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The Asterisk Open Source PBX by Mark Spencer <firstname.lastname@example.org> 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. * SPECIAL NOTE ON TIME Those using SIP phones should be aware the 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