The Asterisk(R) Open Source PBX

[optional]

* xmlSubstituteEntitiesDefault
+* xmlParseMemory
+
parameter name="context" documentationtype="dialplan_context"
+parameter name="extension" documentationtype="dialplan_extension"
+parameter name="priority" documentationtype="dialplan_priority" required="true"
+
[[context,]extension,]priority
+
        By Mark Spencer <markster@digium.com> and the Asterisk.org developer community.
-        Copyright (C) 2001-2021 Sangoma Technologies Corporation and other copyright holders.
+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.
+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 wiki.  In addition you'll find lots
-of information compiled by the Asterisk community at voip-info.org.
-There is a book on Asterisk published by O'Reilly under the Creative Commons
-License. It is available in book stores as well as in a downloadable version on
-the asteriskdocs.org web site.
+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
@@ -27,26 +24,22 @@ 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, Cygwin,
-and the BSD variants.
+operating systems as well, 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 sound card) to install and run Asterisk.
+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
-* QuickNet Internet PhoneJack and LineJack (http://www.quicknet.net)
-* any full duplex sound card supported by ALSA, OSS, or PortAudio
-* any ISDN card supported by mISDN on Linux
-* The Xorcom Astribank channel bank
-* VoiceTronix OpenLine products
+* 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 UPGRADE.txt file in the source directory. There are some files
-and configuration options that you will have to change, even though we
-made every effort possible to maintain backwards compatibility.
-In order to discover new features to use, please check the configuration
-examples in the configs directory of the source code distribution.  For a
-list of new features in this version of Asterisk, see the CHANGES file.
+read the Change Logs.
+
+Change Logs
+
+
 NEW INSTALLATIONS
 Ensure that your system contains a compatible compiler and development
 libraries.  Asterisk requires either the GNU Compiler Collection (GCC) version
@@ -58,108 +51,79 @@ 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.
-So, let's proceed:
-1. Read this file.
-There are more documents than this one in the doc directory.  You may also
-want to check the configuration files that contain examples and reference
-guides in the configs directory.
+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:
 
-Run ./configure
-
-Execute the configure script to guess values for system-dependent
-variables used during compilation.
-
-Run make menuselect [optional]
-
-This is needed if you want to select the modules that will be compiled and to
+

+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 install
+to 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
+
+
+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
+
+
+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 type "core show help" at any time to get help with the system.  For help
-with a specific command, type "core show help ".  To start the PBX using
-your sound card, you can type "console dial" to dial the PBX.  Then you can use
-"console answer", "console hangup", and "console 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).
-"man asterisk" at the Unix/Linux command prompt will give you detailed
+like this:

+*CLI>

+You can type core show help at any time to get help with the system.  For help
+with a specific command, type core 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.
-Feel free to look over the configuration files in /etc/asterisk, where you
-will 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
+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 chan_dahdi.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.
+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.  If your system cannot keep accurate time
-by itself use NTP 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.
+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,
@@ -189,14 +153,22 @@ these changes to take effect.
 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 on various features.
-Again, please read all the configuration samples that include documentation
-on the configuration options.
-Finally, you may wish to visit the support site and join the mailing
-list if you're interested in getting 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] 
+[Home Page] 
+[Support] 
+[Documentation] 
+[Community Forums] 
+[Release Notes] 
+[Security] 
+[Mailing List Archive]

diff --git a/README.md b/README.md
index 063db527e9..b9ab3011de 100644
--- a/README.md
+++ b/README.md
@@ -55,7 +55,7 @@ If you are updating from a previous version of Asterisk, make sure you
 read the Change Logs.
 
 
-[Change Logs](https://downloads.asterisk.org/pub/telephony/asterisk)
+[Change Logs](ChangeLogs/ChangeLog-certified-18.9-cert15.html)
 
 
 ### NEW INSTALLATIONS
diff --git a/contrib/realtime/mysql/mysql_config.sql b/contrib/realtime/mysql/mysql_config.sql
index 47ae56b656..61e19b1ac3 100644
--- a/contrib/realtime/mysql/mysql_config.sql
+++ b/contrib/realtime/mysql/mysql_config.sql
@@ -1358,3 +1358,15 @@ ALTER TABLE ps_endpoints ADD COLUMN tenantid VARCHAR(80);
 
 UPDATE alembic_version SET version_num='655054a68ad5' WHERE alembic_version.version_num = '9f3692b1654b';
 
+-- Running upgrade 655054a68ad5 -> 801b9fced8b7
+
+ALTER TABLE ps_subscription_persistence ADD COLUMN generator_data TEXT;
+
+UPDATE alembic_version SET version_num='801b9fced8b7' WHERE alembic_version.version_num = '655054a68ad5';
+
+-- Running upgrade 801b9fced8b7 -> 4f91fc18c979
+
+ALTER TABLE ps_endpoints ADD COLUMN suppress_moh_on_sendonly ENUM('0','1','off','on','false','true','no','yes');
+
+UPDATE alembic_version SET version_num='4f91fc18c979' WHERE alembic_version.version_num = '801b9fced8b7';
+
diff --git a/contrib/realtime/mysql/mysql_voicemail.sql b/contrib/realtime/mysql/mysql_voicemail.sql
index b30456e41c..88233f9fa9 100644
--- a/contrib/realtime/mysql/mysql_voicemail.sql
+++ b/contrib/realtime/mysql/mysql_voicemail.sql
@@ -33,3 +33,13 @@ ALTER TABLE voicemail_messages MODIFY recording BLOB(4294967295) NULL;
 
 UPDATE alembic_version SET version_num='39428242f7f5' WHERE alembic_version.version_num = 'a2e9769475e';
 
+-- Running upgrade 39428242f7f5 -> 1c55c341360f
+
+UPDATE alembic_version SET version_num='1c55c341360f' WHERE alembic_version.version_num = '39428242f7f5';
+
+-- Running upgrade 1c55c341360f -> 64fae6bbe7fb
+
+DROP INDEX voicemail_messages_dir ON voicemail_messages;
+
+UPDATE alembic_version SET version_num='64fae6bbe7fb' WHERE alembic_version.version_num = '1c55c341360f';
+
diff --git a/contrib/realtime/postgresql/postgresql_config.sql b/contrib/realtime/postgresql/postgresql_config.sql
index 15b8441d64..2357e06f61 100644
--- a/contrib/realtime/postgresql/postgresql_config.sql
+++ b/contrib/realtime/postgresql/postgresql_config.sql
@@ -1472,5 +1472,17 @@ ALTER TABLE ps_endpoints ADD COLUMN tenantid VARCHAR(80);
 
 UPDATE alembic_version SET version_num='655054a68ad5' WHERE alembic_version.version_num = '9f3692b1654b';
 
+-- Running upgrade 655054a68ad5 -> 801b9fced8b7
+
+ALTER TABLE ps_subscription_persistence ADD COLUMN generator_data TEXT;
+
+UPDATE alembic_version SET version_num='801b9fced8b7' WHERE alembic_version.version_num = '655054a68ad5';
+
+-- Running upgrade 801b9fced8b7 -> 4f91fc18c979
+
+ALTER TABLE ps_endpoints ADD COLUMN suppress_moh_on_sendonly ast_bool_values;
+
+UPDATE alembic_version SET version_num='4f91fc18c979' WHERE alembic_version.version_num = '801b9fced8b7';
+
 COMMIT;
 
diff --git a/contrib/realtime/postgresql/postgresql_voicemail.sql b/contrib/realtime/postgresql/postgresql_voicemail.sql
index 75caef9ec1..ada42ee881 100644
--- a/contrib/realtime/postgresql/postgresql_voicemail.sql
+++ b/contrib/realtime/postgresql/postgresql_voicemail.sql
@@ -35,5 +35,15 @@ ALTER TABLE voicemail_messages ALTER COLUMN recording TYPE BYTEA;
 
 UPDATE alembic_version SET version_num='39428242f7f5' WHERE alembic_version.version_num = 'a2e9769475e';
 
+-- Running upgrade 39428242f7f5 -> 1c55c341360f
+
+UPDATE alembic_version SET version_num='1c55c341360f' WHERE alembic_version.version_num = '39428242f7f5';
+
+-- Running upgrade 1c55c341360f -> 64fae6bbe7fb
+
+DROP INDEX voicemail_messages_dir;
+
+UPDATE alembic_version SET version_num='64fae6bbe7fb' WHERE alembic_version.version_num = '1c55c341360f';
+
 COMMIT;
 


Change Log for Release asterisk-certified-18.9-cert15

Links:

Summary:

User Notes:

res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"

app_mixmonitor: Add 'D' option for dual-channel audio.

Upgrade Notes:

Commit Authors:

Issue and Commit Detail:

Closed Issues:

Commits By Author:

Ben Ford (2):

George Joseph (12):

Joshua C. Colp (1):

Marcel Wagner (1):

Mike Bradeen (1):

Naveen Albert (1):

Sean Bright (6):

Shyju Kanaprath (1):

Commit List:

Commit Details:

res_pjsip: Backport pjsip uri utilities.

build_tools: Backport from 18

chan_sip.c: Fix __sip_reliable_xmit build error

chan_dahdi.c: Resolve a format-truncation build warning.

xml.c: Update deprecated libxml2 API usage.

documentation: Update Gosub, Goto, and add new documentationtype.

README.md: Updates and Fixes

README.md: Removed outdated link

general: Fix broken links.

documentation: Add information on running install_prereq script in readme

LICENSE: Update company name, email, and address.

README.md, asterisk.c: Update Copyright Dates

manager.c: Rename restrictedFile to is_restricted_file.

gcc14: Fix issues caught by gcc 14

res_pjsip: Change suppress_moh_on_sendonly to OPT_BOOL_T

res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"

res_pjsip: Move tenantid to end of ast_sip_endpoint

app_mixmonitor: Add 'D' option for dual-channel audio.

core_unreal.c: Fix memory leak in ast_unreal_new_channels()

geolocation.sample.conf: Fix comment marker at end of file

manager.c: Add unit test for Originate app and appdata permissions

alembic: Drop redundant voicemail_messages index.

app_voicemail: add NoOp alembic script to maintain sync

res_pjsip_pubsub: Persist subscription 'generator_data' in sorcery

Fix application references to Background

The Asterisk(R) Open Source PBX

SECURITY

WHAT IS ASTERISK ?

SUPPORTED OPERATING SYSTEMS

Linux

Others

GETTING STARTED

UPGRADING FROM AN EARLIER VERSION

NEW INSTALLATIONS

ABOUT CONFIGURATION FILES

SPECIAL NOTE ON TIME

FILE DESCRIPTORS

MORE INFORMATION