CHANGES

==============================================================================
===
=== This file documents the new and/or enhanced functionality added in
=== the Asterisk versions listed below. This file does NOT include
=== changes in behavior that would not be backwards compatible with
=== previous versions; for that information see the UPGRADE.txt file
=== and the other UPGRADE files for older releases.
===
==============================================================================

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13 to Asterisk 14 --------------------
------------------------------------------------------------------------------

Applications
------------------

BridgeAdd
------------------
 * A new application in Asterisk, this will join the calling channel
   to an existing bridge containing the named channel prefix.

ConfBridge
------------------
 * Added the ability to pass options to MixMonitor when recording is used with
   ConfBridge. This includes the addition of the following configuration
   parameters for the 'bridge' object:
   - record_file_timestamp: whether or not to append the start time to the
     recorded file name
   - record_options: the options to pass to the MixMonitor application
   - record_command: a command to execute when recording is finished
   Note that these options may also be with the CONFBRIDGE function.

SMS
------------------
 * Added the 'n' option, which prevents the SMS from being written to the log
   file. This is needed for those countries with privacy laws that require
   providers to not log SMS content.


CDRs
------------------
cdr_odbc
------------------
 * Added a new configuration option, "newcdrcolumns", which enables use of the
   post-1.8 CDR columns 'peeraccount', 'linkedid', and 'sequence'.

------------------
cdr_csv
------------------
 * Added a new configuration option, "newcdrcolumns", which enables use of the
   post-1.8 CDR columns 'peeraccount', 'linkedid', and 'sequence'.


Channel Drivers
------------------

chan_dahdi
------------------
 * The CALLERID(ani2) value for incoming calls is now populated in featdmf
   signaling mode.  The information was previously discarded.
 * Added the force_restart_unavailable_chans compatibility option.  When
   enabled it causes Asterisk to restart the ISDN B channel if an outgoing
   call receives cause 44 (Requested channel not available).

chan_iax2
------------------
 * The iax.conf forcejitterbuffer option has been removed.  It is now always
   forced if you set iax.conf jitterbuffer=yes.  If you put a jitter buffer
   on a channel it will be on the channel.
 * A new configuration parameters, 'calltokenexpiration', has been added that
   controls the duration before a call token expires. Default duration is 10
   seconds. Setting this to a higher value may help in lagged networks or those
   experiencing high packet loss.

chan_sip
------------------
 * New 'rtpbindaddr' global setting. This allows a user to define which
   ipaddress to bind the rtpengine to. For example, chan_sip might bind
   to eth0 (10.0.0.2) but rtpengine to eth1 (192.168.1.10).
 * DTLS related configuration options can now be set at a general level.
   Enabling DTLS support, though, requires enabling it at the user
   or peer level.

chan_pjsip
------------------
 * New 'user_eq_phone' endpoint setting. This adds a 'user=phone' parameter
   to the request URI and From URI if the user is determined to be a phone number.
 * New 'moh_passthrough' endpoint setting. This will pass hold and unhold requests
   through using SIP re-invites with sendonly and sendrecv accordingly.
 * Added the pjsip.conf system type disable_tcp_switch option.  The option
   allows the user to disable switching from UDP to TCP transports described
   by RFC 3261 section 18.1.1.
 * New 'line' and 'endpoint' options added on outbound registrations. This allows some
   identifying information to be added to the Contact of the outbound registration.
   If this information is present on messages received from the remote server
   the message will automatically be associated with the configured endpoint on the
   outbound registration.

Core
------------------
 * The core of Asterisk uses a message bus called "Stasis" to distribute
   information to internal components. For performance reasons, the message
   distribution was modified to make use of a thread pool instead of a
   dedicated thread per consumer in certain cases. The initial settings for
   the thread pool can now be configured in 'stasis.conf'.

 * A new core DNS API has been implemented which provides a common interface
   for DNS functionality. Modules that use this functionality will require that
   a DNS resolver module is loaded and available.

 * Modified processing of command-line options to first parse only what
   is necessary to read asterisk.conf. Once asterisk.conf is fully loaded,
   the remaining options are processed.  The -X option now applies to
   asterisk.conf only.  To enable #exec for other config files you must
   set execincludes=yes in asterisk.conf.  Any other option set on the
   command-line will now override the equivalent setting from asterisk.conf.

 * The TLS core in Asterisk now supports X.509 certificate subject alternative
   names. This way one X.509 certificate can be used for hosts that can be
   reached under multiple DNS names or for multiple hosts.

 * The Asterisk logging system now supports JSON structured logging. Log
   channels specified in logger.conf or added dynamically via CLI commands now
   support an optional specifier prior to their levels that determines their
   formatting. To set a log channel to format its entries as JSON, a formatter
   of '[json]' can be set, e.g.,
      full => [json]debug,verbose,notice,warning,error

Functions
------------------

CHANNEL
------------------
 * Added CHANNEL(onhold) item that returns 1 (onhold) and 0 (not-onhold) for
   the hold status of a channel.

DTMF Features
------------------
 * The transferdialattempts default value has been changed from 1 to 3. The
   transferinvalidsound has been changed from "pbx-invalid" to "privacy-incorrect".
   These were changed to make DTMF transfers be more user-friendly by default.


Resources
------------------

res_musiconhold
------------------
 * Added sort=randstart to the sort options. It sorts the files by name and
   then chooses the first file to play at random.
 * Added preferchannelclass=no option to prefer the application-passed class
   over the channel-set musicclass. This allows separate hold-music from
   application (e.g. Queue or Dial) specified music.

res_resolver_unbound
------------------
 * Added a res_resolver_unbound module which uses the libunbound resolver library
   to perform DNS resolution. This module requires the libunbound library to be
   installed in order to be used.

res_pjsip
------------------
 * A new SIP resolver using the core DNS API has been implemented. This relies on
   external SIP resolver support in PJSIP which is only available as of PJSIP
   2.4. If this support is unavailable the existing built-in PJSIP SIP resolver
   will be used instead. The new SIP resolver provides NAPTR support, improved
   SRV support, and AAAA record support.

res_pjsip_outbound_registration
-------------------------------
* A new 'fatal_retry_interval' option has been added to outbound registration.
  When set (default is zero), and upon receiving a failure response to an
  outbound registration, registration is retried at the given interval up to
  'max_retries'.

CEL Backends
------------------

cel_pgsql
------------------
 * Added a new option, 'usegmtime', which causes timestamps in CEL events
   to be logged in GMT.

 * Added support to set schema where located the table cel. This settings is
   configurable for cel_pgsql via the 'schema' in configuration file
   cel_pgsql.conf.

CDR Backends
------------------

cdr_adaptive_odbc
------------------
 * Added the ability to set the character to quote identifiers. This
   allows adding the character at the start and end of table and column
   names. This setting is configurable for cdr_adaptive_odbc via the
   quoted_identifiers in configuration file cdr_adaptive_odbc.conf.

Queue
-------------------
 * Added field ReasonPause on QueueMemberStatus if set when paused, the reason
   the queue member was paused.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.6.0 to Asterisk 13.7.0 ------------
------------------------------------------------------------------------------

Codecs
------------------
 * Added format attribute negotiation for the VP8 video codec. Format attribute
   negotiation is provided by the res_format_attr_vp8 module.

ConfBridge
------------------
 * A new "timeout" user profile option has been added. This configures the number
   of seconds that a participant may stay in the ConfBridge after joining. When
   the time expires, the user is ejected from the conference and CONFBRIDGE_RESULT
   is set to "TIMEOUT" on the channel.

chan_sip
------------------
 * The websockets_enabled option has been added to the general section of
   sip.conf.  The option is enabled by default to match the previous behavior.
   The option should be disabled when using res_pjsip_transport_websockets to
   ensure chan_sip will not conflict with PJSIP websockets.

Dialplan Functions
------------------
 * The HOLD_INTERCEPT dialplan function now actually exists in the source tree.
   While support for the events was added in Asterisk 13.4.0, the function
   accidentally never made it in. That function is now present, and will cause
   the 'hold' raised by a channel to be intercepted and converted into an
   event instead.

res_pjsip_outbound_registration
-------------------------------
 * If res_statsd is loaded and a StatsD server is configured, basic statistics
   regarding the state of outbound registrations will now be emitted. This
   includes:
   - A GAUGE statistic for the overall number of outbound registrations, i.e.:
       PJSIP.registrations.count
   - A GAUGE statistic for the overall number of outbound registrations in a
     particular state, e.g.:
       PJSIP.registrations.state.Registered

res_pjsip
------------------
 * The ability to use "like" has been added to the pjsip list and show
   CLI commands.  For instance: CLI> pjsip list endpoints like abc

 * If res_statsd is loaded and a StatsD server is configured, basic statistics
   regarding the state of PJSIP contacts will now be emitted. This includes:
   - A GAUGE statistic for the overall number of contacts in a particular
     state, e.g.:
       PJSIP.contacts.states.Reachable
   - A TIMER statistic for the RTT time for each qualified contact, e.g.:
       PJSIP.contacts.alice@@127.0.0.1:5061.rtt

func_callerid
-------------------
 * CALLERID(pres) is now documented as a valid alternative to setting both
   CALLERID(name-pres) and CALLERID(num-pres) at once.  Some channel drivers,
   like chan_sip, don't make a distinction between the two: they take the
   least public value from name-pres and num-pres.  By using CALLERID(pres)
   for reading and writing, you touch the same combined value in the dialplan.
   The same applies to CONNECTEDLINE(pres), REDIRECTING(orig-pres),
   REDIRECTING(to-pres) and REDIRECTING(from-pres).

res_endpoint_stats
-------------------
 * A new module that emits StatsD statistics regarding Asterisk endpoints.
   This includes a total count of the number of endpoints, the count of the
   number of endpoints in the technology agnostic state of the endpoint -
   online or offline - as well as the number of channels associated with each
   endpoint. These are recorded as three different GAUGE statistics:
    - endpoints.count
    - endpoints.state.{unknown|offline|online}
    - endpoints.{tech}.{resource}.channels


------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.5.0 to Asterisk 13.6.0 ------------
------------------------------------------------------------------------------

Dialplan Functions
------------------
 * The CHANNEL function, when used on a PJSIP channel, now exposes a 'call-id'
   extraction option when using with the 'pjsip' signalling option. It will
   return the SIP Call-ID associated with the INVITE request that established
   the PJSIP channel.

ARI
------------------
 * Two new endpoint related events are now available: PeerStatusChange and
   ContactStatusChange. In particular, these events are useful when subscribing
   to all event sources, as they provide additional endpoint related
   information beyond the addition/removal of channels from an endpoint.

 * Added the ability to subscribe to all ARI events in Asterisk, regardless
   of whether the application 'controls' the resource. This is useful for
   scenarios where an ARI application merely wants to observe the system,
   as opposed to control it. There are two ways to accomplish this:
   (1) Via the WebSocket connection URI. A new query paramter, 'subscribeAll',
       has been added that, when present and True, will subscribe all
       specified applications to all ARI event sources in Asterisk.
   (2) Via the applications resource. An ARI client can, at any time, subscribe
       to all resources in an event source merely by not providing an explicit
       resource. For example, subscribing to an event source of 'channels:'
       as opposed to 'channels:12345' will subscribe the application to all
       channels.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.4.0 to Asterisk 13.5.0 ------------
------------------------------------------------------------------------------

AMI
------------------
 * A new ContactStatus event has been added that reflects res_pjsip contact
   lifecycle changes:  Created, Removed, Reachable, Unreachable, Unknown.

 * Added the Linkedid header to the common channel headers listed for each
   channel in AMI events.

ARI
------------------
 * A new feature has been added that enables the retrieval of modules and
   module information through an HTTP request. Information on a single module
   can be also be retrieved. Individual modules can be loaded to Asterisk, as
   well as unloaded and reloaded.

* A new resource has been added to the 'asterisk' resource, 'config/dynamic'.
   This resource allows for push configuration of sorcery derived objects
   within Asterisk. The resource supports creation, retrieval, updating, and
   deletion. Sorcery derived objects that are manipulated by this resource
   must have a sorcery wizard that supports the desired operations.

 * A new feature has been added that allows for the rotation of log channels
   through HTTP requests.


res_pjsip
------------------
* A new 'g726_non_standard' endpoint option has been added that, when set to
  'yes' and g.726 audio is negotiated, forces the codec to be treated as if it
  is AAL2 packed on the channel.

* A new 'rtp_keepalive' endpoint option has been added. This option specifies
  an interval, in seconds, at which we will send RTP comfort noise packets to
  the endpoint. This functions identically to chan_sip's "rtpkeepalive" option.

* New 'rtp_timeout' and 'rtp_timeout_hold' endpoint options have been added.
  These options specify the amount of time, in seconds, that Asterisk will wait
  before terminating the call due to lack of received RTP. These are identical
  to chan_sip's rtptimeout and rtpholdtimeout options.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.3.0 to Asterisk 13.4.0 ------------
------------------------------------------------------------------------------

chan_pjsip
------------------
 * New 'rpid_immediate' option to control if connected line update information
   goes to the caller immediately or waits for another reason to send the
   connected line information update.  See the online option documentation for
   more information.  Defaults to 'no' as setting it to 'yes' can result in
   many unnecessary messages being sent to the caller.

 * The configuration setting 'progressinband' now defaults to 'no', which
   matches the actual behavior of previous versions.

res_pjsip
------------------
 * A new CLI command has been added: "pjsip show settings", which shows
   both the global and system configuration settings.

 * A new aor option has been added: "qualify_timeout", which sets the timeout
   in seconds for a qualify.  The default is 3 seconds.  This overrides the
   hard coded 32 seconds in pjproject.

 * Endpoint status will now change to "Unreachable" when all contacts are
   unavailable.  When any contact becomes available, the endpoint will status
   will change back to "Reachable".

 * A new global option has been added: "max_initial_qualify_time", which
   sets the maximum amount of time from startup that qualifies should be
   attempted on all contacts.

res_ari_channels
------------------
 * Two new events, 'ChannelHold' and 'ChannelUnhold', have been added to the
   events data model. These events are raised when a channel indicates a hold
   or unhold, respectively.

func_holdintercept
------------------
 * A new dialplan function, HOLD_INTERCEPT, has been added. This function, when
   placed on a channel, intercepts hold/unhold indications signalled by the
   channel and prevents them from moving on to other channels in a bridge with
   the hold initiator. Instead, AMI or ARI events are raised indicating that
   the channel wanted to place someone on hold. This allows external
   applications to implement their own custom hold/unhold logic.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.2.0 to Asterisk 13.3.0 ------------
------------------------------------------------------------------------------

chan_pjsip/app_transfer
------------------
 * The Transfer application, when used with chan_pjsip, now supports using
   a PJSIP endpoint as the transfer destination. This is in addition to
   explicitly specifying a SIP URI to transfer to.

res_ari_channels
------------------
 * The ARI /channels resource now supports a new operation, 'redirect'. The
   redirect operation will perform a technology and state specific redirection
   on the channel to a specified endpoint or destination. In the case of SIP
   technologies, this is either a 302 Redirect response to an on-going INVITE
   dialog or a SIP REFER request.

res_pjsip
------------------
 * A new 'endpoint_identifier_order' option has been added that allows one to
   set the order by which endpoint identifiers are processed and checked. This
   option is specified under the 'global' type configuration section.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.1.0 to Asterisk 13.2.0 ------------
------------------------------------------------------------------------------

 * New 'PJSIP_AOR' and 'PJSIP_CONTACT' dialplan functions have been added which
   allow examining PJSIP AORs or contacts from the dialplan.

res_pjsip_outbound_registration
------------------
 * The 'pjsip send unregister' command now stops further registrations.

 * A new command 'pjsip send register' has been added which allows you to
   start or restart periodic registration.  It can be used after a
   'send unregister' or after a 401 permanent error.

res_pjsip_config_wizard
------------------
 * This is a new module that adds streamlined configuration capability for
   chan_pjsip.  It's targeted at users who have lots of basic configuration
   scenarios like 'phone' or 'agent' or 'trunk'.  Additional information
   can be found in the sample configuration file at
   config/samples/pjsip_wizard.conf.sample.

res_fax
-----------
 * The T.38 negotiation timeout was previously hard coded at 5000 milliseconds
   and is now configurable via the 't38timeout' configuration option in
   res_fax.conf and via the fax options dialplan function 'FAXOPT(t38timeout)'.
   The default remains at 5000 milliseconds.

PJSIP Transports
----------
 * The ca_list_path transport parameter has been added for TLS transports. This
   option behaves similarly to the old sip.conf option "tlscapath". In order to
   use this, you must be using PJProject version 2.4 or higher.

ARI
------------------
 * The Originate operation now takes in an originator channel. The linked ID of
   this originator channel is applied to the newly originated outgoing channel.
   If using CEL this allows an association to be established between the two so
   it can be recognized that the originator is dialing the originated channel.

 * "language" (the default spoken language for the channel) is now included in
   the standard channel state output for suitable events.

 * The POST channels/{id} operation and the POST channels/{id}/continue operation
   now have a new "label" parameter. This allows for origination or continuation
   to a labeled priority in the dialplan instead of requiring a specific priority
   number. The ARI version has been bumped to 1.7.0 as a result.

AMI
------------------
 * "Language" (the default spoken language for the channel) is now included in
   the standard channel state output for suitable events.

 * AMI actions that return a list of events have been made to return consistent
   headers for the action response event starting the list and the list complete
   event.  The AMI version has been bumped to 2.7.0 as a result.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.0.0 to Asterisk 13.1.0 ------------
------------------------------------------------------------------------------

AMI
------------------
 * Event NewConnectedLine is emitted when the connected line information on
   a channel changes.

ARI
------------------
 * Event ChannelConnectedLine is emitted when the connected line information
   on a channel changes.

Core Transfers
-----------------

The features.conf general section has three new configurable options:
    * transferdialattempts
    * transferretrysound
    * transferinvalidsound
For more information on what these options do, see the Asterisk wiki:
 https://wiki.asterisk.org/wiki/x/W4fAAQ

Channel Drivers
------------------

chan_pjsip
------------------
 * New 'media_encryption_optimistic' endpoint setting. This will use SRTP
   when possible but does not consider lack of it a failure.

res_pjsip_endpoint_identifer_ip
------------------
 * New CLI commands have been added: "pjsip show identif(y|ies)", which lists
   all configured PJSIP identify objects
   
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 12 to Asterisk 13 --------------------
------------------------------------------------------------------------------

Overview
------------------

Asterisk 13 is the next Long Term Support (LTS) release of Asterisk. As such,
the focus of development for this release of Asterisk was on improving the
usability and features developed in the previous Standard release, Asterisk 12.
Beyond a general refinement of end user features, development focussed heavily
on the Asterisk APIs - the Asterisk Manager Interface (AMI) and the Asterisk
REST Interface (ARI) - and the PJSIP stack in Asterisk. Some highlights of the
new features include:

* Asterisk security events are now provided via AMI, allowing end users to
  monitor their Asterisk system in real time for security related issues.
* External control of Message Waiting Indicators (MWI) through both AMI and ARI.
* Reception/transmission of out of call text messages using any supported
  channel driver/protocol stack through ARI.
* Resource List Server support in the PJSIP stack, providing subscriptions to
  lists of resources and batched delivery of NOTIFY requests.
* Inter-Asterisk distributed device state and mailbox state using the PJSIP
  stack.

It is important to note that Asterisk 13 is built on the architecture developed
during the previous Standard release, Asterisk 12. Users upgrading to
Asterisk 13 should read about the new features in Asterisk 12 later in this file
(see Functionality changes from Asterisk 11 to Asterisk 12), as well as the
UPGRADE-12.txt delivered with this release. In particular, users upgrading to
Asterisk 13 from a release prior to Asterisk 12 should read the specifications
on AMI, CDRs, and CEL on the Asterisk wiki:
 * AMI - https://wiki.asterisk.org/wiki/x/dAFRAQ
 * CEL - https://wiki.asterisk.org/wiki/x/4ICLAQ
 * CDRs - https://wiki.asterisk.org/wiki/x/pwpRAQ

Many new featuers in Asterisk 13 were introduced in point releases of
Asterisk 12. Following this section - which documents the changes from all
versions of Asterisk 12 to Asterisk 13 - users should examine the new features
that were introduced in the point releases of Asterisk 12, as they are also
included in Asterisk 13.

Finally, all users upgrading to Asterisk 13 should read the UPGRADE.txt file
delivered with this release.


Build System
------------------
 * Sample config files have been moved from configs/ to a sub-folder of that
   directory, samples.

 * The menuselect utility has been pulled into the Asterisk repository. As a
   result, the libxml2 development library is now a required dependency for
   Asterisk.

 * A new Compiler Flag, REF_DEBUG, has been added. When enabled, reference
   counted objects will emit additional debug information to the refs log file
   located in the standard Asterisk log file directory. This log file is useful
   in tracking down object leaks and other reference counting issues. Prior to
   this version, this option was only available by modifying the source code
   directly. This change also includes a new script, refcounter.py, in the
   contrib folder that will process the refs log file. Note that this replaces
   the refcounter utility that could be built from the utils directory.


Applications
------------------

DahdiBarge
------------------
 * This module was deprecated and has been removed. Users of app_dahdibarge
   should use ChanSpy instead.

MixMonitor
------------------
 * New options to play a beep when starting a recording and stopping a recording
   have been added.  The option "p" will play a beep to the channel that starts
   the recording.  The option "P" will play a beep to the channel that stops the
   recording.

Queue
------------------
 * Queue rules can now be stored in a database table, queue_rules. Unlike other
   RealTime tables, the queue_rules table is only examined on module load or
   module reload. A new general setting has been added to queuerules.conf,
   'realtime_rules', which, when set to 'yes', will cause app_queue to look in
   RealTime for additional queue rules to parse. Note that both the file and
   the database can be used as a provide of queue rules when 'realtime_rules'
   is set to 'yes'.

   When app_queue is reloaded, all rules are re-parsed and loaded into memory.
   There is no caching of RealTime queue rules.

ReadFile
------------------
 * This module was deprecated and has been removed. Users of app_readfile
   should use func_env's FILE function instead.

Say
------------------
 * The 'say' family of dialplan applications now support the Japanese
   language. The 'language' parameter in say.conf now recognizes a setting of
   'ja', which will enable Japanese language specific mechanisms for playing
   back numbers, dates, and other items.

SayCountPL
------------------
 * This module was deprecated and has been removed. Users of app_saycountpl
   should use the Say family of applications.

SetMusicOnHold
------------------
 * The SetMusicOnHold dialplan application was deprecated and has been removed.
   Users of the application should use the CHANNEL function's musicclass
   setting instead.

WaitMusicOnHold
------------------
 * The WaitMusicOnHold dialplan application was deprecated and has been
   removed. Users of the application should use MusicOnHold with a duration
   parameter instead.

VoiceMail
------------------
 * VoiceMail and VoiceMailMain now support the Japanese language. The
   'language' parameter in voicemail.conf now recognizes a setting of 'ja',
   which will enable prompts to be played back using a Japanese grammatical
   structure. Additional prompts are necessary for this functionality,
   including:
   - jb-arimasu: there is
   - jb-arimasen: there is not
   - jb-oshitekudasai: please press
   - jb-ni: article ni
   - jb-ga: article ga
   - jb-wa: article wa
   - jb-wo: article wo

 * Add the ability to specify multiple email addresses in configuration,
   separated by a |.


CDR Backends
------------------

cdr_sqlite
-----------------
 * This module was deprecated and has been removed. Users of cdr_sqlite
   should use cdr_sqlite3_custom.

cdr_pgsql
------------------
 * Added the ability to support PostgreSQL application_name on connections.
   This allows PostgreSQL to display the configured name in the
   pg_stat_activity view and CSV log entries. This setting is configurable
   for cdr_pgsql via the appname configuration setting in cdr_pgsql.conf.


CEL Backends
------------------

cel_pgsql
------------------
 * Added the ability to support PostgreSQL application_name on connections.
   This allows PostgreSQL to display the configured name in the
   pg_stat_activity view and CSV log entries. This setting is configurable
   for cel_pgsql via the appname configuration setting in cel_pgsql.conf.


Channel Drivers
------------------

chan_dahdi
------------------
 * SS7 support now requires libss7 v2.0 or later.

 * Added SS7 support for connected line and redirecting.

 * Most SS7 CLI commands are reworked as well as new SS7 commands added.
   See online CLI help.

 * Added several SS7 config option parameters described in
   chan_dahdi.conf.sample.

chan_gtalk
------------------
 * This module was deprecated and has been removed. Users of chan_gtalk
   should use chan_motif.

chan_h323
------------------
 * This module was deprecated and has been removed. Users of chan_h323
   should use chan_ooh323.

chan_jingle
------------------
 * This module was deprecated and has been removed. Users of chan_jingle
   should use chan_motif.

chan_pjsip
------------------
 * Added the CLI command 'pjsip list ciphers' so a user can know what
   OpenSSL names are available on their system for the pjsip.conf cipher
   option.

chan_sip
------------------
 * The SIPPEER dialplan function no longer supports using a colon as a
   delimiter for parameters. The parameters for the function should be
   delimited using a comma.

 * The SIPCHANINFO dialplan function was deprecated and has been removed. Users
   of the function should use the CHANNEL function instead.


Core
------------------

Account Codes
------------------
 * Added functional peeraccount support.  Except for Queue, the
   accountcode propagation is now consistently propagated to outgoing
   channels before dialing.  The channel accountcode can change from its
   original non-empty value on channel creation for the following specific
   reasons.  One, dialplan sets it using CHANNEL(accountcode).  Two, an
   originate method that can specify an accountcode value.  Three, the
   calling channel propagates its peeraccount or accountcode to the
   outgoing channel's accountcode before dialing.  The change has two
   visible effects.  One, local channels now cross accountcode and
   peeraccount across the special bridge between the ;1 and ;2 channels
   just like channels between normal bridges.  Two, the
   CHANNEL(peeraccount) value can now be set before Dial and FollowMe to
   set the accountcode on the outgoing channel(s).

   For Queue, an outgoing channel's non-empty accountcode will not change
   unless explicitly set by CHANNEL(accountcode).  The change has three
   visible effects.  One, local channels now cross accountcode and
   peeraccount across the special bridge between the ;1 and ;2 channels
   just like channels between normal bridges.  Two, the queue member will
   get an accountcode if it doesn't have one and one is available from the
   calling channel's peeraccount.  Three, accountcode propagation includes
   local channel members where the accountcodes are propagated early
   enough to be available on the ;2 channel.

AMI
------------------
 * New DeviceStateChanged and PresenceStateChanged AMI events have been added.
   These events are emitted whenever a device state or presence state change
   occurs. The events are controlled by res_manager_device_state.so and
   res_manager_presence_state.so. If the high frequency of these events is
   problematic for you, do not load these modules.

 * Added DialplanExtensionAdd and DialplanExtensionRemove AMI commands. They
   work in basically the same way as the 'dialplan add extension' and
   'dialplan remove extension' CLI commands respectively.

 * New AMI action LoggerRotate reloads and rotates logger in the same manner
   as CLI command 'logger rotate'

 * New AMI Actions FAXSessions, FAXSession, and FAXStats replicate the
   functionality of CLI commands 'fax show sessions', 'fax show session',
   and fax show stats' respectively.

 * New AMI actions PRIDebugSet, PRIDebugFileSet, and PRIDebugFileUnset
   enable manager control over PRI debugging levels and file output.

 * AMI action PJSIPNotify may now send to a URI instead of only to a PJSIP
   endpoint as long as a default outbound endpoint is set. This also applies
   to the equivalent CLI command (pjsip send notify)

 * The AMI action PJSIPShowEndpoint now includes ContactStatusDetail sections
   that give information on Asterisk's attempts to qualify the endpoint.

 * The DialEnd event will now contain a Forward header if the dial is ending
   due to the call being forwarded. The contents of the Forward header is the
   extension in the number to which the call is being forwarded.

CEL
------------------
 * The "bridge_technology" extra field key has been added to BRIDGE_ENTER
   and BRIDGE_EXIT events.

Features
------------------
 * Channel variables are now substituted in arguments passed to applications
   run by using dynamic features.

TLS
------------------
 * The TLS core in Asterisk now supports Perfect Forward Secrecy (PFS).
   Enabling PFS is attempted by default, and is dependent on the configuration
   of the module using TLS.
   - Ephemeral ECDH (ECDHE) is enabled by default. To disable it, do not
     specify a ECDHE cipher suite in sip.conf, for example:
       tlscipher=AES128-SHA:DES-CBC3-SHA
   - Ephemeral DH (DHE) is disabled by default. To enable it, add DH parameters
     into the private key file, e.g., sip.conf tlsprivatekey. For example, the
     default dh2048.pem - see
     http://www.opensource.apple.com/source/OpenSSL098/OpenSSL098-35.1/src/apps/dh2048.pem?txt
   - Because clients expect the server to prefer PFS, and because OpenSSL sorts
     its cipher suites by bit strength, see "openssl ciphers -v DEFAULT".
     Consider re-ordering your cipher suites in the respective configuration
     file. For example:
       tlscipher=AES128+kEECDH:AES128+kEDH:3DES+kEDH:AES128-SHA:DES-CBC3-SHA:-ADH:-AECDH
     will use PFS when offered by the client. Clients which do not offer PFS
     fall-back to AES-128 (or even 3DES, as recommended by RFC 3261).


Functions
------------------

JACK_HOOK
------------------
 * The JACK_HOOK function now supports audio with a sample rate higher than
   8kHz.


Resources
------------------

res_config_pgsql
------------------
 * Added the ability to support PostgreSQL application_name on connections.
   This allows PostgreSQL to display the configured name in the
   pg_stat_activity view and CSV log entries. This setting is configurable
   for res_config_pgsql via the dbappname configuration setting in
   res_pgsql.conf.

res_pjsip_outbound_publish
------------------
 * A new module, res_pjsip_outbound_publish provides the mechanisms for sending
   PUBLISH requests for specific event packages to another SIP User Agent.

res_pjsip_pubsub
------------------
 * The publish/subscribe core module has been updated to support RFC 4662
   Resource Lists, allowing Asterisk to act as a Resource List Server (RLS).
   Resource lists are configured in pjsip.conf under a new object type,
   resource_list. Resource lists can contain either message-summary or presence
   events, and can be composed of specific resources that provide the event or
   other resource lists.

 * Inbound publication support is provided by a new object, inbound-publication.
   This configures res_pjsip_pubsub to accept PUBLISH requests from a particular
   resource. Which events are accepted is constructed dynamically; see
   res_pjsip_publish_asterisk for more information.

res_pjsip_publish_asterisk
------------------
 * A new module, res_pjsip_publish_asterisk adds support for PUBLISH requests of
   Asterisk information to other Asterisk servers. This module is intended only
   for Asterisk to Asterisk exchanges of information. Currently, this includes
   both mailbox state and device state information.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 12.4.0 to Asterisk 12.5.0 ------------
------------------------------------------------------------------------------

ARI
------------------
 * Stored recordings now support a new operation, copy. This will take an
   existing stored recording and copy it to a new location in the recordings
   directory.

 * LiveRecording objects now have three additional fields that can be reported
   in a RecordingFinished ARI event:
   - total_duration: the duration of the recording
   - talking_duration: optional. The duration of talking detected in the
     recording. This is only available if max_silence_seconds was specified
     when the recording was started.
   - silence_duration: optional. The duration of silence detected in the
     recording. This is only available if max_silence_seconds was specified
     when the recording was started.
   Note that all duration values are reported in seconds.

 * Users of ARI can now send and receive out of call text messages. Messages
   can be sent directly to a particular endpoint, or can be sent to the
   endpoints resource directly and inferred from the URI scheme. Text
   messages are passed to ARI clients as TextMessageReceived events. ARI
   clients can choose to receive text messages by subscribing to the particular
   endpoint technology or endpoints that they are interested in.

 * The applications resource now supports subscriptions to all endpoints of
   a particular channel technology. For example, subscribing to an eventSource
   of 'endpoint:PJSIP' will subscribe to all PJSIP endpoints.

res_pjsip
------------------
 * The endpoint configuration object now supports 'accountcode'. Any channel
   created for an endpoint with this setting will have its accountcode set
   to the specified value.

res_hep_rtcp
------------------
 * A new module, res_hep_rtcp, has been added that will forward RTCP call
   statistics to a HEP capture server. See res_hep for more information.

Functions
------------------
 * Function AUDIOHOOK_INHERIT has been deprecated. Audiohooks are now
   unconditionally inhereted through masquerades. As a side benefit, more
   than one audiohook of a given type may persist through a masquerade now.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 12.3.0 to Asterisk 12.4.0 ------------
------------------------------------------------------------------------------

AgentRequest
------------------
 * Returns new AGENT_STATUS value "NOT_CONNECTED" if the agent fails to
   connect with an incoming caller after being alerted to the presence
   of the incoming caller.  The most likely reason this would happen is
   the agent did not acknowledge the call in time.

AMI
------------------
 * New events have been added for the TALK_DETECT function. When the function
   is used on a channel, ChannelTalkingStart/ChannelTalkingStop events will be
   emitted to connected AMI clients indicating the start/stop of talking on
   the channel.

ARI
------------------
 * New event models have been aded for the TALK_DETECT function. When the
   function is used on a channel, ChannelTalkingStarted/ChannelTalkingFinished
   events will be emitted to connected WebSockets subscribed to the channel,
   indicating the start/stop of talking on the channel.

Functions
------------------
 * A new function, TALK_DETECT, has been added. When set on a channel, this
   fucntion causes events indicating the starting/stoping of talking on said
   channel to be emitted to both AMI and ARI clients.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 12.2.0 to Asterisk 12.3.0 ------------
------------------------------------------------------------------------------

ARI
------------------
 * A new Playback URI 'tone' has been added. Tones are specified either as
   an indication name (e.g. 'tone:busy') from indications.conf or as a tone
   pattern (e.g. 'tone:240/250,0/250'). Tones differ from normal playback
   URIs in that they must be stopped manually and will continue to occupy
   a channel's ARI control queue until they are stopped. They also can not
   be rewound or fastforwarded.

 * User events can now be generated from ARI.  Events can be signalled with
   arbitrary json variables, and include one or more of channel, bridge, or
   endpoint snapshots.  An application must be specified which will receive
   the event message (other applications can subscribe to it).  The message
   will also be delivered via AMI provided a channel is attached.  Dialplan
   generated user event messages are still transmitted via the channel, and
   will only be received by a stasis application they are attached to or if
   the channel is subscribed to.

chan_sip
-----------
 * SIP peers can now specify 'trust_id_outbound' which affects RPID/PAI
   fields for prohibited callingpres information. Values are legacy, no, and
   yes. By default, legacy is used.
   trust_id_outbound=legacy - behavior remains the same as 1.8.26.1. When
     dealing with prohibited callingpres and sendrpid=pai/rpid, RPID/PAI
     headers are appended to outbound SIP messages just as they are with
     allowed callingpres values, but data about the remote party's identity is
     anonymized.
     When sendrpid=rpid, only the remote party's domain is anonymized.
   trust_id_outbound=no - when dealing with prohibited callingpres, RPID/PAI
     headers are not sent.
   trust_id_outbound=yes - RPID/PAI headers are applied with the full remote
     party information in tact even for prohibited callingpres information.
     In the case of PAI, a Privacy: id header will be appended for prohibited
     calling information to communicate that the private information should
     not be relayed to untrusted parties.

res_parking
------------------
 * Manager action 'Park' now takes an additional argument 'AnnounceChannel'
   which can be used to announce the parked call's location to an arbitrary
   channel in a bridge. If 'Channel' and 'TimeoutChannel' are now the two
   parties in a one to one bridge, 'TimeoutChannel' is treated as having
   parked 'Channel' like with the Park Call DTMF feature and will receive
   announcements prior to being hung up.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 12.1.0 to Asterisk 12.2.0 ------------
------------------------------------------------------------------------------

Record
------------------
 * Record application now has an option 'o' which allows 0 to act as an exit
   key setting the RECORD_STATUS variable to 'OPERATOR' instead of 'DTMF'

ChanSpy
--------------------------
 * ChanSpy now accepts a channel uniqueid or a fully specified channel name
   as the chanprefix parameter if the 'u' option is specified.

ConfBridge
--------------------------
 * CONFBRIDGE dialplan function is now capable of creating/modifying dynamic
   conference user menus.

 * CONFBRIDGE dialplan function is now capable of removing dynamic conference
   menus, bridge settings, and user settings that have been applied by the
   CONFBRIDGE dialplan function.

 * The ConfBridge dialplan application now sets a channel variable,
   CONFBRIGE_RESULT, upon exiting. This variable can be used to determine
   how a channel exited the conference.

 * Added conference user option 'announce_join_leave_review'. This option
   implies 'announce_join_leave' with the added effect that the user will
   be asked if they want to confirm or re-record the recording of their
   name when entering the conference

Directory
--------------------------
 * At exit, the Directory application now sets a channel variable
   DIRECTORY_RESULT to one of the following based on the reason for exiting:
     OPERATOR    user requested operator by pressing '0' for operator
     ASSISTANT   user requested assistant by pressing '*' for assistant
     TIMEOUT     user pressed nothing and Directory stopped waiting
     HANGUP      user's channel hung up
     SELECTED    user selected a user from the directory and is routed
     USEREXIT    user pressed '#' from the selection prompt to exit
     FAILED      directory failed in a way that wasn't accounted for. Dang.

Monitor
------------------
 * Monitor() - A new option, B(), has been added that will turn on a periodic
   beep while the call is being recorded.

MusicOnHold
--------------------------
 * MusicOnHold streams (all modes other than "files") now support wide band
   audio too.

Page
--------------------------
 * Added options 'b' and 'B' to apply predial handlers for outgoing calls
   and for the channel executing Page respectively.

PickupChan
--------------------------
 * PickupChan now accepts channel uniqueids of channels to pickup.

Say
--------------------------
 * If a channel variable SAY_DTMF_INTERRUPT is present on a channel and set
   to 'true' (case insensitive), then any Say application (SayNumber,
   SayDigits, SayAlpha, SayAlphaCase, SayUnixTime, and SayCounted) will
   anticipate DTMF. If DTMF is received, these applications will behave like
   the background application and jump to the received extension once a match
   is established or after a short period of inactivity.

MixMonitor
-------------------------
 * A new function, MIXMONITOR, has been added to allow access to individual
   instances of MixMonitor on a channel.

 * A new option, B(), has been added that will turn on a periodic beep while the
   call is being recorded.


Channel Drivers
-------------------------

chan_sip
-------------------------
 * TEL URI support for inbound INVITE requests has been added. chan_sip will
   now handle TEL schemes in the Request and From URIs. The phone-context in
   the Request URI will be stored in the SIPURIPHONECONTEXT channel variable on
   the inbound channel.

Core
------------------
 * Exposed sorcery-based configuration files like pjsip.conf to dialplans via
   the new AST_SORCERY diaplan function.

 * Core Show Locks output now includes Thread/LWP ID if the platform
   supports this feature.

 * New "logger add channel" and "logger remove channel" CLI commands have
   been added to allow creation and deletion of dynamic logger channels
   without configuration changes. These dynamic logger channels will only
   exist until the next restart of asterisk.

ARI
------------------
 * The live recording object on recording events now contains a target_uri
   field which contains the URI of what is being recorded.

 * The bridge type used when creating a bridge is now a comma separated list of
   bridge properties. Valid options are: mixing, holding, dtmf_events, and
   proxy_media.

 * A channelId can now be provided when creating a channel, either in the
   uri (POST channels/my-channel-id) or as query parameter.  A local channel
   will suffix the second channel id with ';2' unless provided as query
   parameter otherChannelId.

 * A bridgeId can now be provided when creating a bridge, either in the uri
   (POST bridges/my-bridge-id) or as a query parameter.

 * A playbackId can be provided when starting a playback, either in the uri
   (POST channels/my-channel-id/play/my-playback-id /
    POST bridges/my-bridge-id/play/my-playback-id)  or as a query parameter.

 * A snoop channel can be started with a snoopId, in the uri or query.

AMI
------------------
 * Originate now takes optional parameters ChannelId and OtherChannelId,
   used to set the UniqueId on creation.  The other id is assigned to the
   second channel when dialing LOCAL, or defaults to appending ;2 if only
   the single Id is given.

 * The Mixmonitor action now has a "Command" header that can be used to
   indicate a post-process command to run once recording finishes.

RealTime
------------------
 * A new set of Alembic scripts has been added for CDR tables. This will create
   a 'cdr' table with the default schema that Asterisk expects.


Functions
------------------
 * A new function was added: PERIODIC_HOOK.  This allows running a periodic
   dialplan hook on a channel.  Any audio generated by this hook will be
   injected into the call.


Resources
------------------

res_hep
------------------
 * A new module, res_hep, has been added, that acts as a generic packet
   capture agent for the Homer Encapsulation Protocol (HEP) version 3.
   It can be configured via hep.conf. Other modules can use res_hep to send
   message traffic to a HEP capture server.

res_hep_pjsip
------------------
 * A new module, res_hep_pjsip, has been added that will forward PJSIP
   message traffic to a HEP capture server. See res_hep for more
   information.

res_pjsip
------------------
 * transport and endpoint ToS options (tos, tos_audio, and tos_video) may now
   be set as the named set of ToS values (cs0-cs7, af11-af43, ef).

 * Added the following new CLI commands:
   - "pjsip show contacts" - list all current PJSIP contacts.
   - "pjsip show contact" - show specific information about a current PJSIP
     contact.
   - "pjsip show channel" - show detailed information about a PJSIP channel.

res_pjsip_multihomed
------------------
 * A new module, res_pjsip_multihomed handles situations where the system
   Asterisk is running out has multiple interfaces. res_pjsip_multihomed
   determines which interface should be used during message sending.

res_pjsip_pidf_digium_body_supplement
------------------
 * A new module, res_pjsip_pidf_digium_body_supplement provides NOTIFY
   request body formatting for presence support in Digium phones.

res_pjsip_send_to_voicemail
------------------
 * A new module, res_pjsip_send_to_voicemail allows for REFER requests with
   particular headers to transfer a PJSIP channel directly to a particular
   extension that has VoiceMail. This is intended to be used with Digium
   phones that support this feature.

res_pjsip_outbound_registration
------------------
 * A new CLI command has been added: "pjsip show registrations", which lists
   all configured PJSIP registrations


------------------------------------------------------------------------------
--- Functionality changes from Asterisk 12.0.0 to Asterisk 12.1.0 ------------
------------------------------------------------------------------------------

AMI
------------------
 * Added a new module that provides AMI control over MWI within Asterisk,
   res_mwi_external_ami. Note that this module depends on res_mwi_external;
   for more information on enabling this module, see res_mwi_external.
   This module provides the MWIGet/MWIUpdate/MWIDelete actions, as well as
   the MWIGet/MWIGetComplete events.

 * The DialStatus field in the DialEnd event can now contain additional
   statuses that convey how the dial operation terminated. This includes
   ABORT, CONTINUE, and GOTO.

 * AMI will now emit security events. A new class authorization has been
   added in manager.conf for the security events, 'security'. The new events
   are:
    - FailedACL - raised when a request violates an ACL check
    - InvalidAccountID - raised when a request fails an authentication
      check due to an invalid account ID
    - SessionLimit - raised when a request fails due to exceeding the
      number of allowed concurrent sessions for a service
    - MemoryLimit - raised when a request fails due to an internal memory
      allocation failure
    - LoadAverageLimit - raised when a request fails because a configured
      load average limit has been reached
    - RequestNotAllowed - raised when a request is not allowed by
      the service
    - AuthMethodNotAllowed - raised when a request used an authentication
      method not allowed by the service
    - RequestBadFormat - raised when a request is received with bad formatting
    - SuccessfulAuth - raised when a request successfully authenticates
    - UnexpectedAddress - raised when a request has a different source address
      then what is expected for a session already in progress with a service
    - ChallengeResponseFailed - raised when a request's attempt to authenticate
      has been challenged, and the request failed the authentication challenge
    - InvalidPassword - raised when a request provides an invalid password
      during an authentication attempt
    - ChallengeSent - raised when an Asterisk service send an authentication
      challenge to a request
    - InvalidTransport - raised when a request attempts to use a transport not
      allowed by the Asterisk service

 * Bridge related events now have two additional fields: BridgeName and
   BridgeCreator. BridgeName is a descriptive name for the bridge;
   BridgeCreator is the name of the entity that created the bridge. This
   affects the following events: ConfbridgeStart, ConfbridgeEnd,
   ConfbridgeJoin, ConfbridgeLeave, ConfbridgeRecord, ConfbridgeStopRecord,
   ConfbridgeMute, ConfbridgeUnmute, ConfbridgeTalking, BlindTransfer,
   AttendedTransfer, BridgeCreate, BridgeDestroy, BridgeEnter, BridgeLeave

ARI
------------------
 * The Bridge data model now contains the additional fields 'name' and
   'creator'. The 'name' field conveys a descriptive name for the bridge;
   the 'creator' field conveys the name of the entity that created the bridge.
   This affects all responses to HTTP requests that return a Bridge data model
   as well as all event derived data models that contain a Bridge data model.
   The POST /bridges operation may now optionally specify a name to give to
   the bridge being created.

 * Added a new ARI resource 'mailboxes' which allows the creation and
   modification of mailboxes managed by external MWI. Modules res_mwi_external
   and res_stasis_mailbox must be enabled to use this resource. For more
   information on external MWI control, see res_mwi_external.

 * Added new events for externally initiated transfers. The event
   BridgeBlindTransfer is now raised when a channel initiates a blind transfer
   of a bridge in the ARI controlled application to the dialplan; the
   BridgeAttendedTransfer event is raised when a channel initiates an
   attended transfer of a bridge in the ARI controlled application to the
   dialplan.

 * Channel variables may now be specified as a body parameter to the
   POST /channels operation. The 'variables' key in the JSON is interpreted
   as a sequence of key/value pairs that will be added to the created channel
   as channel variables. Other parameters in the JSON body are treated as
   query parameters of the same name.

HTTP
------------------
 * Asterisk's HTTP server now supports chunked Transfer-Encoding. This will be
   automatically handled by the HTTP server if a request is received with a
   Transfer-Encoding type of "chunked".

res_pjsip
------------------
 * Path support has been added with the 'support_path' option in registration
   and aor sections.

 * A 'debug' option has been added to the globals section that will allow
   sip messages to be logged.

 * A 'set_var' option has been added to endpoints that will automatically
   set the desired variable(s) on a channel created for that endpoint.

 * Several new tables and columns have been added to the realtime schema for
   the res_pjsip related modules. See the UPGRADE.txt notes for updating
   the database schema.

res_mwi_external
------------------
 * A new module, res_mwi_external, has been added to Asterisk. This module
   acts as a base framework that other modules can build on top of to allow
   an external system to control MWI within Asterisk. For implementations
   that make use of res_mwi_external, see res_mwi_external_ami and
   res_ari_mailboxes. Note that res_mwi_external canflicts with other modules
   that may produce MWI themselves, such as app_voicemail. res_mwi_external
   and other modules that depend on it cannot be built or loaded with
   app_voicemail present.

res_pjsip
------------------
 * DNS functionality will now automatically be enabled if the system configured
   nameservers can be retrieved. If the system configured nameservers can not be
   retrieved the functionality will resort to using system resolution. Functionalty
   such as SRV records and failover will not be available if system resolution
   is in use.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 11 to Asterisk 12 --------------------
------------------------------------------------------------------------------

Overview
------------------

Asterisk 12 is a standard release of the Asterisk project. As such, the
focus of development for this release was on core architectural changes and
major new features. This includes:
 * A more flexible bridging core based on the Bridging API
 * A new internal message bus, Stasis
 * Major standardization and consistency improvements to AMI
 * Addition of the Asterisk RESTful Interface (ARI)
 * A new SIP channel driver, chan_pjsip
In addition, as the vast majority of bridging in Asterisk was migrated to the
Bridging API used by ConfBridge, major changes were made to most of the
interfaces in Asterisk. This includes not only AMI, but also CDRs and CEL.

Specifications have been written for the affected interfaces. These
specifications are available on the Asterisk wiki:
 * AMI - https://wiki.asterisk.org/wiki/x/dAFRAQ
 * CEL - https://wiki.asterisk.org/wiki/x/4ICLAQ
 * CDRs - https://wiki.asterisk.org/wiki/x/pwpRAQ

It is *highly* recommended that anyone migrating to Asterisk 12 read the
information regarding its release both in this file and in the accompanying
UPGRADE.txt file. More detailed information on the major changes can be found
on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/0YCLAQ.


Build System
------------------
 * Added build option DISABLE_INLINE. This option can be used to work around a
   bug in gcc. For more information, see
   http://gcc.gnu.org/bugzilla/show_bug.cgi?id=47816

 * Removed the CHANNEL_TRACE development mode build option. Certain aspects of
   the CHANNEL_TRACE build option were incompatible with the new bridging
   architecture.

 * Asterisk now optionally uses libxslt to improve XML documentation generation
   and maintainability. If libxslt is not available on the system, some XML
   documentation will be incomplete.

 * Asterisk now depends on libjansson. If a package of libjansson is not
   available on your distro, please see http://www.digip.org/jansson/.

 * Asterisk now depends on libuuid and, optionally, uriparser. It is
   recommended that you install uriparser, even if it is optional.

 * The new SIP stack and channel driver uses a particular version of PJSIP.
   Please see https://wiki.asterisk.org/wiki/x/J4GLAQ for more information on
   configuring and installing PJSIP for usage with Asterisk.

 * Optional API was re-implemented to be more portable, and no longer requires
   weak reference support from the compiler. The build option OPTIONAL_API may
   be disabled to disable Optional API support.

Applications
------------------

AgentLogin
------------------
 * Along with AgentRequest, this application has been modified to be a
   replacement for chan_agent. The act of a channel calling the AgentLogin
   application places the channel into a pool of agents that can be
   requested by the AgentRequest application. Note that this application, as
   well as all other agent related functionality, is now provided by the
   app_agent_pool module. See chan_agent and AgentRequest for more information.

 * This application no longer performs agent authentication. If authentication
   is desired, the dialplan needs to perform this function using the
   Authenticate or VMAuthenticate application or through an AGI script before
   running AgentLogin.

 * If this application is called and the agent is already logged in, the
   dialplan will continue exection with the AGENT_STATUS channel variable set
   to ALREADY_LOGGED_IN.

 * The agents.conf schema has changed. Rather than specifying agents on a
   single line in comma delineated fashion, each agent is defined in a separate
   context. This allows agents to use the power of context templates in their
   definition.

 * A number of parameters from agents.conf have been removed. This includes
   maxloginretries, autologoffunavail, updatecdr, goodbye, group, recordformat,
   urlprefix, and savecallsin. These options were obsoleted by the move from
   a channel driver model to the bridging/application model provided by
   app_agent_pool.

AgentRequest
------------------
 * A new application, this will request a logged in agent from the pool and
   bridge the requested channel with the channel calling this application.
   Logged in agents are those channels that called the AgentLogin application.
   If an agent cannot be requested from the pool, the AGENT_STATUS dialplan
   application will be set with an appropriate error value.

AgentMonitorOutgoing
------------------
 * This application has been removed. It was a holdover from when
   AgentCallbackLogin was removed.

AlarmReceiver
------------------
 * Added support for additional Ademco DTMF signalling formats, including
   Express 4+1, Express 4+2, High Speed and Super Fast.

 * Added channel variable ALARMRECEIVER_CALL_LIMIT. This sets the maximum
   call time, in milliseconds, to run the application.

 * Added channel variable ALARMRECEIVER_RETRIES_LIMIT. This sets the
   maximum number of times to retry the call.

 * Added a new configuration option answait. If set, the AlarmReceiver
   application will wait the number of milliseconds specified by answait
   after the channel has answered. Valid values range between 500
   milliseconds and 10000 milliseconds.

 * Added configuration option no_group_meta. If enabled, grouping of metadata
   information in the AlarmReceiver log file will be skipped.

Answer
------------------
 * It is now no longer possible to bypass updating the CDR on the channel
   when answering. CDRs reflect the state of the channel and will always
   reflect the time they were Answered.

BridgeWait
------------------
 * A new application in Asterisk, this will place the calling channel
   into a holding bridge, optionally entertaining them with some form of
   media. Channels participating in a holding bridge do not interact with
   other channels in the same holding bridge. Optionally, however, a channel
   may join as an announcer. Any media passed from an announcer channel is
   played to all channels in the holding bridge. Channels leave a holding
   bridge either when an optional timer expires, or via the ChannelRedirect
   application or AMI Redirect action.

ConfBridge
------------------
 * All participants in a bridge can now be kicked out of a conference room
   by specifying the channel parameter as 'all' in the ConfBridge kick CLI
   command, i.e., 'confbridge kick <conference> all'

 * CLI output for the 'confbridge list' command has been improved. When
   displaying information about a particular bridge, flags will now be shown
   for the participating users indicating properties of that user.

 * The ConfbridgeList event now contains the following fields: WaitMarked,
   EndMarked, and Waiting. This displays additional properties about the
   user's profile, as well as whether or not the user is waiting for a
   Marked user to enter the conference.

 * Added a new option for conference recording, record_file_append. If enabled,
   when the recording is stopped and then re-started, the existing recording
   will be used and appended to.

 * ConfBridge now has the ability to set the language of announcements to the
   conference.  The language can be set on a bridge profile in confbridge.conf
   or by the dialplan function CONFBRIDGE(bridge,language)=en.

ControlPlayback
------------------
 * The channel variable CPLAYBACKSTATUS may now return the value
   'REMOTESTOPPED'. This occurs when playback is stopped by a remote interface,
   such as AMI. See the AMI action ControlPlayback for more information.

Directory
------------------
 * Added the 'a' option, which allows the caller to enter in an additional
   alias for the user in the directory. This option must be used in conjunction
   with the 'f', 'l', or 'b' options. Note that the alias for a user can be
   specified in voicemail.conf.

DumpChan
------------------
 * The output of DumpChan no longer includes the DirectBridge or IndirectBridge
   fields. Instead, if a channel is in a bridge, it includes a BridgeID field
   containing the unique ID of the bridge that the channel happens to be in.

ForkCDR
------------------
 * ForkCDR no longer automatically resets the forked CDR. See the 'r' option
   for more information.

 * Variables are no longer purged from the original CDR. See the 'v' option for
   more information.

 * The 'A' option has been removed. The Answer time on a CDR is never updated
   once set.

 * The 'd' option has been removed. The disposition on a CDR is a function of
   the state of the channel and cannot be altered.

 * The 'D' option has been removed. Who the Party B is on a CDR is a function
   of the state of the respective channels involved in the CDR and cannot be
   altered.

 * The 'r' option has been changed. Previously, ForkCDR always reset the CDR
   such that the start time and, if applicable, the answer time was updated.
   Now, by default, ForkCDR simply forks the CDR, maintaining any times. The
   'r' option now triggers the Reset, setting the start time (and answer time
   if applicable) to the current time. Note that the 'a' option still sets
   the answer time to the current time if the channel was already answered.

 * The 's' option has been removed. A variable can be set on the original CDR
   if desired using the CDR function, and removed from a forked CDR using the
   same function.

 * The 'T' option has been removed. The concept of DONT_TOUCH and LOCKED no
   longer applies in the CDR engine.

 * The 'v' option now prevents the copy of the variables from the original CDR
   to the forked CDR. Previously the variables were always copied but were
   removed from the original. This was changed as removing variables from a CDR
   can have unintended side effects - this option allows the user to prevent
   propagation of variables from the original to the forked without modifying
   the original.

MeetMe
-------------------
 * Added the 'n' option to MeetMe to prevent application of the DENOISE
   function to a channel joining a conference. Some channel drivers that vary
   the number of audio samples in a voice frame will experience significant
   quality problems if a denoiser is attached to the channel; this option gives
   them the ability to remove the denoiser without having to unload func_speex.

MixMonitor
------------------
 * The 'b' option now includes conferences as well as sounds played to the
   participants.

 * The AUDIOHOOK_INHERIT function is no longer needed to keep a MixMonitor
   running during a transfer. If a MixMonitor is started on a channel,
   the MixMonitor will continue to record the audio passing through the
   channel even in the presence of transfers.

NoCDR
------------------
 * The NoCDR application is deprecated. Please use the CDR_PROP function to
   disable CDRs.

 * While the NoCDR application will prevent CDRs for a channel from being
   propagated to registered CDR backends, it will not prevent that data from
   being collected. Hence, a subsequent call to ResetCDR or the CDR_PROP
   function that enables CDRs on a channel will restore those records that have
   not yet been finalized.

ParkAndAnnounce
-------------------
 * The app_parkandannounce module has been removed. The application
   ParkAndAnnounce is now provided by the res_parking module. See the
   res_parking changes for more information.

Queue
-------------------
 * Added queue available hint. The hint can be added to the dialplan using the
   following syntax: exten,hint,Queue:{queue_name}_avail
   For example, if the name of the queue is 'markq':
        exten => 8501,hint,Queue:markq_avail
   This will report 'InUse' if there are no logged in agents or no free agents.
   It will report 'Idle' when an agent is free.

 * Queues now support a hint for member paused state. The hint uses the form
   'Queue:{queue_name}_pause_{member_name}', where {queue_name} and {member_name}
   are the name of the queue and the name of the member to subscribe to,
   respectively. For example: exten => 8501,hint,Queue:sales_pause_mark.
   Members will show as In Use when paused.

 * The configuration options eventwhencalled and eventmemberstatus have been
   removed.  As a result, the AMI events QueueMemberStatus, AgentCalled,
   AgentConnect, AgentComplete, AgentDump, and AgentRingNoAnswer will always be
   sent.  The "Variable" fields will also no longer exist on the Agent* events.
   These events can be filtered out from a connected AMI client using the
   eventfilter setting in manager.conf.

 * The queue log now differentiates between blind and attended transfers. A
   blind transfer will result in a BLINDTRANSFER message with the destination
   context and extension. An attended transfer will result in an
   ATTENDEDTRANSFER message. This message will indicate the method by which
   the attended transfer was completed: "BRIDGE" for a bridge merge, "APP"
   for running an application on a bridge or channel, or "LINK" for linking
   two bridges together with local channels. The queue log will also now detect
   externally initiated blind and attended transfers and record the transfer
   status accordingly.

 * When performing queue pause/unpause on an interface without specifying an
   individual queue, the PAUSEALL/UNPAUSEALL event will only be logged if at
   least one member of any queue exists for that interface.

 * Added the 'queue_log_realtime_use_gmt' option to have timestamps in GMT
   for realtime queue log entries.

ResetCDR
------------------
 * The 'e' option has been deprecated. Use the CDR_PROP function to re-enable
   CDRs when they were previously disabled on a channel.

 * The 'w' and 'a' options have been removed. Dispatching CDRs to registered
   backends occurs on an as-needed basis in order to preserve linkedid
   propagation and other needed behavior.

SayAlphaCase
------------------
 * A new application, this is similar to SayAlpha except that it supports
   case sensitive playback of the specified characters. For example,
   SayAlphaCase(u,aBc) will result in 'a uppercase b c'.

SetAMAFlags
------------------
 * This application is deprecated in favor of CHANNEL(amaflags).

SendDTMF
------------------
 * The SendDTMF application will now accept 'W' as valid input. This will cause
   the application to delay one second while streaming DTMF.

Stasis
------------------
 * A new application in Asterisk 12, this hands control of the channel calling
   the application over to an external system. Currently, external systems
   manipulate channels in Stasis through the Asterisk RESTful Interface (ARI).

UserEvent
------------------
 * UserEvent will now handle duplicate keys by overwriting the previous value
   assigned to the key.

 * In addition to AMI, UserEvent invocations will now be distributed to any
   interested Stasis applications.

VoiceMail
------------------
 * Mailboxes defined by app_voicemail MUST be referenced by the rest of the
   system as mailbox@context.  The rest of the system cannot add @default
   to mailbox identifiers for app_voicemail that do not specify a context
   any longer.  It is a mailbox identifier format that should only be
   interpreted by app_voicemail.

 * The voicemail.conf configuration file now has an 'alias' configuration
   parameter for use with the Directory application. The voicemail realtime
   database table schema has also been updated with an 'alias' column.


Codecs
------------------
 * Pass through support has been added for both VP8 and Opus.

 * Added format attribute negotiation for the Opus codec. Format attribute
   negotiation is provided by the res_format_attr_opus module.


Core
------------------
 * Masquerades as an operation inside Asterisk have been effectively hidden
   by the migration to the Bridging API. As such, many 'quirks' of Asterisk
   no longer occur. This includes renaming of channels, "<ZOMBIE>" channels,
   dropping of frame/audio hooks, and other internal implementation details
   that users had to deal with. This fundamental change has large implications
   throughout the changes documented for this version. For more information
   about the new core architecture of Asterisk, please see the Asterisk wiki.

 * Multiple parties in a bridge may now be transferred. If a participant in a
   multi-party bridge initiates a blind transfer, a Local channel will be used
   to execute the dialplan location that the transferer sent the parties to. If
   a participant in a multi-party bridge initiates an attended transfer,
   several options are possible. If the attended transfer results in a transfer
   to an application, a Local channel is used. If the attended transfer results
   in a transfer to another channel, the resulting channels will be merged into
   a single bridge.

 * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is no longer channel
   driver specific.  If the channel variable is set on the transferrer channel,
   the sound will be played to the target of an attended transfer.

 * The channel variable BRIDGEPEER becomes a comma separated list of peers in
   a multi-party bridge.  The BRIDGEPEER value can have a maximum of 10 peers
   listed.  Any more peers in the bridge will not be included in the list.
   BRIDGEPEER is not valid in holding bridges like parking since those channels
   do not talk to each other even though they are in a bridge.

 * The channel variable BRIDGEPVTCALLID is only valid for two party bridges
   and will contain a value if the BRIDGEPEER's channel driver supports it.

 * A channel variable ATTENDEDTRANSFER is now set which indicates which channel
   was responsible for an attended transfer in a similar fashion to
   BLINDTRANSFER.

 * Modules using the Configuration Framework or Sorcery must have XML
   configuration documentation. This configuration documentation is included
   with the rest of Asterisk's XML documentation, and is accessible via CLI
   commands. See the CLI changes for more information.

AMI (Asterisk Manager Interface)
------------------
 * Major changes were made to both the syntax as well as the semantics of the
   AMI protocol. In particular, AMI events have been substantially improved
   in this version of Asterisk. For more information, please see the AMI
   specification at https://wiki.asterisk.org/wiki/x/dAFRAQ

 * AMI events that reference a particular channel or bridge will now always
   contain a standard set of fields. When multiple channels or bridges are
   referenced in an event, fields for at least some subset of the channels
   and bridges in the event will be prefixed with a descriptive name to avoid
   name collisions. See the AMI event documentation on the Asterisk wiki for
   more information.

 * The CLI command 'manager show commands' no longer truncates command names
   longer than 15 characters and no longer shows authorization requirement
   for commands. 'manager show command' now displays the privileges needed
   for using a given manager command instead.

 * The SIPshowpeer action will now include a 'SubscribeContext' field for a
   peer in its response if the peer has a subscribe context set.

 * The SIPqualifypeer action now acknowledges the request once it has
   established that the request is against a known peer. It also issues a new
   event, 'SIPQualifyPeerDone', once the qualify action has been completed.

 * The PlayDTMF action now supports an optional 'Duration' parameter.  This
   specifies the duration of the digit to be played, in milliseconds.

 * Added VoicemailRefresh action to allow an external entity to trigger mailbox
   updates when changes occur instead of requiring the use of pollmailboxes.

 * Added a new action 'ControlPlayback'. The ControlPlayback action allows an
   AMI client to manipulate audio currently being played back on a channel. The
   supported operations depend on the application being used to send audio to
   the channel. When the audio playback was initiated using the ControlPlayback
   application or CONTROL STREAM FILE AGI command, the audio can be paused,
   stopped, restarted, reversed, or skipped forward. When initiated by other
   mechanisms (such as the Playback application), the audio can be stopped,
   reversed, or skipped forward.

 * Channel related events now contain a snapshot of channel state, adding new
   fields to many of these events.

 * The AMI event 'Newexten' field 'Extension' is deprecated, and may be removed
   in a future release. Please use the common 'Exten' field instead.

 * The AMI event 'UserEvent' from app_userevent now contains the channel state
   fields. The channel state fields will come before the body fields.

 * The AMI events 'ParkedCall', 'ParkedCallTimeOut', 'ParkedCallGiveUp', and
   'UnParkedCall' have changed significantly in the new res_parking module.

   The 'Channel' and 'From' headers are gone. For the channel that was parked
   or is coming out of parking, a 'Parkee' channel snapshot is issued and it
   has a number of fields associated with it. The old 'Channel' header relayed
   the same data as the new 'ParkeeChannel' header.

   The 'From' field was ambiguous and changed meaning depending on the event.
   for most of these, it was the name of the channel that parked the call
   (the 'Parker'). There is no longer a header that provides this channel name,
   however the 'ParkerDialString' will contain a dialstring to redial the
   device that parked the call.

   On UnParkedCall events, the 'From' header would instead represent the
   channel responsible for retrieving the parkee. It receives a channel
   snapshot labeled 'Retriever'. The 'from' field is is replaced with
   'RetrieverChannel'.

   Lastly, the 'Exten' field has been replaced with 'ParkingSpace'.

 * The AMI event 'Parkinglot' (response to 'Parkinglots' command) in a similar
   fashion has changed the field names 'StartExten' and 'StopExten' to
   'StartSpace' and 'StopSpace' respectively.

 * The deprecated use of | (pipe) as a separator in the channelvars setting in
   manager.conf has been removed.

 * Channel Variables conveyed with a channel no longer contain the name of the
   channel as part of the key field, i.e., ChanVariable(SIP/foo): bar=baz is now
   ChanVariable: bar=baz. When multiple channels are present in a single AMI
   event, the various ChanVariable fields will contain a suffix that specifies
   which channel they correspond to.

 * The NewPeerAccount AMI event is no longer raised. The NewAccountCode AMI
   event always conveys the AMI event for a particular channel.

 * All 'Reload' events have been consolidated into a single event type. This
   event will always contain a Module field specifying the name of the module
   and a Status field denoting the result of the reload. All modules now issue
   this event when being reloaded.

 * The 'ModuleLoadReport' event has been removed. Most AMI connections would
   fail to receive this event due to being connected after modules have loaded.
   AMI connections that want to know when Asterisk is ready should listen for
   the 'FullyBooted' event.

 * app_fax now sends the same send fax/receive fax events as res_fax. The
   'FaxSent' event is now the 'SendFAX' event, and the 'FaxReceived' event is
   now the 'ReceiveFAX' event.

 * The 'MusicOnHold' event is now two events: 'MusicOnHoldStart' and
   'MusicOnHoldStop'. The sub type field has been removed.

 * The 'JabberEvent' event has been removed. It is not AMI's purpose to be a
   carrier for another protocol.

 * The Bridge Manager action's 'Playtone' header now accepts more fine-grained
   options. 'Channel1' and 'Channel2' may be specified in order to play a tone
   to the specific channel. 'Both' may be specified to play a tone to both
   channels. The old 'yes' option is still accepted as a way of playing the
   tone to Channel2 only.

 * The AMI 'Status' response event to the AMI Status action replaces the
   'BridgedChannel' and 'BridgedUniqueid' headers with the 'BridgeID' header to
   indicate what bridge the channel is currently in.

 * The AMI 'Hold' event has been moved out of individual channel drivers, into
   core, and is now two events: 'Hold' and 'Unhold'.  The status field has been
   removed.

 * The AMI events in app_queue have been made more consistent with each other.
   Events that reference channels (QueueCaller* and Agent*) will show
   information about each channel.  The (infamous) 'Join' and 'Leave' AMI
   events have been changed to 'QueueCallerJoin' and 'QueueCallerLeave'.

 * The 'MCID' AMI event now publishes a channel snapshot when available and
   its non-channel-snapshot parameters now use either the "MCallerID" or
   'MConnectedID' prefixes with Subaddr*, Name*, and Num* suffixes instead
   of 'CallerID' and 'ConnectedID' to avoid confusion with similarly named
   parameters in the channel snapshot.

 * The AMI events 'Agentlogin' and 'Agentlogoff' have been renamed
   'AgentLogin' and 'AgentLogoff' respectively.

 * The 'Channel' key used in the 'AlarmClear', 'Alarm', and 'DNDState' has been
   renamed "DAHDIChannel" since it does not convey an Asterisk channel name.

 * 'ChannelUpdate' events have been removed.

 * All AMI events now contain a 'SystemName' field, if available.

 * Local channel optimization is now conveyed in two events:
   'LocalOptimizationBegin' and 'LocalOptimizationEnd'. The Begin event is sent
   when the Local channel driver begins attempting to optimize itself out of
   the media path; the End event is sent after the channel halves have
   successfully optimized themselves out of the media path.

 * Local channel information in events is now prefixed with 'LocalOne' and
   'LocalTwo'. This replaces the suffix of '1' and '2' for the two halves of
   the Local channel. This affects the 'LocalBridge', 'LocalOptimizationBegin',
   and 'LocalOptimizationEnd' events.

 * The option 'allowmultiplelogin' can now be set or overriden in a particular
   account. When set in the general context, it will act as the default
   setting for defined accounts.

 * The 'BridgeAction' event was removed. It technically added no value, as the
   Bridge Action already receives confirmation of the bridge through a
   successful completion Event.

 * The 'BridgeExec' events were removed. These events duplicated the events that
   occur in the Briding API, and are conveyed now through BridgeCreate,
   BridgeEnter, and BridgeLeave events.

 * The 'RTCPSent'/'RTCPReceived' events have been significantly modified from
   previous versions. They now report all SR/RR packets sent/received, and
   have been restructured to better reflect the data sent in a SR/RR. In
   particular, the event structure now supports multiple report blocks.

 * Added 'BlindTransfer' and 'AttendedTransfer' events. These events are
   raised when a blind transfer/attended transfer completes successfully.
   They contain information about the transfer that just completed, including
   the location of the transfered channel.

 * Added a 'security' class to AMI which outputs the required fields for
   security messages similar to the log messages from res_security_log

 * The AMI event 'ExtensionStatus' now contains a 'StatusText' field
   that describes the status value in a human readable string.

CDR (Call Detail Records)
------------------
 * Significant changes have been made to the behavior of CDRs. The CDR engine
   was effectively rewritten and built on the Stasis message bus. For a full
   definition of CDR behavior in Asterisk 12, please read the specification
   on the Asterisk wiki (wiki.asterisk.org).

 * CDRs will now be created between all participants in a bridge. For each
   pair of channels in a bridge, a CDR is created to represent the path of
   communication between those two endpoints. This lets an end user choose who
   to bill for what during bridge operations with multiple parties.

 * The duration, billsec, start, answer, and end times now reflect the times
   associated with the current CDR for the channel, as opposed to a cumulative
   measurement of all CDRs for that channel.

 * When a CDR is dispatched, user defined CDR variables from both parties are
   included in the resulting CDR. If both parties have the same variable, only
   the Party A value is provided.

 * Added a new option to cdr.conf, 'debug'. When enabled, significantly more
   information regarding the CDR engine is logged as verbose messages. This
   option should only be used if the behavior of the CDR engine needs to be
   debugged.

 * Added CLI command 'cdr set debug {on|off}'. This toggles the 'debug' setting
   normally configured in cdr.conf.

 * Added CLI command 'cdr show active {channel}'. When {channel} is not
   specified, this command provides a summary of the channels with CDR
   information and their statistics. When {channel} is specified, it shows
   detailed information about all records associated with {channel}.

CEL (Channel Event Logging)
------------------
 * CEL has undergone significant rework in Asterisk 12, and is now built on the
   Stasis message bus. Please see the specification for CEL on the Asterisk
   wiki at https://wiki.asterisk.org/wiki/x/4ICLAQ for more detailed
   information.

 * The 'extra' field of all CEL events that use it now consists of a JSON blob
   with key/value pairs which are defined in the Asterisk 12 CEL documentation.

 * BLINDTRANSFER events now report the transferee bridge unique
   identifier, extension, and context in a JSON blob as the extra string
   instead of the transferee channel name as the peer.

 * ATTENDEDTRANSFER events now report the peer as NULL and additional
   information in the 'extra' string as a JSON blob. For transfers that occur
   between two bridged channels, the 'extra' JSON blob contains the primary
   bridge unique identifier, the secondary channel name, and the secondary
   bridge unique identifier. For transfers that occur between a bridged channel
   and a channel running an app, the 'extra' JSON blob contains the primary
   bridge unique identifier, the secondary channel name, and the app name.

 * LOCAL_OPTIMIZE events have been added to convey local channel
   optimizations with the record occurring for the semi-one channel and
   the semi-two channel name in the peer field.

 * BRIDGE_START, BRIDGE_END, BRIDGE_UPDATE, 3WAY_START, 3WAY_END, CONF_ENTER,
   CONF_EXIT, CONF_START, and CONF_END events have all been removed. These
   events have been replaced by BRIDGE_ENTER/BRIDGE_EXIT. The BRIDGE_ENTER
   and BRIDGE_EXIT events are raised when a channel enters/exits any bridge,
   regardless of whether or not that bridge happens to contain multiple
   parties.

CLI
-------------------
 * When compiled with '--enable-dev-mode', the astobj2 library will now add
   several CLI commands that allow for inspection of ao2 containers that
   register themselves with astobj2. The CLI commands are 'astobj2 container
   dump', 'astobj2 container stats', and 'astobj2 container check'.

 * Added specific CLI commands for bridge inspection. This includes 'bridge
   show all', which lists all bridges in the system, and 'bridge show {id}',
   which provides specific information about a bridge.

 * Added CLI command 'bridge destroy'. This will destroy the specified bridge,
   ejecting the channels currently in the bridge. If the channels cannot
   continue in the dialplan or application that put them in the bridge, they
   will be hung up.

 * Added command 'bridge kick'. This will eject a single channel from a bridge.

 * Added commands to inspect and manipulate the registered bridge technologies.
   This include 'bridge technology show', which lists the registered bridge
   technologies, as well as 'bridge technology {suspend|unsuspend} {tech}',
   which controls whether or not a registered bridge technology can be used
   during smart bridge operations. If a technology is suspended, it will not
   be used when a bridge technology is picked for channels; when unsuspended,
   it can be used again.

 * The command 'config show help {module} {type} {option}' will show
   configuration documentation for modules with XML configuration
   documentation. When {module}, {type}, and {option} are omitted, a listing
   of all modules with registered documentation is displayed. When {module}
   is specified, a listing of all configuration types for that module is
   displayed, along with their synopsis. When {module} and {type} are
   specified, a listing of all configuration options for that type are
   displayed along with their synopsis. When {module}, {type}, and {option}
   are specified, detailed information for that configuration option is
   displayed.

 * Added 'core show sounds' and 'core show sound' CLI commands. These display
   a listing of all installed media sounds available on the system and
   detailed information about a sound, respectively.

 * 'xmldoc dump' has been added. This CLI command will dump the XML
   documentation DOM as a string to the specified file. The Asterisk core
   will populate certain XML elements pulled from the source files with
   additional run-time information; this command lets a user produce the
   XML documentation with all information.

Features
-------------------
 * Parking has been pulled from core and placed into a separate module called
   res_parking. See Parking changes below for more details. Configuration for
   parking should now be performed in res_parking.conf. Configuration for
   parking in features.conf is now unsupported.

 * Core attended transfers now have several new options. While performing an
   attended transfer, the transferer now has the following options:
   - *1 - cancel the attended transfer (configurable via atxferabort)
   - *2 - complete the attended transfer, dropping out of the call
          (configurable via atxfercomplete)
   - *3 - complete the attended transfer, but stay in the call. This will turn
          the call into a multi-party bridge (configurable via atxferthreeway)
   - *4 - swap to the other party. Once an attended transfer has begun, this
          options may be used multiple times (configurable via atxferswap)

 * For DTMF blind and attended transfers, the channel variable TRANSFER_CONTEXT
   must be on the channel initiating the transfer to have any effect.

 * The BRIDGE_FEATURES channel variable would previously only set features for
   the calling party and would set this feature regardless of whether the
   feature was in caps or in lowercase. Use of a caps feature for a letter
   will now apply the feature to the calling party while use of a lowercase
   letter will apply that feature to the called party.

 * Add support for automixmon to the BRIDGE_FEATURES channel variable.

 * The channel variable DYNAMIC_PEERNAME is redundant with BRIDGEPEER and is
   removed.  The more useful DYNAMIC_WHO_ACTIVATED gives the channel name that
   activated the dynamic feature.

 * The channel variables DYNAMIC_FEATURENAME and DYNAMIC_WHO_ACTIVATED are set
   only on the channel executing the dynamic feature.  Executing a dynamic
   feature on the bridge peer in a multi-party bridge will execute it on all
   peers of the activating channel.

 * You can now have the settings for a channel updated using the FEATURE()
   and FEATUREMAP() functions inherited to child channels by setting
   FEATURE(inherit)=yes.

 * automixmon now supports additional channel variables from automon including:
   TOUCH_MIXMONITOR_PREFIX, TOUCH_MIXMONITOR_MESSAGE_START,
   and TOUCH_MIXMONITOR_MESSAGE_STOP

 * A new general features.conf option 'recordingfailsound' has been added which
   allowssetting a failure sound for a user tries to invoke a recording feature
   such as automon or automixmon and it fails.

 * It is no longer necessary (or possible) to define the ATXFER_NULL_TECH in
   features.c for atxferdropcall=no to work properly. This option now just
   works.

Logging
-------------------
 * Added log rotation strategy 'none'. If set, no log rotation strategy will
   be used. Given that this can cause the Asterisk log files to grow quickly,
   this option should only be used if an external mechanism for log management
   is preferred.

Realtime
------------------
 * Dynamic realtime tables for SIP Users can now include a 'path' field. This
   will store the path information for that peer when it registers. Realtime
   tables can also use the 'supportpath' field to enable Path header support.

 * LDAP realtime configurations for SIP Users now have the AstAccountPathSupport
   objectIdentifier. This maps to the supportpath option in sip.conf.

Sorcery
------------------
 * Sorcery is a new data abstraction and object persistence API in Asterisk. It
   provides modules a useful abstraction on top of the many storage mechanisms
   in Asterisk, including the Asterisk Database, static configuration files,
   static Realtime, and dynamic Realtime. It also provides a caching service.
   Users can configure a hierarchy of data storage layers for specific modules
   in sorcery.conf.

 * All future modules which utilize Sorcery for object persistence must have a
   column named "id" within their schema when using the Sorcery realtime module.
   This column must be able to contain a string of up to 128 characters in length.

Security Events Framework
------------------
 * Security Event timestamps now use ISO 8601 formatted date/time instead of
   the "seconds-microseconds" format that it was using previously.

Stasis Message Bus
------------------
 * The Stasis message bus is a publish/subscribe message bus internal to
   Asterisk. Many services in Asterisk are built on the Stasis message bus,
   including AMI, ARI, CDRs, and CEL. Parameters controlling the operation of
   Stasis can be configured in stasis.conf. Note that these parameters operate
   at a very low level in Asterisk, and generally will not require changes.

Channel Drivers
------------------
 * When a channel driver is configured to enable jiterbuffers, they are now
   applied unconditionally when a channel joins a bridge. If a jitterbuffer
   is already set for that channel when it enters, such as by the JITTERBUFFER
   function, then the existing jitterbuffer will be used and the one set by
   the channel driver will not be applied.

chan_agent
------------------
 * chan_agent has been removed and replaced with AgentLogin and AgentRequest
   dialplan applications provided by the app_agent_pool module. Agents are
   connected with callers using the new AgentRequest dialplan application.
   The Agents:<agent-id> device state is available to monitor the status of an
   agent. See agents.conf.sample for valid configuration options.

 * The updatecdr option has been removed. Altering the names of channels on a
   CDR is not supported - the name of the channel is the name of the channel,
   and pretending otherwise helps no one. The AGENTUPDATECDR channel variable
   has also been removed, for the same reason.

 * The endcall and enddtmf configuration options are removed.  Use the
   dialplan function CHANNEL(dtmf-features) to set DTMF features on the agent
   channel before calling AgentLogin.

chan_bridge
------------------
 * chan_bridge has been removed. Its functionality has been incorporated
   directly into the ConfBridge application itself.

chan_dahdi
------------------
 * Added the CLI command 'pri destroy span'. This will destroy the D-channel
   of the specified span and its B-channels. Note that this command should
   only be used if you understand the risks it entails.

 * The CLI command 'dahdi destroy channel' is now 'dahdi destroy channels'.
   A range of channels can be specified to be destroyed. Note that this command
   should only be used if you understand the risks it entails.

 * Added the CLI command 'dahdi create channels'. A range of channels can be
   specified to be created, or the keyword 'new' can be used to add channels
   not yet created.

 * The script specified by the chan_dahdi.conf mwimonitornotify option now gets
   the exact configured mailbox name.  For app_voicemail mailboxes this is
   mailbox@context.

 * Added mwi_vm_boxes that also must be configured for ISDN MWI to be enabled.

chan_iax2
------------------
 * IPv6 support has been added.  We are now able to bind to and
   communicate using IPv6 addresses.

chan_local
------------------
 * The /b option has been removed.

 * chan_local moved into the system core and is no longer a loadable module.

chan_mobile
------------------
 * Added general support for busy detection.

 * Added ECAM command support for Sony Ericsson phones.

chan_pjsip
------------------
 * A new SIP channel driver for Asterisk, chan_pjsip is built on the PJSIP
   SIP stack. A collection of resource modules provides the bulk of the SIP
   functionality. For more information on the new SIP channel driver, see
   https://wiki.asterisk.org/wiki/x/JYGLAQ

chan_sip
------------------
 * Added support for RFC 3327 "Path" headers. This can be enabled in sip.conf
   using the 'supportpath' setting, either on a global basis or on a peer basis.
   This setting enables Asterisk to route outgoing out-of-dialog requests via a
   set of proxies by using a pre-loaded route-set defined by the Path headers in
   the REGISTER request. See Realtime updates for more configuration information.

 * The SIP_CODEC family of variables may now specify more than one codec. Each
   codec must be separated by a comma. The first codec specified is the
   preferred codec for the offer. This allows a dialplan writer to specify both
   audio and video codecs, e.g., Set(SIP_CODEC=ulaw,h264)

 * The 'callevents' parameter has been removed. Hold AMI events are now raised