from within handler 23 years ago
poll.c Make it build and run on MacOS X 23 years ago
postgres_cdr.sql Fix SQL description for longer channels 22 years ago
privacy.c
retrieve_extensions_from_mysql.pl Add the flags column so that if it's set to '1' then that record is not included in the output extensions file 23 years ago
retrieve_sip_conf_from_mysql.pl Add flags column so that we can exclude some records from being published in the output file 23 years ago
rtp.c Change the warning message if we can't do native bridge because of diffrent codecs 23 years ago
safe_asterisk
sample.call
say.c Fix hours 21-23 (bug #592) 22 years ago
sched.c Unlock while processing schedule queue 23 years ago
sounds.txt Add voicemail prepending feature plus forwarding to many extensions if you specify exten1*exten2*.....# 22 years ago
srv.c More cleanups and OSX fixes for 10.3 23 years ago
tdd.c
term.c Add "crt" to list that knows colorization (#410) 23 years ago
translate.c Fix bug #111 23 years ago
ulaw.c
valgrind-RedHat-8.0.supp
vmail.cgi
vmdb.sql

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