CHANGES

   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
   in the core, and can be filtered out using the 'eventfilter' parameter
   in manager.conf.

 * Added 'ignore_requested_pref'. When enabled, this will use the preferred
   codecs configured for a peer instead of the requested codec.

 * The option "register_retry_403" has been added to chan_sip to work around
   servers that are known to erroneously send 403 in response to valid
   REGISTER requests and allows Asterisk to continue attepmting to connect.

chan_skinny
------------------
 * Added the 'immeddialkey' parameter. If set, when the user presses the
   configured key the already entered number will be immediately dialed. This
   is useful when the dialplan allows for variable length pattern matching.
   Valid options are '*' and '#'.

 * Added the 'callfwdtimeout' parameter. This configures the amount of time (in
   milliseconds) before a call forward is considered to not be answered.

 * The 'serviceurl' parameter allows Service URLs to be attached to line
   buttons.


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

AGENT
------------------
 * The password option has been disabled, as the AgentLogin application no
   longer provides authentication.

AUDIOHOOK_INHERIT
------------------
 * Due to changes in the Asterisk core, this function is no longer needed to
   preserve a MixMonitor on a channel during transfer operations and dialplan
   execution. It is effectively obsolete.

CDR (function)
------------------
 * The 'amaflags' and 'accountcode' attributes for the CDR function are
   deprecated. Use the CHANNEL function instead to access these attributes.

 * The 'l' option has been removed. When reading a CDR attribute, the most
   recent record is always used. When writing a CDR attribute, all non-finalized
   CDRs are updated.

 * The 'r' option has been removed, for the same reason as the 'l' option.

 * The 's' option has been removed, as LOCKED semantics no longer exist in the
   CDR engine.

CDR_PROP
------------------
 * A new function CDR_PROP has been added. This function lets you set properties
   on a channel's active CDRs. This function is write-only. Properties accept
   boolean values to set/clear them on the channel's CDRs. Valid properties
   include:
   - 'party_a' - make this channel the preferred Party A in any CDR between two
     channels. If two channels have this property set, the creation time of the
     channel is used to determine who is Party A. Note that dialed channels are
     never Party A in a CDR.
   - 'disable' - disable CDRs on this channel. This is analogous to the NoCDR
     application when set to True, and analogous to the 'e' option in ResetCDR
     when set to False.

CHANNEL
------------------
 * Added the argument 'dtmf_features'. This sets the DTMF features that will be
   enabled on a channel when it enters a bridge. Allowed values are 'T', 'K',
   'H', 'W', and 'X', and are analogous to the parameters passed to the Dial
   application.

 * Added the argument 'after_bridge_goto'. This can be set to a parseable Goto
   string, i.e., [[context],extension],priority. If set on a channel, if a
   channel leaves a bridge but is not hung up it will resume dialplan execution
   at that location.

JITTERBUFFER
------------------
 * JITTERBUFFER now accepts an argument of 'disabled' which can be used
   to remove jitterbuffers previously set on a channel with JITTERBUFFER.
   The value of this setting is ignored when disabled is used for the argument.

PJSIP_DIAL_CONTACTS
------------------
 * A new function provided by chan_pjsip, this function can be used in
   conjunction with the Dial application to construct a dial string that will
   dial all contacts on an Address of Record associated with a chan_pjsip
   endpoint.

PJSIP_MEDIA_OFFER
------------------
 * Provided by chan_pjsip, this function sets the codecs to be offerred on the
   outbound channel prior to dialing.

REDIRECTING
------------------
 * Redirecting reasons can now be set to arbitrary strings. This means
   that the REDIRECTING dialplan function can be used to set the redirecting
   reason to any string. It also allows for custom strings to be read as the
   redirecting reason from SIP Diversion headers.

SPEECH_ENGINE
------------------
 * The SPEECH_ENGINE function now supports read operations. When read from, it
   will return the current value of the requested attribute.

VMCOUNT:
------------------
 * 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.


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

res_agi (Asterisk Gateway Interface)
------------------
 * The manager event AGIExec has been split into AGIExecStart and AGIExecEnd.

 * The manager event AsyncAGI has been split into AsyncAGIStart, AsyncAGIExec,
   and AsyncAGIEnd.

 * The CONTROL STREAM FILE command now accepts an offsetms parameter. This
   will start the playback of the audio at the position specified. It will
   also return the final position of the file in 'endpos'.

 * The CONTROL STREAM FILE command will now populate the CPLAYBACKSTATUS
   channel variable if the user stopped the file playback or if a remote
   entity stopped the playback. If neither stopped the playback, it will
   indicate the overall success/failure of the playback. If stopped early,
   the final offset of the file will be set in the CPLAYBACKOFFSET channel
   variable.

 * The SAY ALPHA command now accepts an additional parameter to control
   whether it specifies the case of uppercase, lowercase, or all letters to
   provide functionality similar to SayAlphaCase.

res_ari (Asterisk RESTful Interface) (and others)
------------------
 * The Asterisk RESTful Interface (ARI) provides a mechanism to expose and
   control telephony primitives in Asterisk by remote client. This includes
   channels, bridges, endpoints, media, and other fundamental concepts. Users
   of ARI can develop their own communications applications, controlling
   multiple channels using an HTTP RESTful interface and receiving JSON events
   about the objects via a WebSocket connection. ARI can be configured in
   Asterisk via ari.conf. For more information on ARI, see
   https://wiki.asterisk.org/wiki/x/0YCLAQ

res_parking
-------------------
 * Parking has been extracted from the Asterisk core as a loadable module,
   res_parking. Configuration for parking is now provided by res_parking.conf.
   Configuration through features.conf is no longer supported.

 * res_parking uses the configuration framework. If an invalid configuration is
   supplied, res_parking will fail to load or fail to reload. Previously,
   invalid configurations would generally be accepted, with certain errors
   resulting in individually disabled parking lots.

 * Parked calls are now placed in bridges. While this is largely an
   architectural change, it does have implications on how channels in a parking
   lot are viewed. For example, commands that display channels in bridges will
   now also display the channels in a parking lot.

 * The order of arguments for the new parking applications have been modified.
   Timeout and return context/exten/priority are now implemented as options,
   while the name of the parking lot is now the first parameter. See the
   application documentation for Park, ParkedCall, and ParkAndAnnounce for more
   in-depth information as well as syntax.

 * Extensions are by default no longer automatically created in the dialplan to
   park calls or pickup parked calls. Generation of dialplan extensions can be
   enabled using the 'parkext' configuration option.

 * ADSI functionality for parking is no longer supported. The 'adsipark'
   configuration option has been removed as a result.

 * The PARKINGSLOT channel variable has been deprecated in favor of
   PARKING_SPACE to match the naming scheme of the new system.

 * PARKING_SPACE and PARKEDLOT channel variables will now be set for a parked
   channel even when the configuration option 'comebactoorigin' is enabled.

 * A new CLI command 'parking show' has been added. This allows a user to
   inspect the parking lots that are currently in use.
   'parking show <parkinglot>' will also show the parked calls in a specific
   parking lot.

 * The CLI command 'parkedcalls' is now deprecated in favor of
   'parking show <parkinglot>'.

 * The AMI command 'ParkedCalls' will now accept a 'ParkingLot' argument which
   can be used to get a list of parked calls for a specific parking lot.

 * The AMI command 'Park' field 'Channel2' has been deprecated and replaced
   with 'TimeoutChannel'. If both 'Channel2' and 'TimeoutChannel' are
   specified, 'TimeoutChannel' will be used. The field 'TimeoutChannel' is no
   longer a required argument.

 * The ParkAndAnnounce application is now provided through res_parking instead
   of through the separate app_parkandannounce module.

 * ParkAndAnnounce will no longer go to the next position in dialplan on timeout
   by default. Instead, it will follow the timeout rules of the parking lot. The
   old behavior can be reproduced by using the 'c' option.

 * Dynamic parking lots will now fail to be created under the following
   conditions:
   - if the parking lot specified by PARKINGDYNAMIC does not exist
   - if they require exclusive park and parkedcall extensions which overlap
     with existing parking lots.

 * Dynamic parking lots will be cleared on reload for dynamic parking lots that
   currently contain no calls. Dynamic parking lots containing parked calls
   will persist through the reloads without alteration.

 * If 'parkext_exclusive' is set for a parking lot and that extension is
   already in use when that parking lot tries to register it, this is now
   considered a parking system configuration error. Configurations which do
   this will be rejected.

 * Added channel variable PARKER_FLAT. This contains the name of the extension
   that would be used if 'comebacktoorigin' is enabled. This can be useful when
   comebacktoorigin is disabled, but the dialplan or an external control
   mechanism wants to use the extension in the park-dial context that was
   generated to re-dial the parker on timeout.

res_pjsip (and many others)
------------------
 * A large number of resource modules make up the SIP stack based on pjsip.
   The chan_pjsip channel driver users these resource modules to provide
   various SIP functionality in Asterisk. The majority of configuration for
   these modules is performed in pjsip.conf. Other modules may use their
   own configuration files.

 * Added 'set_var' option for an endpoint. For each variable specified that
   variable gets set upon creation of a channel involving the endpoint.

res_rtp_asterisk
------------------
 * ICE/STUN/TURN support in res_rtp_asterisk has been made optional.  To enable
   them, an Asterisk-specific version of PJSIP needs to be installed.
   Tarballs are available from https://github.com/asterisk/pjproject/tags/.

res_statsd/res_chan_stats
------------------
 * A new resource module, res_statsd, has been added, which acts as a statsd
   client. This module allows Asterisk to publish statistics to a statsd
   server. In conjunction with res_chan_stats, it will publish statistics about
   channels to the statsd server. It can be configured via res_statsd.conf.

res_xmpp
------------------
 * Device state for XMPP buddies is now available using the following format:
   XMPP/<client name>/<buddy address>