CHANGES

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

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

ARI
-----------------
 * A new ARI method has been added to the channels resource. "create" allows for
 you to create a new channel and place that channel into a Stasis application. This
 is similar to origination except that the specified channel is not dialed. This
 allows for an application writer to create a channel, perform manipulations on it,
 and then delay dialing the channel until later.

 * To complement the "create" method, a "dial" method has been added to the channels
 resource in order to place a call to a created channel.

 * All operations that initiate playback of media on a resource now support
   a list of media URIs. The list of URIs are played in the order they are
   presented to the resource. A new event, "PlaybackContinuing", is raised when
   a media URI finishes but before the next media URI starts. When a list is
   played, the "Playback" model will contain the optional attribute
   "next_media_uri", which specifies the next media URI in the list to be played
   back to the resource. The "PlaybackFinished" event is raised when all media
   URIs are done.

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

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

ChanSpy
------------------
 * Added the 'l' option, which forces ChanSpy's audiohook to use a long queue
   to store the audio frames. This option is useful if audio loss is
   experienced when using ChanSpy, but may introduce some delay in the audio
   feed on the listening channel.

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

ControlPlayback
------------------
 * Remote files can now be retrieved and played back. See the Playback
   dialplan application for more details.

Playback
------------------
 * Remote files can now be retrieved and played back via the Playback and other
   media playback dialplan applications. This is done by directly providing
   the URL to play to the dialplan application:
     same => n,Playback(http://1.1.1.1/howler-monkeys-fl.wav)
   Note that unlike 'normal' media files, the entire URI to the file must be
   provided, including the file extension. Currently, on HTTP and HTTPS URI
   schemes are supported.

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


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

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


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

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

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

chan_sip
------------------
 * New 'rtpbindaddr' global setting. This allows a user to define which
   ipaddress to bind the rtpengine to. For example, chan_sip might bind
   to eth0 (10.0.0.2) but rtpengine to eth1 (192.168.1.10).
 * DTLS related configuration options can now be set at a general level.
   Enabling DTLS support, though, requires enabling it at the user
   or peer level.
 * Added the possibility to set the From: header through the the SIP dial
   string (populating the fromuser/fromdomain fields), complementing the
   [!dnid] option for the To: header that has existed since 1.6.0 (1d6b192).
   NOTE: This is again separated by an exclamation mark, so the To: header may
   not contain one of those.

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

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

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

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

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

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

 * The core now supports a 'media cache', which stores temporary media files
   retrieved from external sources. CLI commands have been added to manipulate
   and display the cached files, including:
   - 'media cache show <all>' - show all cached media files, or details about
     one particular cached media file
   - 'media cache refresh <item>' - force a refresh of a particular media file
     in the cache
   - 'media cache delete <item>' - remove an item from the cache
   - 'media cache create <uri>' - retrieve a URI and store it in the cache

 * The ability for hints to be automatically created as a result of device state
   changes now exists in the PBX. This functionality is referred to as "autohints"
   and is configurable in extensions.conf by placing "autohints=yes" in the
   context. If enabled then a hint will be automatically created with the name of
   the device.

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

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

CURL
------------------
 * The CURL function now supports a write option, which will save the retrieved
   file to a location on disk. As an example:
     same => n,Set(CURL(https://1.1.1.1/foo.wav)=/tmp/foo.wav)
   will save 'foo.wav' to /tmp.

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


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

res_http_media_cache
------------------
 * A backend for the core media cache, this module retrieves media files from
   a remote HTTP(S) server and stores them in the core media cache for later
   playback.

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

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

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

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

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

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

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

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

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

Queue
-------------------
 * Added field ReasonPause on QueueMemberStatus if set when paused, the reason
   the queue member was paused.
 * Added field LastPause on QueueMemberStatus for time when started the last
   pause for a queue member.
 * Show the time when started the last pause for queue member on CLI for command
   'queue show'.

--- Functionality changes from Asterisk 13.9.0 to Asterisk 13.10.0 -----------
------------------------------------------------------------------------------

res_fax
------------------
 * Added FAXMODE variable to let dialplan know what fax transport was used.
   FAXMODE variable is set to either "audio" or "T38".

res_pjsip
------------------
 * Added new status Updated to AMI event ContactStatus on update registration

 * Added "reg_server" to contacts.
   If the Asterisk system name is set in asterisk.conf, it will be stored
   into the "reg_server" field in the ps_contacts table to facilitate
   multi-server setups.

app_confbridge
------------------
 * Added a bridge profile option called regcontext that allows you to
   dynamically register the conference bridge name as an extension into
   the specified context.  This allows tracking down conferences on multi-
   server installations via alternate means (DUNDI for example). By default
   this feature is not used.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.8.0 to Asterisk 13.9.0 ------------
------------------------------------------------------------------------------

res_parking:
 - The dynamic parking lot creation channel variables PARKINGDYNAMIC,
   PARKINGDYNCONTEXT, PARKINGDYNEXTEN, and PARKINGDYNPOS are now looked
   for in the parker's channel instead of the parked channel.  This is only
   of significance if the parker uses blind transfer or the DTMF one-step
   parking feature.  You need to use the double underscore '__' inheritance
   for these variables.  The indefinite inheritance is also recommended
   for the PARKINGEXTEN variable.

res_pjsip
------------------
 * Added new global option (disable_multi_domain) to pjsip.
   Disabling Multi Domain can improve realtime performace by reducing
   number of database requsts.

chan_pjsip
------------------
 * Added 'pjsip show channelstats' CLI command.

res_pjsip_outbound_publish
------------------
 * Added support for setting the transport used on outbound publish
   using the transport configuration option.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.7.0 to Asterisk 13.8.0 ------------
------------------------------------------------------------------------------

res_pjsip_caller_id
------------------
 * Per RFC3325, the 'From' header is now anonymized on outgoing calls when
   caller id presentation is prohibited.

res_pjsip_config_wizard
------------------
 * A new command (pjsip export config_wizard primitives) has been added that
   will export all the pjsip objects it created to the console or a file
   suitable for reuse in a pjsip.conf file.

Build System
------------------
 * To help insure that Asterisk is compiled and run with the same known
   version of pjproject, a new option (--with-pjproject-bundled) has been
   added to ./configure.  When specified, the version of pjproject specified
   in third-party/versions.mak will be downloaded and configured.  When you
   make Asterisk, the build process will also automatically build pjproject
   and Asterisk will be statically linked to it.  Once a particular version
   of pjproject is configured and built, it won't be configured or built
   again unless you run a 'make distclean'.

   To facilitate testing, when 'make install' is run, the pjsua and pjsystest
   utilities and the pjproject python bindings will be installed in
   ASTDATADIR/third-party/pjproject.

   The default behavior remains building with the shared pjproject
   installation, if any.

app_confbridge
------------------
 * Added CONFBRIDGE_INFO(muted,) for querying the muted conference state.

 * Added Muted header to AMI ConfbridgeListRooms action response list events
   to indicate the muted conference state.

 * Added Muted column to CLI "confbridge list" output to indicate the muted
   conference state and made the locked column a yes/no value instead of a
   locked/unlocked value.

REDIRECTING(reason)
------------------
 * The REDIRECTING(reason) value is now treated consistently between
   chan_sip and chan_pjsip.

   Both channel drivers match incoming reason values with values documented
   by REDIRECTING(reason) and values documented by RFC5806 regardless of
   whether they are quoted or not.  RFC5806 values are mapped to the
   equivalent REDIRECTING(reason) documented value and is set in
   REDIRECTING(reason).  e.g., an incoming RFC5806 'unconditional' value or a
   quoted string version ('"unconditional"') is converted to
   REDIRECTING(reason)'s 'cfu' value.  The user's dialplan only needs to deal
   with 'cfu' instead of any of the aliases.

   The incoming 480 response reason text supported by chan_sip checks for
   known reason values and if not matched then puts quotes around the reason
   string and assigns that to REDIRECTING(reason).

   Both channel drivers send outgoing known REDIRECTING(reason) values as the
   unquoted RFC5806 equivalent.  User custom values are either sent as is or
   with added quotes if SIP doesn't allow a character within the value as
   part of a RFC3261 Section 25.1 token.  Note that there are still
   limitations on what characters can be put in a custom user value.  e.g.,
   embedding quotes in the middle of the reason string is just going to cause
   you grief.

 * Setting a REDIRECTING(reason) value now recognizes RFC5806 aliases.
   e.g., Setting REDIRECTING(reason) to 'unconditional' is converted to the
   'cfu' value.

res_pjproject
------------------
 * This module is the successor of res_pjsip_log_forwarder.  As well as
   handling the log forwarding (which now displays as 'pjproject:0' instead
   of 'pjsip:0'), it also adds a 'pjproject show buildopts' command to the CLI.
   This displays the compiled-in options of the pjproject installation
   Asterisk is currently running against.

 * Another feature of this module is the ability to map pjproject log levels
   to Asterisk log levels, or to suppress the pjproject log messages
   altogether.  Many of the messages emitted by pjproject itself are the result
   of errors which Asterisk will ultimately handle so the messages can be
   misleading or just noise.  A new config file (pjproject.conf) has been added
   to configure the mapping and a new CLI command (pjproject show log mappings)
   has been added to display the mappings currently in use.

res_pjsip
------------------
 * Transports are now reloadable.  In testing, no in-progress calls were
   disrupted if the ip address or port weren't changed, but the possibility
   still exists.  To make sure there are no unintentional drops, a new option
   'allow_reload', which defaults to 'no' has been added to transport.  If
   left at the default, changes to the particular transport will be ignored.
   If set to 'yes', changes (if any) will be applied.

 * Added new global option (regcontext) to pjsip. When set, Asterisk will
   dynamically create and destroy a NoOp priority 1 extension
   for a given endpoint who registers or unregisters with us.

 * Endpoints and aors can now be identified by the username and realm in an
   incoming Authorization header.  To use this feature, add "auth_username"
   to your endpoint's "identify_by" list.  You can combine "auth_username"
   and the original "username" to test both the From/To and Authorization
   headers.  For endpoints, the order is controlled by the global
   "endpoint_identifier_order" setting.  For matching aors to an endpoint
   for inbound registration, the order is controlled by this option.

 * In conjunction with the "auth_username" change, 3 new options have been
   added to the global configuration object that control how many unidentified
   requests over a certain period from the same IP address can be received
   before a security altert is generated.  A new CLI command
   "pjsip show unidentified_requests" will list the current candidates.

res_pjsip_history
------------------
 * A new module, res_pjsip_history, has been added that provides SIP history
   viewing/filtering from the CLI. The module is intended to be used on systems
   with busy SIP traffic, where existing forms of viewing SIP messages - such
   as the res_pjsip_logger - may be inadequate. The module provides two new
   CLI commands:
   - 'pjsip set history {on|off|clear}' - this enables/disables SIP history
     capturing, as well as clears an existing history capture. Note that SIP
     packets captured are stored in memory until cleared. As a result, the
     history capture should only be used for debugging/viewing purposes, and
     should *NOT* be left permanently enabled on a system.
   - 'pjsip show history' - displays the captured SIP history. When invoked
     with no options, the entire captured history is displayed. Two options
     are available:
     -- 'entry <num>' - display a detailed view of a single SIP message in
        the history
     -- 'where ...' - filter the history based on some expression. For more
        information on filtering, view the current CLI help for the
        'pjsip show history' command.

Voicemail
------------------
 * app_voicemail and res_mwi_external can now be built together.  The default
   remains to build app_voicemail and not res_mwi_external but if they are
   both built, the load order will cause res_mwi_external to load first and
   app_voicemail will be skipped.  Use 'preload=app_voicemail.so' in
   modules.conf to force app_voicemail to be the voicemail provider.

res_pjsip_sdp_rtp
------------------
 * A new option (bind_rtp_to_media_address) has been added to endpoint which
   will cause res_pjsip_sdp_rtp to actually bind the RTP instance to the
   media_address as well as using it in the SDP.  If set, RTP packets will now
   originate from the media address instead of the operating system's "primary"
   ip address.

res_rtp_asterisk
------------------
 * A new configuration section - ice_host_candidates - has been added to
   rtp.conf, allowing automatically discovered ICE host candidates to be
   overriden. This allows an Asterisk server behind a 1:1 NAT to send its
   external IP as a host candidate rather than relying on STUN to discover it.

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

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

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

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

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

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

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

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

res_sorcery_memory_cache
------------------------
 * A new caching strategy, full_backend_cache, has been added which caches
   all stored objects in the backend. When enabled all objects will be
   expired or go stale according to the configuration. As well when enabled
   all retrieval operations will be performed against the cache instead of
   the backend.

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

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


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

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

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

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

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

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

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

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

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

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


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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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


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

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

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


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

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

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

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

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

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

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

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

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

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

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

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


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

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

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


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

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


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

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