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 14 to Asterisk 15 --------------------
------------------------------------------------------------------------------

Build System
------------------
 * LOW_MEMORY no longer has an effect on Asterisk ABI.  Symbols that were
   previously suppressed by LOW_MEMORY are now replaced by stub functions.
   Asterisk built with LOW_MEMORY can now successfully load binary modules
   built without LOW_MEMORY and vice versa.

chan_sip
------------------
 * If an offer is received with optional SRTP (a media stream with RTP/AVP but
   which contains a crypto line) chan_sip will now accept it and enable SRTP.
   If you would like to do optional SRTP on outbound you will need to create
   a dialplan that dials with it enabled initially and if it fails fall back to
   without.

res_pjsip
------------------
 * Added endpoint configuration parameter "preferred_codec_only".
   This allow asterisk response to a SIP invite with the single most
   preferred codec rather than advertising all joint codec capabilities.
   This limits the other side's codec choice to exactly what we prefer.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 14.0.0 to Asterisk 14.1.0 ----------
------------------------------------------------------------------------------

Build System
------------------
 * The res_digium_phone, codec_g729a, codec_silk, codec_siren7 and
   codec_siren14 binary modules hosted at downloads.digium.com can now be
   automatically downloaded and installed during the Asterisk install
   process.  If selected in menuselect, when 'make install' is run, the
   script will check the downloads site for a new version and download
   and install it if needed.  The '--with-externals-cache' option to
   ./configure can be used to specify a location to cache the latest
   tarballs so they don't have to be re-downloaded for every install.

app_voicemail
------------------
 * Added "tps_queue_high" and "tps_queue_low" options.
   The options can modify the taskprocessor alert levels for this module.
   Additional information can be found in the sample configuration file at
   config/samples/voicemail.conf.sample.

res_pjsip_mwi
------------------
 * Added "mwi_tps_queue_high" and "mwi_tps_queue_low" global configuration
   options to tune taskprocessor alert levels.

 * Added "mwi_disable_initial_unsolicited" global configuration option
   to disable sending unsolicited MWI to all endpoints on startup.
   Additional information can be found in the sample configuration file at
   config/samples/pjsip.conf.sample.

chan_pjsip
------------------
 * A new dialplan function, PJSIP_SEND_SESSION_REFRESH, has been added. When
   invoked, a re-INVITE or UPDATE request will be sent immediately to the
   endpoint underlying the channel. When used in combination with the existing
   dialplan function PJSIP_MEDIA_OFFER, this allows the formats on a PJSIP
   channel to be re-negotiated and updated after session set up.

res_pjsip
------------------
 * A new endpoint configuration parameter 'contact_user' has been added which
   when set will override the default user set on Contact headers in outgoing
   requests.

 * If you are using a sorcery realtime backend to store global res_pjsip
   options (ps_globals table) then you now have to do a res_pjsip reload for
   changes to these options to take effect.  If you are using pjsip.conf to
   configure these options then you already had to do a reload after making
   changes.

 * Added "ignore_uri_user_options" global configuration option for
   compatibility with an ITSP that sends URI user field options.  When enabled
   the user field is truncated at the first semicolon.
   Example:
   URI: "sip:1235557890;phone-context=national@x.x.x.x;user=phone"
   The user field is "1235557890;phone-context=national"
   Which is truncated to this: "1235557890"

   Note: The caller-id and redirecting number strings obtained from incoming
   SIP URI user fields are now always truncated at the first semicolon.

app_confbridge
------------------
  * Some sounds played into the bridge are played asynchronously. This, for
    instance, allows a channel to immediately exit the ConfBridge without having
    to wait for a leave announcement to play.

app_dial
------------------
 * Added the "Q" option which sets the Q.850/Q.931 cause on unanswered channels
   when another channel answers the call.  The default of ANSWERED_ELSEWHERE
   is unchanged.

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

AMI
-----------------
 * A new event, "DialState" has been added. This is similar to "DialBegin" and
 "DialEnd" in that it tracks the state of a dialed call. The difference is that
 this indicates some intermediate state change in the dial attempt, such as
 "RINGING", "PROGRESS", or "PROCEEDING".

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.

 * Stored recordings now allow for the media associated with a stored recording
   to be retrieved. The new route, GET /recordings/stored/{name}/file, will
   transmit the raw media file to the requester as binary.


 * "Dial" events have been modified to not only be sent when dialing begins and ends.
 They now are also sent for intermediate states, such as "RINGING", "PROGRESS", and
 "PROCEEDING".

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.

Codecs
------------------
 * Added format attribute negotiation for the iLBC audio codec. Format attribute
   negotiation is provided by the res_format_attr_ilbc module. iLBC 20 is the
   default now. Falls back to iLBC 30, when the remote party requests this.

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.

FollowMe
------------------
 * It is now possible to disable the prompt from a callee by setting
   'enable_callee_prompt = no' in followme.conf.

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.

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'.

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.


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.

 * Plaintext auth mode is deprecated and removed from possible default modes.

chan_rtp (was chan_multicast_rtp)
------------------
 * Added unicast RTP support and renamed chan_multicast_rtp to chan_rtp.

 * The format for dialing a unicast RTP channel is:
   UnicastRTP/<destination-addr>[/[<options>]]
   Where <destination-addr> is something like '127.0.0.1:5060'.
   Where <options> are in standard Asterisk flag options format:
   c(<codec>) - Specify which codec/format to use such as 'ulaw'.
   e(<engine>) - Specify which RTP engine to use such as 'asterisk'.

 * New options were added for a multicast RTP channel.  The format for
   dialing a multicast RTP channel is:
   MulticastRTP/<type>/<destination-addr>[/[<control-addr>][/[<options>]]]
   Where <type> can be either 'basic' or 'linksys'.
   Where <destination-addr> is something like '224.0.0.3:5060'.
   Where <control-addr> is something like '127.0.0.1:5060'.
   Where <options> are in standard Asterisk flag options format:
   c(<codec>) - Specify which codec/format to use such as 'ulaw'.
   i(<address>) - Specify the interface address from which multicast RTP
     is sent.
   l(<enable>) - Set whether packets are looped back to the sender.  The
     enable value can be 0 to set looping to off and non-zero to set
     looping on.
   t(<ttl>) - Set the time-to-live (TTL) value for multicast packets.

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.

 * Session-Timers (RFC 4028) work for TCP (and TLS) transports as well now.
   Previously Asterisk dropped calls only with UDP transports. However with
   longer international calls via TCP, the SIP channel might break, because
   all hops on the Internet route must stay online (have not a single power
   outage, for example). Therefore with Session-Timers enabled (which are
   enabled at default), you might see additional dropped calls. Consequently
   please, consider to go for session-timers=refuse in your sip.conf.

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 device state 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 a device state hint will be automatically created
   with the name of the device.

* If Asterisk is built with systemd support, and run under systemd, it will
  notify systemd of its state using sd_notify. Use 'Type=notify' in
  asterisk.service.

Functions
------------------
 * The func_odbc global option "single_db_connection" default value has been
   changed to 'no'.


Formats
------------------
 * New module format_ogg_speex added which supports Speex codec inside
   Ogg containers (filename extension .spx).


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_info_empty
--------------------
 * A new module that can respond to empty Content-Type INFO packets during call.
   Some SBCs will terminate a call if their empty INFO packets are not responded
   to within a predefined time.

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'.

res_pjsip_outbound_publish
------------------
 * Added a new multi_user option that when set to 'yes' allows a given configuration
   to be used for multiple users.


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.

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'.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.10.0 to Asterisk 13.11.0 ----------
------------------------------------------------------------------------------

chan_dahdi
------------------
 * Added "faxdetect_timeout" option.
   The option determines how many seconds into a call before faxdetect
   is disabled for the call.  Setting the value to zero disables the timeout.

res_pjsip
------------------
 * Added "fax_detect_timeout" to endpoint.
   The option determines how many seconds into a call before fax_detect
   is disabled for the call.  Setting the value to zero disables the timeout.

 * Added "subscribe_context" to endpoint.
   If specified, incoming SUBSCRIBE requests will be searched for the matching
   extension in the indicated context. If no "subscribe_context" is specified,
   then the "context" setting is used.

res_rtp_asterisk
------------------
 * The DTLS part in Asterisk now supports Perfect Forward Secrecy (PFS).
   Enabling PFS is attempted by default, and is dependent on the configuration
   of the module using TLS.
   - Ephemeral ECDH (ECDHE) is enabled by default. To disable it, do not
     specify a ECDHE cipher suite in sip.conf, for example:
       dtlscipher=AES128-SHA
   - Ephemeral DH (DHE) is disabled by default. To enable it, add DH parameters
     into the private key file, e.g., sip.conf dtlsprivatekey. For example:
       openssl dhparam -out ./dh.pem 2048
   - Because clients expect the server to prefer PFS, and because OpenSSL sorts
     its cipher suites by bit strength, see "openssl ciphers -v DEFAULT".
     Consider re-ordering your cipher suites in the respective configuration
     file. For example:
       dtlscipher=ECDHE-ECDSA-AES128-GCM-SHA256:ECDHE-RSA-AES128-GCM-SHA256
     which forces PFS and requires at least DTLS 1.2.

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

Core
------------------
 * A channel variable FORWARDERNAME is now set which indicates which channel
   was responsible for a forwarding requests received on dial attempt.

func_odbc
------------------
 * Added new global option "single_db_connection".
   Enabling this option func_odbc will use a single database connection per DSN.
   This option is enabled by default.

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 "via_addr", "via_port", "call_id" to contacts.
   As res_pjsip_nat rewrites contact's address, only the last Via header
   can contain the source address of registered endpoint.
   Also Call-Id header may contain the source address of registered endpoint.
   Added new fields ViaAddress,CallID to AMI event ContactStatus

 * Endpoint IP Access Controls
   Added new configuration Endpoint options:
    "acl" - list of IP ACL section names in acl.conf
    "deny" - List of IP addresses to deny access from
    "permit" - List of IP addresses to permit access from
    "contact_acl" - List of Contact ACL section names in acl.conf
    "contact_deny" - List of Contact header addresses to deny
    "contact_permit" - List of Contact header addresses to permit

 * 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.

 * When starting Asterisk, received traffic will now be ignored until Asterisk
   has loaded all modules and is fully booted.

res_hep
------------------
 * Added a new option, 'uuid_type', that sets the preferred source of the Homer
   correlation UUID. The valid options are:
   - call-id: Use the PJSIP SIP Call-ID header value
   - channel: Use the Asterisk channel name
   The default value is 'call-id'. In the event that a HEP module cannot find a
   valid value using the specified 'uuid_type', the module may fallback to a
   more readily available source for the correlation UUID.

res_odbc
------------------
 * A new option has been added, 'max_connections', which sets the maximum number
   of concurrent connections to the database. This option defaults to 1 which
   returns the behavior to that of Asterisk 13.7 and prior.

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.

Codecs
------------------
 * Added the associated format name to 'core show codecs'.

res_ari_channels
------------------
 * Added 'formats' to channel create/originate to allow setting the allowed
   formats for a channel when no originator channel is available.  Especially
   useful for Local channel creation where no other format information is
   available.  'core show codecs' can now be used to look up suitable format
   names.

------------------------------------------------------------------------------
--- 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 ------------
------------------------------------------------------------------------------