asterisk/UPGRADE.txt

===========================================================
===
=== Information for upgrading between Asterisk versions
===
=== These files document all the changes that MUST be taken
=== into account when upgrading between the Asterisk
=== versions listed below. These changes may require that
=== you modify your configuration files, dialplan or (in
=== some cases) source code if you have your own Asterisk
=== modules or patches. These files also include advance
=== notice of any functionality that has been marked as
=== 'deprecated' and may be removed in a future release,
=== along with the suggested replacement functionality.
===
=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
=== UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6
=== UPGRADE-1.8.txt -- Upgrade info for 1.6 to 1.8
=== UPGRADE-10.txt  -- Upgrade info for 1.8 to 10
=== UPGRADE-11.txt  -- Upgrade info for 10 to 11
===
===========================================================

From 12.7.0 to 12.8.0:

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

From 12.6.0 to 12.7.0:

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

 - Added the pjsip.conf system type disable_tcp_switch option.  The option
   allows the user to disable switching from UDP to TCP transports described
   by RFC 3261 section 18.1.1.

From 12.6.0 to 12.6.1:
- Due to the POODLE vulnerability (see 
  https://cve.mitre.org/cgi-bin/cvename.cgi?name=CVE-2014-3566), the
  default TLS method for TLS clients will no longer allow SSLv3. As
  SSLv2 was already deprecated, it is no longer allowed by default as
  well. TLS servers no longer allow SSLv2 or SSLv3 connections. This
  affects the chan_sip channel driver, AMI, and the Asterisk HTTP server.

- The res_jabber resource module no longer uses SSLv3 to connect to an
  XMPP server. It will now only use TLSv1 or later methods.

From 12.5.0 to 12.6.0:

ConfBridge:
 - Added 'Admin' header to ConfbridgeJoin, ConfbridgeLeave, ConfbridgeMute,
   ConfbridgeUnmute, and ConfbridgeTalking AMI events.

ControlPlayback:
 - The ControlPlayback and 'control stream file' AGI command will no longer
   implicitly answer the channel. If you do not answer the channel prior to
   using either this application or AGI command, you must send Progress
   first.

From 12.4.0 to 12.5.0:

ARI:
 - The ARI version has been changed from 1.4.0 to 1.5.0. This is to reflect
   the backwards compatible changes listed in the CHANGES file.

AMI:
 - The AMI version has been changed from 2.4.0 to 2.5.0. This is to reflect
   the backwards compatible changes listed in the CHANGES file.

From 12.3.2 to 12.4.0:

 - The safe_asterisk script was previously not installed on top of an existing
   version. This caused bug-fixes in that script not to be deployed. If your
   safe_asterisk script is customized, be sure to keep your changes. Custom
   values for variables should be created in *.sh file(s) inside
   ASTETCDIR/startup.d/. See ASTERISK-21965.

 - Changed a log message in safe_asterisk and the $NOTIFY mail subject. If
   you use tools to parse either of them, update your parse functions
   accordingly. The changed strings are:
   - "Exited on signal $EXITSIGNAL" => "Asterisk exited on signal $EXITSIGNAL."
   - "Asterisk Died" => "Asterisk on $MACHINE died (sig $EXITSIGNAL)"

AMI:
 - The AMI version has been changed from 2.3.0 to 2.4.0. This is to reflect
   the backwards compatible changes listed in the CHANGES file.

ARI:
 - Added a compatibility option 'websocket_write_timeout'.  When a websocket
   connection exists where Asterisk writes a substantial amount of data to
   the connected client, and the connected client is slow to process the
   received data, the socket may be disconnected.  In such cases, it may be
   necessary to adjust this value.
   Default is 100 ms.

 - The ARI version has been changed from 1.3.0 to 1.4.0. This is to reflect
   the backwards compatible changes listed in the CHANGES file.

chan_dahdi:
 - Added the inband_on_setup_ack compatibility option to chan_dahdi.conf to
   deal with switches that don't send an inband progress indication in the
   SETUP ACKNOWLEDGE message.

chan_pjsip:
 - Added a compatibility option 'websocket_write_timeout'.  When a websocket
   connection exists where Asterisk writes a substantial amount of data to
   the connected client, and the connected client is slow to process the
   received data, the socket may be disconnected.  In such cases, it may be
   necessary to adjust this value.
   Default is 100 ms.

 - Added a 'force_avp' option to chan_pjsip which will force the usage of
   'RTP/AVP', 'RTP/AVPF', 'RTP/SAVP', or 'RTP/SAVPF' as the media transport type
   in SDP offers depending on settings, even when DTLS is used for media
   encryption.

 - Added a 'media_use_received_transport' option to chan_pjsip which will
   cause the SDP answer to use the media transport as received in the SDP
   offer.

chan_sip:
 - Added a compatibility option 'websocket_write_timeout'.  When a websocket
   connection exists where Asterisk writes a substantial amount of data to
   the connected client, and the connected client is slow to process the
   received data, the socket may be disconnected.  In such cases, it may be
   necessary to adjust this value.
   Default is 100 ms.

 - Added a 'force_avp' option for chan_sip. When enabled this option will
   cause the media transport in the offer or answer SDP to be 'RTP/AVP',
   'RTP/AVPF', 'RTP/SAVP', or 'RTP/SAVPF' even if a DTLS stream has been
   configured. This option can be set to improve interoperability with WebRTC
   clients that don't use the RFC defined transport for DTLS.

 - The 'dtlsverify' option in chan_sip now has additional values besides
   'yes' and 'no'. If 'yes' is specified both the certificate and fingerprint
   will be verified. If 'no' is specified then neither the certificate or
   fingerprint is verified. If 'certificate' is specified then only the
   certificate is verified. If 'fingerprint' is specified then only the
   fingerprint is verified.

 - A 'dtlsfingerprint' option has been added to chan_sip which allows the
   hash to be specified for the DTLS fingerprint placed in SDP. Supported
   values are 'sha-1' and 'sha-256' with 'sha-256' being the default.

HTTP:
 - Added support for persistent HTTP connections.  To enable persistent
   HTTP connections configure the keep alive time between HTTP requests.  The
   keep alive time between HTTP requests is configured in http.conf with the
   session_keep_alive parameter.

From 12.3.0 to 12.3.1:

 - MixMonitor AMI actions now require users to have authorization classes.
   * MixMonitor - system
   * MixMonitorMute - call or system
   * StopMixMonitor - call or system

 - Added http.conf session_inactivity timer option to close HTTP connections
   that aren't doing anything.

 - Removed the undocumented manager.conf block-sockets option.  It interferes with
   TCP/TLS inactivity timeouts.

From 12.2.0 to 12.3.0:

 - The asterisk command line -I option and the asterisk.conf internal_timing
   option are removed and always enabled if any timing module is loaded.

AMI:
 - The AMI version has been changed from 2.2.0 to 2.3.0. This is to reflect
   the backwards compatible changes listed in the CHANGES file.

ARI:
 - The ARI version has been changed from 1.2.0 to 1.3.0. This is to reflect
   the backwards compatible changes listed in the CHANGES file.

Build Options:
 - Added a new Compiler Flag, REF_DEBUG. When enabled, reference counted
   objects will emit additional debug information to the refs log file located
   in the standard Asterisk log file directory. This log file is useful in
   tracking down object leaks and other reference counting issues. Prior to
   this version, this option was only available by modifying the source code
   directly. This change also includes a new script, refcounter.py, in the
   contrib folder that will process the refs log file.

chan_sip:
 - Made set SIPREFERREDBYHDR as inheritable for better chan_pjsip

From 12.1.0 to 12.2.0:

AMI:
 - The AMI version has been changed from 2.1.0 to 2.2.0. This is to reflect
   the backwards compatible changes listed in the CHANGES file.

ARI:
 - The ARI version has been changed from 1.1.0 to 1.2.0. This is to reflect
   the backwards compatible changes listed in the CHANGES file.

 - A bug fix in bridge creation has caused a behavioural change in how
   subscriptions are created for bridges. A bridge created through ARI, does
   not, by itself, have a subscription created for any particular Stasis
   application. When a channel in a Stasis application joins a bridge, an
   implicit event subscription is created for that bridge as well. Previously,
   when a channel left such a bridge, the subscription was leaked; this allowed
   for later bridge events to continue to be pushed to the subscribed
   applications. That leak has been fixed; as a result, bridge events that were
   delivered after a channel left the bridge are no longer delivered. An
   application must subscribe to a bridge through the applications resource if
   it wishes to receive all events related to a bridge.

ConfBridge:
 - The sound_place_into_conference sound used in Confbridge is now deprecated
   and is no longer functional since it has been broken since its inception
   and the fix involved using a different method to achieve the same goal. The
   new method to achieve this functionality is by using sound_begin to play
   a sound to the conference when waitmarked users are moved into the
   conference.

IAX2:
 - When communicating with a peer on an Asterisk 1.4 or earlier system, the
   chan_iax2 parameter 'connectedline' must be set to "no" in iax.conf. This
   prevents an incompatible connected line frame from an Astersik 1.8 or later
   system from causing a hangup in an Asterisk 1.4 or earlier system. Note that
   this particular incompatibility has always existed between 1.4 and 1.8 and
   later versions; this upgrade note is simply informing users of its existance.

ODBC:
- A compatibility setting, allow_empty_string_in_nontext, has been added to
  res_odbc.conf. When enabled (default behavior), empty column values are
  stored as empty strings during realtime updates. Disabling this option
  causes empty column values to be stored as NULLs for non-text columns.

  Disable it for PostgreSQL backends in order to avoid errors caused by
  updating integer columns with an empty string instead of NULL
  (sippeers, sipregs, ..).

PJSIP:
 - The PJSIP registrar now stores the contents of the User-Agent header of
   incoming REGISTER requests for each contact that is registered. If using
   realtime for PJSIP contacts, this means that the schema has been updated to
   add a user_agent column. An alembic revision has been added to facilitate
   this update.
 
 - PJSIP endpoints now have a "message_context" option that can be used to
   determine where to route incoming MESSAGE requests from the endpoint.

Realtime Configuration:
 - PJSIP endpoint columns 'tos_audio' and 'tos_video' have been changed from
   yes/no enumerators to string values. 'cos_audio' and 'cos_video' have been
   changed from yes/no enumerators to integer values. PJSIP transport column
   'tos' has been changed from a yes/no enumerator to a string value. 'cos' has
   been changed from a yes/no enumerator to an integer value.

 - The 'queues' and 'queue_members' realtime tables have been added to the
   config Alembic scripts.

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

From 12.0.0 to 12.1.0:

- The per console verbose level feature as previously implemented caused a
  large performance penalty.  The fix required some minor incompatibilities
  if the new rasterisk is used to connect to an earlier version.  If the new
  rasterisk connects to an older Asterisk version then the root console verbose
  level is always affected by the "core set verbose" command of the remote
  console even though it may appear to only affect the current console.  If
  an older version of rasterisk connects to the new version then the
  "core set verbose" command will have no effect.

ARI:
 - The ARI version has been changed from 1.0.0 to 1.1.0. This is to reflect
   the backwards compatible changes listed below.

 - Added a new ARI resource 'mailboxes' which allows the creation and
   modification of mailboxes managed by external MWI. Modules res_mwi_external
   and res_stasis_mailbox must be enabled to use this resource.

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

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

 - Subscribing to the same device state twice now responds with success
   instead of returning error on the second attempt.

AMI:
 - The AMI version has been changed from 2.0.0 to 2.1.0. This is to reflect
   the backwards compatible changes listed below.

 - The DialStatus field in the DialEnd event can now have additional values.
   This includes ABORT, CONTINUE, and GOTO.

 - The res_mwi_external_ami module can, if loaded, provide additional AMI
   actions and events that convey MWI state within Asterisk. This includes
   the MWIGet, MWIUpdate, and MWIDelete actions, as well as the MWIGet and
   MWIGetComplete events that occur in response to an MWIGet action.

 - AMI now contains a new class authorization, 'security'. This is used with
   the following new events: FailedACL, InvalidAccountID, SessionLimit,
   MemoryLimit, LoadAverageLimit, RequestNotAllowed, AuthMethodNotAllowed,
   RequestBadFormat, SuccessfulAuth, UnexpectedAddress, ChallengeResponseFailed,
   InvalidPassword, ChallengeSent, and InvalidTransport.

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

CDRs:
 - The "endbeforehexten" setting now defaults to "yes", instead of "no".
   When set to "no", yhis setting will cause a new CDR to be generated when a
   channel enters into hangup logic (either the 'h' extension or a hangup
   handler subroutine). In general, this is not the preferred default: this
   causes extra CDRs to be generated for a channel in many common dialplans.

CLI commands:
 - "core show settings" now lists the current console verbosity in addition
   to the root console verbosity.

 - "core set verbose" has not been able to support the by module verbose
   logging levels since verbose logging levels were made per console.  That
   syntax is now removed and a silence option added in its place.

Configuration Files:
 - The 'verbose' setting in logger.conf still takes an optional argument,
   specifying the verbosity level for each logging destination.  However,
   the default is now to once again follow the current root console level.
   As a result, using the AMI Command action with "core set verbose" could
   again set the root console verbose level and affect the verbose level
   logged.

 - res_fax now returns the correct rates for V.27ter (4800 or 9600 bit/s).
   Because of this the default settings would not load, so the minrate (minimum
   transmission rate) option in res_fax.conf was changed to default to 4800
   since that is the minimum rate for v.27 which is included in the default
   modem options.

Realtime Configuration:
 - WARNING: The database migration script that adds the 'extensions' table for
   realtime had to be modified due to an error when installing for MySQL.  The
   'extensions' table's 'id' column was changed to be a primary key.  This could
   potentially cause a migration problem.  If so, it may be necessary to
   manually alter the affected table/column to bring it back in line with the
   migration scripts.

 - New columns have been added to realtime tables for 'support_path' on
   ps_registrations and ps_aors and for 'path' on ps_contacts for the new
   SIP Path support in chan_pjsip.

 - The following new tables have been added for pjsip realtime: 'ps_systems',
   'ps_globals', 'ps_tranports', 'ps_registrations'.

 - The following columns were added to the 'ps_aors' realtime table:
   'maximum_expiration', 'outbound_proxy', and 'support_path'.

 - The following columns were added to the 'ps_contacts' realtime table:
   'outbound_proxy' and 'path'.

 - New columns have been added to the ps_endpoints realtime table for the
   'media_address', 'redirect_method' and 'set_var' options.  Also the
   'mwi_fromuser' column was renamed to 'mwi_from_user'.

 - A new column was added to the 'ps_globals' realtime table for the 'debug'
   option.

From 11 to 12:
There are many significant architectural changes in Asterisk 12. It is
recommended that you not only read through this document for important
changes that affect an upgrade, but that you also read through the CHANGES
document in depth to better understand the new options available to you.

Additional information on the architectural changes made in Asterisk can be
found on the Asterisk wiki (https://wiki.asterisk.org)

Of particular note, the following systems in Asterisk underwent significant
changes. Documentation for the changes and a specification for their
behavior in Asterisk 12 is also available on the Asterisk wiki.
 - AMI: Many events were changed, and the semantics of channels and bridges
        were defined. In particular, how channels and bridges behave under
        transfer scenarios and situations involving multiple parties has
        changed significantly. See https://wiki.asterisk.org/wiki/x/dAFRAQ
        for more information.
 - CDR: CDR logic was extracted from the many locations it existed in across
        Asterisk and implemented as a consumer of Stasis message bus events.
        As a result, consistency of records has improved significantly and the
        behavior of CDRs in transfer scenarios has been defined in the CDR
        specification. However, significant behavioral changes in CDRs resulted
        from the transition. The most significant change is the addition of
        CDR entries when a channel who is the Party A in a CDR leaves a bridge.
        See https://wiki.asterisk.org/wiki/x/pwpRAQ for more information.
 - CEL: Much like CDRs, CEL was removed from the many locations it existed in
        across Asterisk and implemented as a consumer of Stasis message bus
        events. It now closely follows the Bridging API model of channels and
        bridges, and has a much closer consistency of conveyed events as AMI.
        For the changes in events, see https://wiki.asterisk.org/wiki/x/4ICLAQ.

Build System:
 - Removed the CHANNEL_TRACE development mode build option. Certain aspects of
   the CHANNEL_TRACE build option were incompatible with the new bridging
   architecture.

 - Asterisk now depends on libjansson, libuuid and optionally (but recommended)
   libxslt and uriparser.

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

AgentLogin and chan_agent:
 - Along with AgentRequest, this application has been modified to be a
   replacement for chan_agent. The chan_agent module and the Agent channel
   driver have been removed from Asterisk, as the concept of a channel driver
   proxying in front of another channel driver was incompatible with the new
   architecture (and has had numerous problems through past versions of
   Asterisk). The act of a channel calling the AgentLogin application places the
   channel into a pool of agents that can be requested by the AgentRequest
   application. Note that this application, as well as all other agent related
   functionality, is now provided by the app_agent_pool module.

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

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

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

 - The AGENTUPDATECDR channel variable has also been removed, for the same
   reason as the updatecdr option.

 - 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.

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

Answer
 - It is no longer possible to bypass updating the CDR when answering a
   channel. CDRs are based on the channel state and will be updated when
   the channel is Answered.

ControlPlayback
 - The channel variable CPLAYBACKSTATUS may now return the value
   'REMOTESTOPPED' when playback is stopped by an external entity.

DISA
 - This application now has a dependency on the app_cdr module. It uses this
   module to hide the CDR created prior to execution of the DISA application.

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:
 - Nearly every parameter in ForkCDR has been updated and changed to reflect
   the changes in CDRs. Please see the documentation for the ForkCDR
   application, as well as the CDR specification on the Asterisk wiki.

NoCDR:
 - The NoCDR application has been deprecated. Please use the CDR_PROP function
   to disable CDRs on a channel.

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

ResetCDR:
 - 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.
 - The 'e' option is deprecated. Please use the CDR_PROP function to enable
   CDRs on a channel that they were previously disabled on.
 - The ResetCDR application is no longer a part of core Asterisk, and instead
   is now delivered as part of app_cdr.

Queues:
 - Queue strategy rrmemory now has a predictable order similar to strategy
   rrordered. Members will be called in the order that they are added to the
   queue.

 - Removed the queues.conf check_state_unknown option.  It is no longer
   necessary.

 - It is now possible to play the Queue prompts to the first user waiting in a
   call queue. Note that this may impact the ability for agents to talk with
   users, as a prompt may still be playing when an agent connects to the user.
   This ability is disabled by default but can be enabled on an individual
   queue using the 'announce-to-first-user' option.

 - 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.

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

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. Systems
   using voicemail with realtime should update their schemas accordingly.

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_bridge
 - chan_bridge is removed and its functionality is incorporated into ConfBridge
   itself.

chan_dahdi:
 - Analog port dialing and deferred DTMF dialing for PRI now distinguishes
   between 'w' and 'W'.  The 'w' pauses dialing for half a second.  The 'W'
   pauses dialing for one second.

 - The default for inband_on_proceeding has changed to no.

 - 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.

 - 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_local:
 - The /b option has been removed.

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

chan_sip:
 - 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.

 - 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.

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.

 - The following channel variables have changed behavior which is described in
   the CHANGES file: TRANSFER_CONTEXT, BRIDGEPEER, BRIDGEPVTCALLID,
   ATTENDED_TRANSFER_COMPLETE_SOUND, DYNAMIC_FEATURENAME, and DYNAMIC_PEERNAME.

AMI (Asterisk Manager Interface):
 - Version 1.4 - The details of what happens to a channel when a masquerade
   happens (transfers, parking, etc) have changed.
   - The Masquerade event now includes the Uniqueid's of the clone and original
     channels.
   - Channels no longer swap Uniqueid's as a result of the masquerade.
   - Instead of a shell game of renames, there's now a single rename, appending
     <ZOMBIE> to the name of the original channel.

 - *Major* changes were made to both the syntax as well as the semantics of the
   AMI protocol. In particular, AMI events have been substantially modified
   and improved in this version of Asterisk. The major event changes are listed
   below.
   - NewPeerAccount has been removed. NewAccountCode is raised instead.
   - Reload events have been consolidated and standardized.
   - ModuleLoadReport has been removed.
   - FaxSent is now SendFAX; FaxReceived is now ReceiveFAX. This standardizes
     app_fax and res_fax events.
   - MusicOnHold has been replaced with MusicOnHoldStart and MusicOnHoldStop.
   - JabberEvent has been removed.
   - Hold is now in the core and will now raise Hold and Unhold events.
   - Join is now QueueCallerJoin.
   - Leave is now QueueCallerLeave.
   - Agentlogin/Agentlogoff is now AgentLogin/AgentLogoff, respectively.
   - ChannelUpdate has been removed.
   - Local channel optimization is now conveyed via LocalOptimizationBegin and
     LocalOptimizationEnd.
   - BridgeAction and BridgeExec have been removed.
   - BlindTransfer and AttendedTransfer events were added.
   - Dial is now DialBegin and DialEnd.
   - DTMF is now DTMFBegin and DTMFEnd.
   - Bridge has been replaced with BridgeCreate, BridgeEnter, BridgeLeave, and
     BridgeDestroy
   - MusicOnHold has been replaced with MusicOnHoldStart and MusicOnHoldStop
   - AGIExec is now AGIExecStart and AGIExecEnd
   - AsyncAGI is now AsyncAGIStart, AsyncAGIExec, and AsyncAGIEnd

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

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

 - 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 '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.

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

 - The SIP SIPqualifypeer action now sends a response indicating it will qualify
   a peer once a peer has been found to qualify.  Once the qualify has been
   completed it will now issue a SIPqualifypeerdone event.

 - 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 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 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.

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.

 - CDR backends can no longer be unloaded while billing data is in flight. This
   helps to prevent loss of billing data during restarts and shutdowns.

CEL:
 - The Uniqueid field for a channel is now a stable identifier, and will not
   change due to transfers, parking, etc.

 - 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. A summary of the affected events is below:
   - 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.
   - BLINDTRANSFER/ATTENDEDTRANSFER events now report the peer as NULL and
     additional information in the extra string field.

Dialplan Functions:

 - Certain dialplan functions have been marked as 'dangerous', and may only be
   executed from the dialplan. Execution from extenal sources (AMI's GetVar and
   SetVar actions; etc.) may be inhibited by setting live_dangerously in the
   [options] section of asterisk.conf to no. SHELL(), channel locking, and
   direct file read/write functions are marked as dangerous. DB_DELETE() and
   REALTIME_DESTROY() are marked as dangerous for reads, but can now safely
   accept writes (which ignore the provided value).
 - The default value for live_dangerously was changed from yes (in Asterisk 11
   and earlier) to no (in Asterisk 12 and greater).

Dialplan:
 - All channel and global variable names are evaluated in a case-sensitive
   manner. In previous versions of Asterisk, variables created and evaluated in
   the dialplan were evaluated case-insensitively, but built-in variables and
   variable evaluation done internally within Asterisk was done
   case-sensitively.

 - Asterisk has always had code to ignore dash '-' characters that are not
   part of a character set in the dialplan extensions.  The code now
   consistently ignores these characters when matching dialplan extensions.

 - BRIDGE_FEATURES channel variable is now casesensitive for feature letter
   codes. Uppercase variants apply them to the calling party while lowercase
   variants apply them to the called party.

Features:
 - The features.conf [applicationmap] <FeatureName>  ActivatedBy option is
   no longer honored.  The feature is always activated by the channel that has
   DYNAMIC_FEATURES defined on it when it enters the bridge. Use predial to set
   different values of DYNAMIC_FEATURES on the channels

 - Executing a dynamic feature on the bridge peer in a multi-party bridge will
   execute it on all peers of the activating channel.

 - There is no longer an explicit 'features reload' CLI command. Features can
   still be reloaded using 'module reload features'.

 - 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.

Parking:
 - Parking has been extracted from the Asterisk core as a loadable module,
   res_parking.

 - Configuration is found in res_parking.conf. It is no longer supported in
   features.conf

 - The arguments for the Park, ParkedCall, and ParkAndAnnounce applications 
   have been modified significantly. See the application documents for 
   specific details.

 - Numerous changes to Parking related applications, AMI and CLI commands and
   internal inter-workings  have been made. Please read the CHANGES file for 
   the detailed list.

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

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.

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.

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/.


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