CHANGES

   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>
   If any resource is available the device state is considered to be not in use.
   If no resources exist or all are unavailable the device state is considered
   to be unavailable.


Scripts
------------------

Realtime/Database Scripts
------------------
 * Asterisk previously included example db schemas in the contrib/realtime/
   directory of the source tree.  This has been replaced by a set of database
   migrations using the Alembic framework.  This allows you to use alembic to
   initialize the database for you.  It will also serve as a database migration
   tool when upgrading Asterisk in the future.

   See contrib/ast-db-manage/README.md for more details.

sip_to_res_pjsip.py
-------------------
 * A new script has been added in the contrib/scripts/sip_to_res_pjsip folder.
   This python script will convert an existing sip.conf file to a
   pjsip.conf file, for use with the chan_pjsip channel driver. This script
   is meant to be an aid in converting an existing chan_sip configuration to
   a chan_pjsip configuration, but it is expected that configuration beyond
   what the script provides will be needed.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 10 to Asterisk 11 --------------------
------------------------------------------------------------------------------

Build System
-------------------
 * The Asterisk build system will now build and install a shared library
   (libasteriskssl.so) used to wrap various initialization and shutdown functions
   from the libssl and libcrypto libraries provided by OpenSSL. This is done so
   that Asterisk can ensure that these functions do *not* get called by any
   modules that are loaded into Asterisk, since they should only be called once
   in any single process. If desired, this feature can be disabled by supplying
   the "--disable-asteriskssl" option to the configure script.

 * A new make target, 'full', has been added to the Makefile.  This performs
   the same compilation actions as make all, but will also scan the entirety of
   each source file for documentation.  This option is needed to generate AMI
   event documentation.  Note that your system must have Python in order for
   this make target to succeed.

 * The optimization portion of the build system has been reworked to avoid
   broken builds on certain architectures.  All architecture-specific
   optimization has been removed in favor of using -march=native to allow gcc
   to detect the environment in which it is running when possible.  This can
   be toggled as BUILD_NATIVE under "Compiler Flags" in menuselect.

 * BUILD_CFLAGS and BUILD_LDFLAGS can now be passed to menuselect, e.g.,
   make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever"

 * Remove "asterisk/version.h" in favor of "asterisk/ast_version.h".  If you
   previously parsed the header file to obtain the version of Asterisk, you
   will now have to go through Asterisk to get the version information.


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

Bridge
-------------------
 * Added 'F()' option. Similar to the dial option, this can be supplied with
   arguments indicating where the callee should go after the caller is hung up,
   or without options specified, the priority after the Queue will be used.


ConfBridge
-------------------
 * Added menu action admin_toggle_mute_participants.  This will mute / unmute
   all non-admin participants on a conference.  The confbridge configuration
   file also allows for the default sounds played to all conference users when
   this occurs to be overriden using sound_participants_unmuted and
   sound_participants_muted.

 * Added menu action participant_count.  This will playback the number of
   current participants in a conference.

 * Added announcement configuration option to user profile. If set the sound
   file will be played to the user, and only the user, upon joining the
   conference bridge.

 * Added record_file_append option that defaults to "yes", but if set to no
   will create a new file between each start/stop recording.


Dial
-------------------
 * Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller
   channels respectively before the callee channels are called.


ExternalIVR
-------------------
 * Added support for IPv6.

 * Add interrupt ('I') command to ExternalIVR.  Sending this command from an
   external process will cause the current playlist to be cleared, including
   stopping any audio file that is currently playing.  This is useful when you
   want to interrupt audio playback only when specific DTMF is entered by the
   caller.


FollowMe
-------------------
 * A new option, 'I' has been added to app_followme. By setting this option,
   Asterisk will not update the caller with connected line changes when they
   occur.  This is similar to app_dial and app_queue.

 * The 'N' option is now ignored if the call is already answered.

 * Added 'b' and 'B' options to FollowMe that execute a Gosub on callee
   and caller channels respectively before the callee channels are called.

 * The winning FollowMe outgoing call is now put on hold if the caller put it on
   hold.


MixMonitor
------------------
 * MixMonitor hooks now have IDs associated with them which can be used to
   assign a target to StopMixMonitor. Use of MixMonitor's i(variable) option
   will allow storage of the MixMontior ID in a channel variable. StopMixmonitor
   now accepts that ID as an argument.

 * Added 'm' option, which stores a copy of the recording as a voicemail in the
   indicated mailboxes.


MySQL
-------------------
 * The connect action in app_mysql now allows you to specify a port number to
   connect to.  This is useful if you run a MySQL server on a non-standard
   port number.


OSP Applications
-------------------
 * Increased the default number of allowed destinations from 5 to 12.


Page
-------------------
 * The app_page application now no longer depends on DAHDI or app_meetme.  It
   has been re-architected to use app_confbridge internally.


Queue
-------------------
 * Added queue options autopausebusy and autopauseunavail for automatically
   pausing a queue member when their device reports busy or congestion.

 * The 'ignorebusy' option for queue members has been deprecated in favor of
   the option 'ringinuse. Also a 'queue set ringinuse' CLI command has been
   added as well as an AMI action 'QueueMemberRingInUse' to set this variable on a
   per interface basis. Individual ringinuse values can now be set in
   queues.conf via an argument to member definitions. Lastly, the queue
   'ringinuse' setting now only determines defaults for the per member
   'ringinuse' setting and does not override per member settings like it does
   in earlier versions.

 * Added 'F()' option. Similar to the dial option, this can be supplied with
   arguments indicating where the callee should go after the caller is hung up,
   or without options specified, the priority after the Queue will be used.

 * Added new option log_member_name_as_agent, which will cause the membername to
   be logged in the agent field for ADDMEMBER and REMOVEMEMBER queue events if a
   state_interface has been set.

 * Add queue monitoring hints.  exten => 8501,hint,Queue:markq.

 * App_queue will now play periodic announcements for the caller that
   holds the first position in the queue while waiting for answer.

SayUnixTime
------------------
 * Added 'j' option to SayUnixTime. SayUnixTime no longer auto jumps to extension
   when receiving DTMF.  Use the 'j' option to enable extension jumping. Also
   changed arguments to SayUnixTime so that every option is truly optional even
   when using multiple options (so that j option could be used without having to
   manually specify timezone and format) There are other benefits, e.g., format
   can now be used without specifying time zone as well.


Voicemail
------------------
 * Addition of the VM_INFO function - see Function changes.

 * The imapserver, imapport, and imapflags configuration options can now be
   overriden on a user by user basis.

 * When voicemail plays a message's envelope with saycid set to yes, when
   reaching the caller id field it will play a recording of a file with the same
   base name as the sender's callerid if there is a similarly named file in
   <astspooldir>/recordings/callerids/

 * Voicemails now contains a unique message identifier "msg_id", which is stored
   in the message envelope with the sound files.  IMAP backends will now store
   the message identifiers with a header of "X-Asterisk-VM-Message-ID".  ODBC
   backends will store the message identifier in a "msg_id" column.  See
   UPGRADE.txt for more information.

 * Added VoiceMailPlayMsg application.  This application will play a single
   voicemail message from a mailbox.  The result of the application, SUCCESS or
   FAILED, is stored in the channel variable VOICEMAIL_PLAYBACKSTATUS.


Functions
------------------
 * Hangup handlers can be attached to channels using the CHANNEL() function.
   Hangup handlers will run when the channel is hung up similar to the h
   extension. The hangup_handler_push option will push a GoSub compatible
   location in the dialplan onto the channel's hangup handler stack.  The
   hangup_handler_pop option will remove the last added location, and optionally
   replace it with a new GoSub compatible location.  The hangup_handler_wipe
   option will remove all locations on the stack, and optionally add a new
   location.

 * The expression parser now recognizes the ABS() absolute value function,
   which will convert negative floating point values to positive values.

 * FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan
   control of faxdetect.

 * Addition of the VM_INFO function that can be used to retrieve voicemail
   user information, such as the email address and full name.
   The MAILBOX_EXISTS dialplan function has been deprecated in favour of
   VM_INFO.

 * The REDIRECTING function now supports the redirecting original party id
   and reason.

 * Two new functions have been added: FEATURE() and FEATUREMAP().  FEATURE()
   lets you set some of the configuration options from the [general] section
   of features.conf on a per-channel basis.  FEATUREMAP() lets you customize
   the key sequence used to activate built-in features, such as blindxfer,
   and automon.  See the built-in documentation for details.

 * MESSAGE(from) for incoming SIP messages now returns "display-name" <uri>
   instead of simply the uri.  This is the format that MessageSend() can use
   in the from parameter for outgoing SIP messages.

 * Added the PRESENCE_STATE function.  This allows retrieving presence state
   information from any presence state provider.  It also allows setting
   presence state information from a CustomPresence presence state provider.
   See AMI/CLI changes for related commands.

 * Added the AMI_CLIENT function to make manager account attributes available
   to the dialplan. It currently supports returning the current number of
   active sessions for a given account.

 * Added support for private party ID information to CALLERID, CONNECTEDLINE,
   and the REDIRECTING functions.


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

chan_local
------------------
 * Added a manager event "LocalBridge" for local channel call bridges between
   the two pseudo-channels created.


chan_dahdi
------------------
 * Added dialtone_detect option for analog ports to disconnect incoming
   calls when dialtone is detected.

 * Added option colp_send to send ISDN connected line information.  Allowed
   settings are block, to not send any connected line information; connect, to
   send connected line information on initial connect; and update, to send
   information on any update during a call.  Default is update.

 * Add options namedcallgroup and namedpickupgroup to support installations
   where a higher number of groups (>64) is required.

 * Added support to use private party ID information with PRI calls.


chan_motif
------------------
 * A new channel driver named chan_motif has been added which provides support for
   Google Talk and Jingle in a single channel driver. This new channel driver includes
   support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk,
   hold, unhold, and ringing notification. It is also compliant with the current Jingle
   specification, current Google Jingle specification, and the original Google Talk
   protocol.


chan_ooh323
------------------
 * Added NAT support for RTP.  Setting in config is 'nat', which can be set
   globally and overriden on a peer by peer basis.

 * Direct media functionality has been added. Options in config are:
   directmedia (directrtp) and directrtpsetup (earlydirect)

 * ChannelUpdate events now contain a CallRef header.


chan_sip
------------------
 * Asterisk will no longer substitute CID number for CID name in the display
   name field if CID number exists without a CID name. This change improves
   compatibility with certain device features such as Avaya IP500's directory
   lookup service.

 * A new setting for autocreatepeer (autocreatepeer=persistent) allows peers
   created using that setting to not be removed during SIP reload.

 * Added settings recordonfeature and recordofffeature.  When receiving an INFO
   request with a "Record:" header, this will turn the requested feature on/off.
   Allowed values are 'automon', 'automixmon', and blank to disable.  Note that
   dynamic features must be enabled and configured properly on the requesting
   channel for this to function properly.

 * Add support to realtime for the 'callbackextension' option.

 * When multiple peers exist with the same address, but differing
   callbackextension options, incoming requests that are matched by address
   will be matched to the peer with the matching callbackextension if it is
   available.

 * Two new NAT options, auto_force_rport and auto_comedia, have been added
   which set the force_rport and comedia options automatically if Asterisk
   detects that an incoming SIP request crossed a NAT after being sent by
   the remote endpoint.

 * The default global nat setting in sip.conf has been changed from force_rport
   to auto_force_rport.

 * NAT settings are now a combinable list of options. The equivalent of the
   deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.

 * Adds an option send_diversion which can be disabled to prevent
   diversion headers from automatically being added to INVITE requests.

 * Add support for lightweight NAT keepalive. If enabled a blank packet will
   be sent to the remote host at a given interval to keep the NAT mapping open.
   This can be enabled using the keepalive configuration option.

 * Add option 'tonezone' to specify country code for indications.  This option
   can be set both globally and overridden for specific peers.

 * The SIP Security Events Framework now supports IPv6.

 * Add a new setting for directmedia, 'outgoing', to alleviate INVITE glares
   between multiple user agents. When set, for directmedia reinvites,
   Asterisk will not send an immediate reinvite on an incoming call leg. This
   option is useful when peered with another SIP user agent that is known to
   send immediate direct media reinvites upon call establishment.

 * Add support for WebSocket transport. This can be configured using 'ws' or 'wss'
   as the transport.

 * Add options subminexpiry and submaxexpiry to set limits of subscription
   timer independently from registration timer settings. The setting of the
   registration timer limits still is done by options minexpiry, maxexpiry
   and defaultexpiry. For backwards compatibility the setting of minexpiry
   and maxexpiry also is used to configure the subscription timer limits if
   subminexpiry and submaxexpiry are not set in sip.conf.

 * Set registration timer limits to default values when reloading sip
   configuration and values are not set by configuration.

 * Add options namedcallgroup and namedpickupgroup to support installations
   where a higher number of groups (>64) is required.

 * When a MESSAGE request is received, the address the request was received from
   is now saved in the SIP_RECVADDR variable.

 * Add ANI2/OLI parsing for SIP.  The "From" header in INVITE requests is now
   parsed for the presence of "isup-oli", "ss7-oli", or "oli" tags.  If present,
   the ANI2/OLI information is set on the channel, which can be retrieved using
   the CALLERID function.

 * Peers can now be configured to support negotiation of ICE candidates using
   the setting icesupport.  See res_rtp_asterisk changes for more information.

 * Added support for format attribute negotiation.  See the Codecs changes for
   more information.

 * Extra headers specified with SIPAddHeader are sent with the REFER message
   when using Transfer application. See refer_addheaders in sip.conf.sample.

 * Added support to use private party ID information with calls.

 * Adds an option discard_remote_hold_retrieval that when set stops telling
   the peer to start music on hold.


chan_skinny
------------------
 * Added skinny version 17 protocol support.


chan_unistim
--------------------
 * Added option 'dtmf_duration' allowing playback time of DTMF tones to be set

 * Modified option 'date_format' to allow options to display date in 31Jan and Jan31
   formats as options 0 and 1. The previous options 0 and 1 now map to options 2 and 3
   as per the UNISTIM protocol.

 * Fixed issues with dialtone not matching indications.conf and mute stopping rx
   as well as tx. Also fixed issue with call "Timer" displaying as French "Dur\E9e"

 * Added ability to use multiple lines for a single phone.  This allows multiple
   calls to occur on a single phone, using callwaiting and switching between calls.

 * Added option 'sharpdial' allowing end dialing by pressing # key

 * Added option 'interdigit_timer' to control phone dial timeout

 * Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance

 * Added global 'debug' option, that enables debug in channel driver

 * Added ability to translate on-screen menu in multiple languages. Tested on
   Russian languages.  Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
   ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
   menu of phone

 * In addition to English added French and Russian languages for on-screen menus

 * Reworked dialing number input: added dialing by timeout, immediate dial on
   on dialplan compare, phone number length now not limited by screen size

 * Added ability to pickup a call using features.conf defined value and
   on-screen key


chan_mISDN:
------------------
 * Add options namedcallgroup and namedpickupgroup to support installations
   where a higher number of groups (>64) is required.

 * Added support to use private party ID information with calls.


Core
------------------
 * The minimum DTMF duration can now be configured in asterisk.conf
   as "mindtmfduration". The default value is (as before) set to 80 ms.
   (previously it was only available in source code)

 * Named ACLs can now be specified in acl.conf and used in configurations that
   use ACLs. As a general rule, if some derivative of 'permit' or 'deny' is
   used to specify an ACL, a similar form of 'acl' will add a named ACL to the
   working ACL. In addition, some CLI commands have been added to provide
   show information and allow for module reloading - see CLI Changes.

 * Rules in ACLs (specified using 'permit' and 'deny') can now contain multiple
   items (separated by commas), and items in the rule can be negated by prefixing
   them with '!'. This simplifies Asterisk Realtime configurations, since it is no
   longer necessray to control the order that the 'permit' and 'deny' columns are
   returned from queries.

 * DUNDi now allows the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to
   be used within the dynamic weight attribute when specifying a mapping.

 * CEL backends can now be configured to show "USER_DEFINED" in the EventName
   header, instead of putting the user defined event name there.  When enabled
   the UserDefType header is added for user defined events.  This feature is
   enabled with the setting show_user_defined.

 * Macro has been deprecated in favor of GoSub.  For redirecting and connected
   line purposes use the following variables instead of their macro equivalents:
   REDIRECTING_SEND_SUB, REDIRECTING_SEND_SUB_ARGS, CONNECTED_LINE_SEND_SUB,
   CONNECTED_LINE_SEND_SUB_ARGS.  For CCSS, use cc_callback_sub instead of
   cc_callback_macro in channel configurations.

 * Asterisk can now use a system-provided NetBSD editline library (libedit) if it
   is available.

 * Call files now support the "early_media" option to connect with an outgoing
   extension when early media is received.

 * Added support to use private party ID information with calls.


AGI
------------------
 * A new channel variable, AGIEXITONHANGUP, has been added which allows
   Asterisk to behave like it did in Asterisk 1.4 and earlier where the
   AGI application would exit immediately after a channel hangup is detected.

 * IPv6 addresses are now supported when using FastAGI (agi://).  Hostnames
   are resolved and each address is attempted in turn until one succeeds or
   all fail.


AMI (Asterisk Manager Interface)
------------------
 * The originate action now has an option "EarlyMedia" that enables the
   call to bridge when we get early media in the call. Previously,
   early media was disregarded always when originating calls using AMI.

 * Added setvar= option to manager accounts (much like sip.conf)

 * Originate now generates an error response if the extension given is not found
   in the dialplan

 * MixMonitor will now show IDs associated with the mixmonitor upon creating
   them if the i(variable) option is used. StopMixMonitor will accept
   MixMonitorID as an option to close specific MixMonitors.

 * The SIPshowpeer manager action response field "SIP-Forcerport" has been
   updated to include information about peers configured with
   nat=auto_force_rport by returning "A" if auto_force_rport is set and nat is
   detected, and "a" if it is set and nat is not detected. "Y" and "N" are still
   returned if auto_force_rport is not enabled.

 * Added SIPpeerstatus manager command which will generate PeerStatus events
   similar to the existing PeerStatus events found in chan_sip on demand.

 * Hangup now can take a regular expression as the Channel option.  If you want
   to hangup multiple channels, use /regex/ as the Channel option.  Existing
   behavior to hanging up a single channel is unchanged, but if you pass a regex,
   the manager will send you a list of channels back that were hung up.

 * Support for IPv6 addresses has been added.

 * AMI Events can now be documented in the Asterisk source. Note that AMI event
   documentation is only generated when Asterisk is compiled using 'make full'.
   See the CLI section for commands to display AMI event information.