CHANGES

==============================================================================
===
=== This file documents the new and/or enhanced functionality added in
=== the Asterisk versions listed below. This file does NOT include
=== changes in behavior that would not be backwards compatible with
=== previous versions; for that information see the UPGRADE.txt file
=== and the other UPGRADE files for older releases.
===
==============================================================================
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 11 to Asterisk 12 --------------------
------------------------------------------------------------------------------

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

AgentMonitorOutgoing
------------------
 * The 'c' option has been removed. It is not possible to modify the name of a
   channel involved in a CDR.

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, 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.

 * 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. 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.

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.

Queue
-------------------
 * Add queue available hint.  exten => 8501,hint,Queue:markq_avail
   Note: the suffix '_avail' after the queuename.
   Reports 'InUse' for no logged in agents or no free agents.
   Reports 'Idle' when an agent is free.

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.

SetAMAFlags
------------------
 * This application is deprecated in favor of the CHANNEL function.


Core
------------------
 * 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.

AMI (Asterisk Manager Interface)
------------------
 * 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.

 * 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.

 * Added 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.
   First, channel snapshot data is included for both the parker and the parkee
   in lieu of the "From" and "Channel" fields. They follow standard channel
   snapshot format but each field is suffixed with 'Parker' or 'Parkee'
   depending on which side it applies to. The 'Exten' field is replaced with
   'ParkingSpace' since the registration of extensions as for parking spaces
   is no longer mandatory.

 * 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".

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.

CDR (Call Detail Records)
------------------
 * Significant changes have been made to the behavior of CDRs. 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 multi-party bridges or bridge operations during
   transfer scenarios.

 * 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.

Features
-------------------
 * 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 automixmonitor to the BRIDGE_FEATURES channel variable.

 * Parking has been pulled from core and placed into a separate module called
   res_parking. See Parking changes below for more details.

 * 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.

Logging
-------------------
 * 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.

Parking
-------------------
 * Parking is now implemented as a module instead of as core functionality.
   The preferred way to configure parking is now through res_parking.conf while
   configuration through features.conf is not currently supported.

 * Parked calls are now placed in bridges. This is a largely architectural change,
   but it could have some implications in allowing for new parked call retrieval
   methods and the contents of parking lots will be visible though certain bridge
   commands.

 * The order of arguments for the new parking applications are different from the
   old ones to be more intuitive. Timeout and return context/exten/priority are now
   implemented as options. parking_lot_name 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 no longer automatically created in the dialplan to park calls,
   pickup parked calls, etc by default.

 * adsipark is no longer supported under the new parking model

 * 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 comebactoorigin=yes

 * New CLI command 'parking show' allows you to inspect the currently in use
   parking lots. 'parking show <parkinglot>' will also show the parked calls
   in that 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 only for a specific parking lot.

 * 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.

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
------------------
 * 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.


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
------------------
 * 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 as the updatecdr option.

chan_local
------------------
 * The /b option is 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_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.


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

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.

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.


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

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

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.

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

Sorcery
------------------
 * 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.

app_userevent
------------------
 * UserEvent will now handle duplicate keys by overwriting the previous value
   assigned to the key. UserEvent invocations will also be distributed to any
   interested res_stasis applications.

------------------------------------------------------------------------------
--- 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 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.

 * The AMI Hangup event now includes the AccountCode header so you can easily
   correlate with AMI Newchannel events.

 * The QueueMemberStatus, QueueMemberAdded, and QueueMember events now include
   the StateInterface of the queue member.

 * Added AMI event SessionTimeout in the Call category that is issued when a
   call is terminated due to either RTP stream inactivity or SIP session timer
   expiration.

 * CEL events can now contain a user defined header UserDefType.  See core
   changes for more information.

 * OOH323 ChannelUpdate events now contain a CallRef header.

 * Added PresenceState command.  This command will report the presence state for
   the given presence provider.

 * Added Parkinglots command.  This will list all parking lots as a series of
   AMI Parkinglot events.

 * Added MessageSend command.  This behaves in the same manner as the
   MessageSend application, and is a technolgoy agnostic mechanism to send out
   of call text messages.

 * Added "message" class authorization.  This grants an account permission to
   send out of call messages.  Write-only.


CLI
-------------------
 * The "dialplan add include" command has been modified to create context a context
   if one does not already exist. For instance, "dialplan add include foo into bar"
   will create context "bar" if it does not already exist.

 * A  "dialplan remove context" command has been added to remove a context from
   the dialplan

 * The "mixmonitor list <channel>" command will now show MixMonitor ID, and the
   filenames of all running mixmonitors on a channel.

 * The debug level of "pri set debug" is now a bitmask ranging from 0 to 15 if
   numeric instead of 0, 1, or 2.

 * "stun show status" will show a table describing how the STUN client is
   behaving.

 * "acl show [named acl]" will show information regarding a Named ACL.  The
   acl module can be reloaded with "reload acl".

 * Added CLI command to display AMI event information - "manager show events",
   which shows a list of all known and documented AMI events, and "manager show
   event [event name]", which shows detail information about a specific AMI
   event.

 * The result of the CLI command "queue show" now includes the state interface
   information of the queue member.

 * The command "core set verbose" will now set a separate level of logging for
   each remote console without affecting any other console.

 * Added command "cdr show pgsql status" to check connection status

 * "sip show channel" will now display the complete route set.

 * Added "presencestate list" command.  This command will list all custom
   presence states that have been set by using the PRESENCE_STATE dialplan
   function.

 * Added "presencestate change <entity> <state>[,<subtype>[,message[,options]]]"
   command.  This changes a custom presence to a new state.


Codecs
-------------------
 * Codec lists may now be modified by the '!' character, to allow succinct
   specification of a list of codecs allowed and disallowed, without the
   requirement to use two different keywords.  For example, to specify all
   codecs except g729 and g723, one need only specify allow=all,!g729,!g723.

 * Add support for parsing SDP attributes, generating SDP attributes, and
   passing it through. This support includes codecs such as H.263, H.264, SILK,