sysmocom fork of asterisk; containing code of ongoing developments.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6026 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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agi | ||
apps | ||
build_tools | ||
cdr | ||
channels | ||
codecs | ||
configs | ||
contrib | ||
db1-ast | ||
doc | ||
editline | ||
formats | ||
funcs | ||
images | ||
include | ||
keys | ||
patches | ||
pbx | ||
redhat | ||
res | ||
sounds | ||
stdtime | ||
utils | ||
.cleancount | ||
.cvsignore | ||
acl.c | ||
aescrypt.c | ||
aeskey.c | ||
aesopt.h | ||
aestab.c | ||
alaw.c | ||
app.c | ||
ast_expr.y | ||
ast_expr2.fl | ||
ast_expr2.y | ||
asterisk.8.gz | ||
asterisk.c | ||
asterisk.sgml | ||
astmm.c | ||
autoservice.c | ||
BUGS | ||
callerid.c | ||
cdr.c | ||
CHANGES | ||
channel.c | ||
chanvars.c | ||
cli.c | ||
coef_in.h | ||
coef_out.h | ||
config.c | ||
config_old.c | ||
CREDITS | ||
db.c | ||
dlfcn.c | ||
dns.c | ||
dnsmgr.c | ||
dsp.c | ||
ecdisa.h | ||
enum.c | ||
file.c | ||
frame.c | ||
fskmodem.c | ||
HARDWARE | ||
image.c | ||
indications.c | ||
io.c | ||
jitterbuf.c | ||
jitterbuf.h | ||
LICENSE | ||
loader.c | ||
logger.c | ||
Makefile | ||
manager.c | ||
md5.c | ||
mkpkgconfig | ||
muted.c | ||
muted.conf.sample | ||
pbx.c | ||
plc.c | ||
poll.c | ||
privacy.c | ||
README | ||
README.fpm | ||
rtp.c | ||
sample.call | ||
say.c | ||
sched.c | ||
SECURITY | ||
sounds.txt | ||
srv.c | ||
strcompat.c | ||
tdd.c | ||
term.c | ||
translate.c | ||
ulaw.c | ||
UPGRADE.txt | ||
utils.c |
The Asterisk Open Source PBX by Mark Spencer <markster@digium.com> Copyright (C) 2001-2005 Digium, 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.asterisk.org In addition you'll find lots 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 and is also available under alternative licenses negotiated directly with Digium, Inc. If you obtained Asterisk under the GPL, then the GPL applies to all loadable modules used on your system as well, except as defined below. Digium, Inc. (formerly Linux Support Services) retains copyright and/or a sufficient license to all components 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/license 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 exception in your GPL'd products (although if you've written a module for Asterisk we would strongly encourage you to make the same exception that we do). Specific permission is also granted to OpenSSL and OpenH323 to link with 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. * OPERATING SYSTEMS == Linux == The Asterisk Open Source PBX is developed and tested primarily on the GNU/Linux operating system, and is supported on every major GNU/Linux distribution. == Others == Asterisk has also been 'ported' and reportedly runs properly on other operating systems as well, including Sun Solaris, Apple's Mac OS X, and the BSD variants. * GETTING STARTED First, be sure you've got supported hardware (but note that you don't need ANY special hardware, not even a soundcard) to install and run Asterisk. Supported telephony hardware includes: * All Wildcard (tm) products from Digium (www.digium.com) * QuickNet Internet PhoneJack and LineJack (http://www.quicknet.net) * any full duplex sound card supported by ALSA or OSS * ISDN4Linux compatible ISDN card * VoiceTronix OpenLine products Hint: CAPI compatible ISDN cards can be run using the add-on channel chan_capi. Second, ensure that your system contains a compatible compiler and development libraries. Asterisk requires either the GNU Compiler Collection (GCC) version 3.0 or higher, or a compiler that supports the C99 specification and some of the gcc language extensions. In addition, your system needs to have the C library headers available, and the headers and libraries for OpenSSL and zlib. On many distributions, these files are installed by packages with names like 'libc-devel', 'openssl-devel' and 'zlib-devel' or similar. So let's proceed: 1) Run "make" Assuming the build completes successfully: 2) Run "make install" Each time you update or checkout from CVS, you are strongly encouraged to ensure all previous object files are removed to avoid internal inconsistency in Asterisk. Normally, this is automatically done with the presence of the file .cleancount, which increments each time a 'make clean' is required, and the file .lastclean, which contains the last .cleancount used. If this is your first time working with Asterisk, you may wish to install the sample PBX, with demonstration extensions, etc. If so, run: 3) "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) then 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 zapata.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 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 card, obtaining the settings from the variables specified above. * SPECIAL NOTE ON TIME Those using SIP phones should be aware that Asterisk is sensitive to large jumps in time. Manually changing the system time using date(1) (or other similar commands) may cause SIP registrations and other internal processes to fail. If your system cannot keep accurate time by itself use NTP (http://www.ntp.org/) to keep the system clock synchronized to "real time". NTP is designed to keep the system clock synchronized by speeding up or slowing down the system clock until it is synchronized to "real time" rather than by jumping the time and causing discontinuities. Most Linux distributions include precompiled versions of NTP. Beware of some time synchronization methods that get the correct real time periodically and then manually set the system clock. Apparent time changes due to daylight savings time are just that, apparent. The use of daylight savings time in a Linux system is purely a user interface issue and does not affect the operation of the Linux kernel or Asterisk. The system clock on Linux kernels operates on UTC. UTC does not use daylight savings time. Also note that this issue is separate from the clocking of TDM channels, and is known to at least affect SIP registrations. * FILE DESCRIPTORS Depending on the size of your system and your configuration, Asterisk can consume a large number of file descriptors. In UNIX, file descriptors are used for more than just files on disk. File descriptors are also used for handling network communication (e.g. SIP, IAX2, or H.323 calls) and hardware access (e.g. analog and digital trunk hardware). Asterisk accesses many on-disk files for everything from configuration information to voicemail storage. Most systems limit the number of file descriptors that Asterisk can have open at one time. This can limit the number of simultaneous calls that your system can handle. For example, if the limit is set at 1024 (a common default value) Asterisk can handle approxiately 150 SIP calls simultaneously. To change the number of file descriptors follow the instructions for your system below: == PAM-based Linux System == If your system uses PAM (Pluggable Authentication Modules) edit /etc/security/limits.conf. Add these lines to the bottom of the file: root soft nofile 4096 root hard nofile 8196 asterisk soft nofile 4096 asterisk hard nofile 8196 (adjust the numbers to taste). You may need to reboot the system for these changes to take effect. == Generic UNIX System == If there are no instructions specifically adapted to your system above you can try adding the command "ulimit -n 8192" to the script that starts Asterisk. * 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