0a46be95cb
One of the problems with TONE_DETECT as it was originally written is that if a tone is detected multiple times, it can trigger the redirect logic multiple times as well. For example, if we do an async goto in the dialplan after detecting a tone, because the detector is still active until explicitly disabled, if we detect the tone again, we will branch again and start executing that dialplan a second time. This is rarely ever desired behavior, and can happen if the detector is not removed quickly enough. Add a new option, 'e', which automatically disables the detector once the desired number of matches have been heard. This eliminates the potential race condition where previously the detector would need to be disabled immediately, but doing so quickly enough was not guaranteed. This also allows match criteria to be retained longer if needed, so the detector does not need to be destroyed prematurely. Resolves: #1390 UserNote: The 'e' option for TONE_DETECT now allows detection to be disabled automatically once the desired number of matches have been fulfilled, which can help prevent race conditions in the dialplan, since TONE_DETECT does not need to be disabled after a hit. |
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| addons | ||
| agi | ||
| apps | ||
| autoconf | ||
| bridges | ||
| build_tools | ||
| cdr | ||
| cel | ||
| channels | ||
| codecs | ||
| configs | ||
| contrib | ||
| doc | ||
| formats | ||
| funcs | ||
| images | ||
| include | ||
| main | ||
| menuselect | ||
| pbx | ||
| phoneprov | ||
| res | ||
| rest-api | ||
| rest-api-templates | ||
| sounds | ||
| static-http | ||
| tests | ||
| third-party | ||
| utils | ||
| .cleancount | ||
| .gitignore | ||
| bootstrap.sh | ||
| BSDmakefile | ||
| BUGS | ||
| config.guess | ||
| config.sub | ||
| configure | ||
| configure.ac | ||
| COPYING | ||
| CREDITS | ||
| default.exports | ||
| install-sh | ||
| LICENSE | ||
| Makefile | ||
| Makefile.moddir_rules | ||
| Makefile.rules | ||
| makeopts.in | ||
| missing | ||
| mkinstalldirs | ||
| README-addons.txt | ||
| README-SERIOUSLY.bestpractices.md | ||
| README.md | ||
| sample.call | ||
| SECURITY.md | ||
| Zaptel-to-DAHDI.txt | ||
The Asterisk(R) Open Source PBX
By Mark Spencer <markster@digium.com> and the Asterisk.org developer community.
Copyright (C) 2001-2025 Sangoma Technologies Corporation and other copyright holders.
SECURITY
It is imperative that you read and fully understand the contents of the security information document before you attempt to configure and run an Asterisk server.
See Important Security Considerations for more information.
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. However, Asterisk supports more telephony interfaces than just Internet telephony. Asterisk also has a vast amount of support for traditional PSTN telephony, as well.
For more information on the project itself, please visit the Asterisk Home Page and the official Asterisk Documentation.
SUPPORTED 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, Apple's Mac OS X, and the BSD variants.
GETTING STARTED
Most users are using VoIP/SIP exclusively these days but if you need to interface to TDM or analog services or devices, be sure you've got supported hardware.
Supported telephony hardware includes:
- All Analog and Digital Interface cards from Sangoma
- Any full duplex sound card supported by PortAudio
- The Xorcom Astribank channel bank
UPGRADING FROM AN EARLIER VERSION
If you are updating from a previous version of Asterisk, make sure you read the Change Logs.
NEW INSTALLATIONS
Ensure that your system contains a compatible compiler and development libraries. Asterisk requires either the GNU Compiler Collection (GCC) version 4.1 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 ncurses.
There are many modules that have additional dependencies. To see what
libraries are being looked for, see ./configure --help, or run
make menuselect to view the dependencies for specific modules.
On many distributions, these dependencies are installed by packages with names
like 'glibc-devel', 'ncurses-devel', 'openssl-devel' and 'zlib-devel'
or similar. The contrib/scripts/install_prereq script can be used to install
the dependencies for most Debian and Redhat based Linux distributions.
The script also handles SUSE, Arch, Gentoo, FreeBSD, NetBSD and OpenBSD but
those distributions mightnoit have complete support or they might be out of date.
So, let's proceed:
-
Read the documentation.
The Asterisk Documentation website has full information for building, installing, configuring and running Asterisk. -
Run
./configure
Execute the configure script to guess values for system-dependent variables used during compilation. If the script indicates that some required components are missing, you can run./contrib/scripts/install_prereq installto install the necessary components. Note that this will install all dependencies for every functionality of Asterisk. After running the script, you will need to rerun./configure. -
Run
make menuselect
This is needed if you want to select the modules that will be compiled and to check dependencies for various optional modules. -
Run
make
Assuming the build completes successfully: -
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: -
Run
make samples
Doing so will overwrite any existing configuration files you have installed. -
Finally, you can launch Asterisk in the foreground mode (not a daemon) 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 typecore show helpat any time to get help with the system. For help with a specific command, typecore show help <command>.
man asterisk at the Unix/Linux command prompt will give you detailed
information on how to start and stop Asterisk, as well as all the command
line options for starting 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 statements in the form
variable = value although you may see variable => value in older samples.
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. For this reason, you should always use a time synchronization package to keep your system time accurate. All OS/distributions make one or more of the following packages available:
- ntpd/ntpsec
- chronyd
- systemd-timesyncd
Be sure to install and configure one (and only one) of them.
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 approximately 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
Visit the Asterisk Documentation website for more documentation on various features and please read all the configuration samples that include documentation on the configuration options.
Finally, you may wish to join the Asterisk Community Forums
Welcome to the growing worldwide community of Asterisk users!
Mark Spencer, and the Asterisk.org development community
Asterisk is a trademark of Sangoma Technologies Corporation
[Sangoma](https://www.sangoma.com/)[Home Page](https://www.asterisk.org)[Support](https://www.asterisk.org/support)[Documentation](https://docs.asterisk.org)[Community Forums](https://community.asterisk.org)[Release Notes](https://github.com/asterisk/asterisk/releases)[Security](https://docs.asterisk.org/Deployment/Important-Security-Considerations/)[Mailing List Archive](https://lists.digium.com)