1 The Asterisk Open Source PBX
2 by Mark Spencer <markster@digium.com>
3 and the Asterisk.org developer community
5 Copyright (C) 2001-2005 Digium, Inc.
6 and other copyright holders.
7 ================================================================
10 It is imperative that you read and fully understand the contents of
11 the SECURITY file before you attempt to configure and run an Asterisk
15 Asterisk is an Open Source PBX and telephony toolkit. It is, in a
16 sense, middleware between Internet and telephony channels on the bottom,
17 and Internet and telephony applications at the top. For more information
18 on the project itself, please visit the Asterisk home page at:
20 http://www.asterisk.org
22 In addition you'll find lots of information compiled by the Asterisk
23 community on this Wiki:
25 http://www.voip-info.org/wiki-Asterisk
27 There is a book on Asterisk published by O'Reilly under the
28 Creative Commons License. It is available in book stores as well
29 as in a downloadable version on the http://www.asteriskdocs.org
32 * SUPPORTED OPERATING SYSTEMS
35 The Asterisk Open Source PBX is developed and tested primarily on the
36 GNU/Linux operating system, and is supported on every major GNU/Linux
40 Asterisk has also been 'ported' and reportedly runs properly on other
41 operating systems as well, including Sun Solaris, Apple's Mac OS X, and
46 First, be sure you've got supported hardware (but note that you don't need
47 ANY special hardware, not even a soundcard) to install and run Asterisk.
49 Supported telephony hardware includes:
51 * All Wildcard (tm) products from Digium (www.digium.com)
52 * QuickNet Internet PhoneJack and LineJack (http://www.quicknet.net)
53 * any full duplex sound card supported by ALSA or OSS
54 * VoiceTronix OpenLine products
56 The are several drivers for ISDN BRI cards available from third party sources.
57 Check the voip-info.org wiki for more information on chan_capi, chan_misdn and
60 * UPGRADING FROM VERSION 1.0
62 If you are updating from a previous version of Asterisk, make sure you
63 read the UPGRADE.txt file in the source directory. There are some files
64 and configuration options that you will have to change, even though we
65 made every effort possible to maintain backwards compatibility.
67 In order to discover new features to use, please check the configuration
68 examples in the /configs directory of the source code distribution.
69 To discover the major new features of Asterisk 1.2, please visit
70 http://www.astricon.net/asterisk1-2/
74 Ensure that your system contains a compatible compiler and development
75 libraries. Asterisk requires either the GNU Compiler Collection (GCC) version
76 3.0 or higher, or a compiler that supports the C99 specification and some of
77 the gcc language extensions. In addition, your system needs to have the C
78 library headers available, and the headers and libraries for OpenSSL,
80 On many distributions, these files are installed by packages with names like
81 'glibc-devel', 'ncurses-devel', 'openssl-devel' and 'zlib-devel' or similar.
87 Assuming the build completes successfully:
91 Each time you update or checkout from CVS, you are strongly encouraged
92 to ensure all previous object files are removed to avoid internal
93 inconsistency in Asterisk. Normally, this is automatically done with
94 the presence of the file .cleancount, which increments each time a 'make clean'
95 is required, and the file .lastclean, which contains the last .cleancount used.
97 If this is your first time working with Asterisk, you may wish to install
98 the sample PBX, with demonstration extensions, etc. If so, run:
102 Doing so will overwrite any existing config files you have.
104 Finally, you can launch Asterisk in the foreground mode (not a daemon)
109 You'll see a bunch of verbose messages fly by your screen as Asterisk
110 initializes (that's the "very very verbose" mode). When it's ready, if
111 you specified the "c" then you'll get a command line console, that looks
116 You can type "help" at any time to get help with the system. For help
117 with a specific command, type "help <command>". To start the PBX using
118 your sound card, you can type "dial" to dial the PBX. Then you can use
119 "answer", "hangup", and "dial" to simulate the actions of a telephone.
120 Remember that if you don't have a full duplex sound card (and Asterisk
121 will tell you somewhere in its verbose messages if you do/don't) then it
122 won't work right (not yet).
124 "man asterisk" at the Unix/Linux command prompt will give you detailed
125 information on how to start and stop Asterisk, as well as all the command
126 line options for starting Asterisk.
128 Feel free to look over the configuration files in /etc/asterisk, where
129 you'll find a lot of information about what you can do with Asterisk.
131 * ABOUT CONFIGURATION FILES
133 All Asterisk configuration files share a common format. Comments are
134 delimited by ';' (since '#' of course, being a DTMF digit, may occur in
135 many places). A configuration file is divided into sections whose names
136 appear in []'s. Each section typically contains two types of statements,
137 those of the form 'variable = value', and those of the form 'object =>
138 parameters'. Internally the use of '=' and '=>' is exactly the same, so
139 they're used only to help make the configuration file easier to
140 understand, and do not affect how it is actually parsed.
142 Entries of the form 'variable=value' set the value of some parameter in
143 asterisk. For example, in zapata.conf, one might specify:
147 in order to indicate to Asterisk that the switch they are connecting to is
148 of the type "national". In general, the parameter will apply to
149 instantiations which occur below its specification. For example, if the
150 configuration file read:
152 switchtype = national
158 the "national" switchtype would be applied to channels one through
159 four and channels 10 through 12, whereas the "dms100" switchtype would
160 apply to channels 25 through 47.
162 The "object => parameters" instantiates an object with the given
163 parameters. For example, the line "channel => 25-47" creates objects for
164 the channels 25 through 47 of the card, obtaining the settings
165 from the variables specified above.
167 * SPECIAL NOTE ON TIME
169 Those using SIP phones should be aware that Asterisk is sensitive to
170 large jumps in time. Manually changing the system time using date(1)
171 (or other similar commands) may cause SIP registrations and other
172 internal processes to fail. If your system cannot keep accurate time
173 by itself use NTP (http://www.ntp.org/) to keep the system clock
174 synchronized to "real time". NTP is designed to keep the system clock
175 synchronized by speeding up or slowing down the system clock until it
176 is synchronized to "real time" rather than by jumping the time and
177 causing discontinuities. Most Linux distributions include precompiled
178 versions of NTP. Beware of some time synchronization methods that get
179 the correct real time periodically and then manually set the system
182 Apparent time changes due to daylight savings time are just that,
183 apparent. The use of daylight savings time in a Linux system is
184 purely a user interface issue and does not affect the operation of the
185 Linux kernel or Asterisk. The system clock on Linux kernels operates
186 on UTC. UTC does not use daylight savings time.
188 Also note that this issue is separate from the clocking of TDM
189 channels, and is known to at least affect SIP registrations.
193 Depending on the size of your system and your configuration,
194 Asterisk can consume a large number of file descriptors. In UNIX,
195 file descriptors are used for more than just files on disk. File
196 descriptors are also used for handling network communication
197 (e.g. SIP, IAX2, or H.323 calls) and hardware access (e.g. analog and
198 digital trunk hardware). Asterisk accesses many on-disk files for
199 everything from configuration information to voicemail storage.
201 Most systems limit the number of file descriptors that Asterisk can
202 have open at one time. This can limit the number of simultaneous
203 calls that your system can handle. For example, if the limit is set
204 at 1024 (a common default value) Asterisk can handle approxiately 150
205 SIP calls simultaneously. To change the number of file descriptors
206 follow the instructions for your system below:
208 == PAM-based Linux System ==
210 If your system uses PAM (Pluggable Authentication Modules) edit
211 /etc/security/limits.conf. Add these lines to the bottom of the file:
213 root soft nofile 4096
214 root hard nofile 8196
215 asterisk soft nofile 4096
216 asterisk hard nofile 8196
218 (adjust the numbers to taste). You may need to reboot the system for
219 these changes to take effect.
221 == Generic UNIX System ==
223 If there are no instructions specifically adapted to your system
224 above you can try adding the command "ulimit -n 8192" to the script
225 that starts Asterisk.
229 See the doc directory for more documentation on various features. Again,
230 please read all the configuration samples that include documentation on
231 the configuration options.
233 Finally, you may wish to visit the web site and join the mailing list if
234 you're interested in getting more information.
236 http://www.asterisk.org/support
238 Welcome to the growing worldwide community of Asterisk users!