UPGRADE.txt

===== WARNING, THIS FILE IS OBSOLETE AND WILL BE REMOVED IN A FUTURE VERSION =====
See 'Upgrade Notes' in the CHANGES file

===========================================================
===
=== THIS FILE IS AUTOMATICALLY GENERATED DURING THE RELEASE
=== PROCESS. DO NOT MAKE CHANGES HERE. INSTEAD, REFER TO
=== doc/UPGRADE-staging/README.md FOR MORE DETAILS.
===
=== Information for upgrading between Asterisk versions
===
=== This file documents all the changes that MUST be taken
=== into account when upgrading between certain Asterisk
=== versions. 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. This file also includes 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.
===
===========================================================

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 20.1.0 to Asterisk 20.2.0 ------------
------------------------------------------------------------------------------

app_playback
------------------
 * In Asterisk 11, if a channel was redirected away during Playback(),
   the PLAYBACKSTATUS variable would be set to SUCCESS. In Asterisk 12
   (specifically commit 7d9871b3940fa50e85039aef6a8fb9870a7615b9) that
   behavior was inadvertently changed and the same operation would result
   in the PLAYBACKSTATUS variable being set to FAILED. The Asterisk 11
   behavior has been restored.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 20.0.0 to Asterisk 20.1.0 ------------
------------------------------------------------------------------------------

AMI (Asterisk Manager Interface)
------------------
 * Previously, GetConfig and UpdateConfig were able to access files outside of
   the Asterisk configuration directory. Now this access is put behind the
   live_dangerously configuration option in asterisk.conf, which is disabled by
   default. If access to configuration files outside of the Asterisk configuation
   directory is required via AMI, then the live_dangerously configuration option
   must be set to yes.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 19.0.0 to Asterisk 20.0.0 ------------
------------------------------------------------------------------------------

res_crypto
------------------
 * In addition to only paying attention to files ending with .key or .pub
   in the keys directory, we now also ignore any files which aren't regular
   files.

------------------------------------------------------------------------------
--- New functionality introduced in Asterisk 20.0.0 --------------------------
------------------------------------------------------------------------------

res_monitor
------------------
 * This module is no longer built by default in
   accordance with the Module Deprecation Policy.
   If you require this functionality you will need
   to enable it for building in menuselect. Note
   that in the future res_monitor will be removed.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 19.0.0 to Asterisk 20.0.0 ------------
------------------------------------------------------------------------------

AMI
------------------
 * The XML Manager Event Interface (amxml) now generates attribute names
   that are compliant with the XML 1.1 specification. Previously, an
   attribute name that started with a digit would be rendered as-is, even
   though attribute names must not begin with a digit. We now prefix
   attribute names that start with a digit with an underscore ('_') to
   prevent XML validation failures.

STIR/SHAKEN
------------------
 * The STIR/SHAKEN configuration option has been split into
   4 different choices: off, attest, verify, and on. Off and
   on behave the same way as before. Attest will only perform
   attestation on the endpoint, and verify will only perform
   verification on the endpoint.

chan_iax2
------------------
 * Encryption is now supported for RSA authentication.

   Currently, these auth configurations will cause a crash:
   auth = md5,rsa
   auth = plaintext,md5,rsa

   With a patched peer, the following will cause a crash:
   auth = rsa
   auth = md5,rsa
   auth = plaintext,md5,rsa

   If both the peer and user are patches, no crash occurs.
   Existing good configurations should continue to work.

res_http_media_cache
------------------
 * When fetching a file for playback from a URL, Asterisk will now first
   use the value of the Content-Type header in the HTTP response to
   determine the format of the audio data, and only if it is unable to do
   that will it attempt to parse the URL and extract the extension from
   the path portion. Previously Asterisk would first look at the end of
   the URL, which may have included query string parameters or a URL
   fragment, which was error prone.

res_pjsip
------------------
 * The 'async_operations' setting on transports is no longer
   obeyed and instead is always set to 1. This is due to the
   functionality not being applicable to Asterisk and causing
   excess unnecessary memory usage. This setting will now be
   ignored but can also be removed from the configuration file.

------------------------------------------------------------------------------
--- New functionality introduced in Asterisk 19.0.0 --------------------------
------------------------------------------------------------------------------

Log Rotate
------------------
 * The sample logger files have been changed to have .log as their file
   extension. This was done so that when attached to issues on the issue
   tracker, they are able to be opened in the browser for convenience.
   Because of this, the asterisk.logrotate script has been updated to look
   for .log extensions instead of no extension for files such as full
   and messages.

chan_sip
------------------
 * chan_sip is no longer built by default. To build it, make sure to
   enable it when running 'make menuselect'

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 18.0.0 to Asterisk 19.0.0 ------------
------------------------------------------------------------------------------

STIR/SHAKEN
------------------
 * The configuration option public_key_url in stir_shaken.conf
   has been renamed to public_cert_url to better fit what it
   contains. Only the name has changed - functionality is the
   same.

 * STIR/SHAKEN originally needed an origid to be specified in
   stir_shaken.conf under the certificate config object in
   order to work. Now, one is automatically created by
   generating a UUID, as recommended by RFC8588. Any origid
   you have in your stir_shaken.conf will need to be removed
   for the module to read in certificates.

menuselect
------------------
 * menuselect --enable, --disable, --enable-category and --disable-category will
   now fail with a non-zero exit code instead of silently failing if an invalid
   option or category is specified.

res_srtp
------------------
 * SRTP replay protection has been added to res_srtp and
   a new configuration option "srtpreplayprotection" has
   been added to the rtp.conf config file.  For security
   reasons, the default setting is "yes".  Buggy clients
   may not handle this correctly which could result in
   no, or one way, audio and Asterisk error messages like
   "replay check failed".

------------------------------------------------------------------------------
--- New functionality introduced in Asterisk 18.0.0 --------------------------
------------------------------------------------------------------------------

Core
------------------
 * The ast_format_cap_from_stream_topology() function has been renamed
   to ast_stream_topology_get_formats().

app_bridgeaddchan
------------------
 * The BridgeAdd application now behaves more like the Bridge application.
   The application now sets the BRIDGERESULT channel variable to indicate
   what happened when the channel resumes in dialplan.  This is instead of
   hanging up the channel on failure conditions.

app_mixmonitor
------------------
 * In Asterisk 13.29, a new option flag was added to MixMonitor (the 'S'
   option) that when combined with the r() or t() options would inject
   silence into these files if audio was going to be written to one and
   not that other. This allowed the files specified by r() and t() to
   subsequently be mixed outside of Asterisk and be appropriately
   synchronized. This behavior is now the default, and a new option has
   been added to disable this behavior if desired (the 'n' option).

app_queue
------------------
 * The 'Reason' header in the QueueMemberPause AMI Event has been
   removed. The 'PausedReason' header should be used instead.

 * If they are not specified in [general], "shared_lastcall" and "autofill"
   now always default to OFF.  Before this version, they would be off ('no') if
   queues.conf did not have a [general] section, but on ('yes') if it did.

app_voicemail
------------------
 * The MessageExists dialplan application and the MESSAGE_EXISTS dialplan
   function were removed. The were deprecated in Asterisk 1.6.0 and
   Asterisk 11.0.0 respectively. The VM_INFO() dialplan function is the
   supported mechanism to query the status of a given mailbox.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 17.0.0 to Asterisk 18.0.0 ------------
------------------------------------------------------------------------------

AMI
------------------
 * The AMI Originate action, which optionally takes a dialplan application as
   an argument, no longer accepts "Originate" as the application due to
   security concerns.

ARI
------------------
 * The "TextMessageReceived" event used to include a list of "TextMessageVariable"
   objects as part of its output. Due to a couple of bugs in Asterisk a list of
   received variables was never included even if ones were available. However,
   variables set to send would be (which they should have not been), but would
   fail validation due to the bad formatting.

   So basically there was no way to get a "TextMessageReceived" event with
   variables. Due to this the API has changed. The "TextMessageVariable" object
   no longer exists. "TextMessageReceived" now returns a JSON object of key/value
   pairs. So for instance instead of a list of "TextMessageVariable" objects:

   [ TextMessageVariable, TextMessageVariable, TextMessageVariable]

   where a TextMessageVariable was supposed to be:

   { "key": "<var name>", "value":, "<var value>" }

   The output is now just:

   { "<var name>": "<var value>" }

   This aligns more with how variables are specified when sending a message, as
   well as other variable lists in ARI.

Core
------------------
 * The streams API function ast_stream_get_formats is
   now defined as returning the format capabilities const.
   This has always been the case but was never enforced
   through the API itself. Any consumer of this API that
   is not treating the formats as immutable should update
   their code to create a new format capabilities and set
   it on the stream instead.

res_stasis
------------------
 * The "TextMessageReceived" event used to include a list of "TextMessageVariable"
   objects as part of its output. Due to a couple of bugs in Asterisk a list of
   received variables was never included even if ones were available. However,
   variables set to send would be (which they should have not been), but would
   fail validation due to the bad formatting.

   So basically there was no way to get a "TextMessageReceived" event with
   variables. Due to this the API has changed. The "TextMessageVariable" object
   no longer exists. "TextMessageReceived" now returns a JSON object of key/value
   pairs. So for instance instead of a list of "TextMessageVariable" objects:

   [ TextMessageVariable, TextMessageVariable, TextMessageVariable]

   where a TextMessageVariable was supposed to be:

   { "key": "<var name>", "value":, "<var value>" }

   The output is now just:

   { "<var name>": "<var value>" }

   This aligns more with how variables are specified when sending a message, as
   well as other variable lists in ARI.

res_stir_shaken
------------------
 * A new directory has been added under the default (e.g., /var/lib/asterisk) -
   inside the 'keys' directory - named 'stir_shaken'. This directory will
   hold public keys that have been downloaded for STIR/SHAKEN verification.

------------------------------------------------------------------------------
--- New functionality introduced in Asterisk 17.0.0 --------------------------
------------------------------------------------------------------------------

Applications
------------------
 * The JabberStatus application, deprecated in Asterisk 12, has been removed.

Bridging
------------------
 * The bridging core no longer uses the stasis cache for bridge
   snapshots.  The latest bridge snapshot is now stored on the
   ast_bridge structure itself.

   The following APIs are no longer available since the stasis cache
   is no longer used:
     ast_bridge_topic_cached()
     ast_bridge_topic_all_cached()

   A topic pool is now used for individual bridge topics.

   The ast_bridge_cache() function was removed since there's no
   longer a separate container of snapshots.

   A new function "ast_bridges()" was created to retrieve the
   container of all bridges.  Users formerly calling
   ast_bridge_cache() can use the new function to iterate over
   bridges and retrieve the latest snapshot directly from the
   bridge.

   The ast_bridge_snapshot_get_latest() function was renamed to
   ast_bridge_get_snapshot_by_uniqueid().

   A new function "ast_bridge_get_snapshot()" was created to retrieve
   the bridge snapshot directly from the bridge structure.

   The ast_bridge_topic_all() function now returns a normal topic
   not a cached one so you can't use stasis cache functions on it
   either.

   The ast_bridge_snapshot_type() stasis message now has the
   ast_bridge_snapshot_update structure as it's data.  It contains
   the last snapshot and the new one.

Build
------------------
 * Asterisk headers are no longer installed and uninstalled automatically when
   performing a "make install" or a "make uninstall".  To install/uninstall the
   headers, use "make install-headers" and "make uninstall-headers".  The headers
   also continue to be uninstalled when performing a "make uninstall-all".

Channels
------------------
 * The core no longer uses the stasis cache for channels snapshots.
   The following APIs are no longer available:
       ast_channel_topic_cached()
       ast_channel_topic_all_cached()
   The ast_channel_cache_all() and ast_channel_cache_by_name() functions
   now returns an ao2_container of ast_channel_snapshots rather than a
   container of stasis_messages therefore you can't call stasis_cache
   functions on it.
   The ast_channel_topic_all() function now returns a normal topic,
   not a cached one so you can't use stasis cache functions on it either.
   The ast_channel_snapshot_type() stasis message now has the
   ast_channel_snapshot_update structure as it's data.
   ast_channel_snapshot_get_latest() still returns the latest snapshot.

chan_sip
------------------
 * The chan_sip module is now deprecated, users should migrate to the
   replacement module chan_pjsip.  See guides at the Asterisk Wiki:
     https://wiki.asterisk.org/wiki/x/tAHOAQ
     https://wiki.asterisk.org/wiki/x/hYCLAQ

func_callerid
------------------
 * The CALLERPRES() dialplan function, deprecated in Asterisk 1.8, has been
   removed.

res_parking
------------------
 * The PARKINGSLOT channel variable, deprecated in Asterisk 12 in favor of the
   PARKING_SPACE channel variable, will no longer be set.

res_xmpp
------------------
 * The JabberStatus application, deprecated in Asterisk 12, has been removed.

------------------------------------------------------------------------------
--- Functionality changes from Asterisk 16.0.0 to Asterisk 17.0.0 ------------
------------------------------------------------------------------------------

Core
------------------
 * res_pjsip_pubsub is now required so call transfer progress can be monitored
   and reported in the channel variable TRANSFERSTATUS.

app_voicemail.c
------------------
 * The "Voicemail Build Options" section of menuselect has been removed along with
   the FILE_STORAGE, ODBC_STORAGE and IMAP_STORAGE menuselect options.  All 3 variants
   of the voicemail app can now be built at the same by enabling app_voicemail,
   app_voicemail_imap, and app_voicemail_odbc under the "Applications" section.
   By default, only app_voicemail is enabled.  Also, the modules.conf sample has
   been updated to "noload" app_voicemail_imap and app_voicemail_odbc should they
   all be built.  Packagers must update their build scripts appropriately.

chan_pjsip
------------------
 * res_pjsip_pubsub is now required so call transfer progress can be monitored
   and reported in the channel variable TRANSFERSTATUS.

New in 16.0.0:

app_fax:
 - The app_fax module is now deprecated, users should migrate to the
   replacement module res_fax.

app_macro:
 - The app_macro module is now deprecated and by default it is no longer
   built.  Users should migrate to app_stack (Gosub).  A warning is logged
   the first time any Macro is used.

AMI:
 - The ContactStatus and Status fields for the manager events ContactStatus
   and ContactStatusDetail are now set to "NonQualified" when a contact exists
   but has not been qualified.
 - The ContactStatus event will no longer be sent by PJSIP when a device
   refreshes its registration.
 - The "Newexten" event is now part of the "dialplan" class. The documentation
   for Asterisk 15 already specified this, but the implementation was actually
   using the "call" class instead.

ARI:
 - The ContactInfo event's contact_status field is now set to "NonQualified"
   when a contact exists but has not been qualified.

Build System:
 - MALLOC_DEBUG no longer has an effect on Asterisk's ABI.  Asterisk built
   with MALLOC_DEBUG can now successfully load binary modules built without
   MALLOC_DEBUG and vice versa.  Third-party pre-compiled modules no longer
   need to have a special build with it enabled.

 - Asterisk now depends on libjansson >= 2.11.  If this version is not
   available on your distro you can use `./configure --with-jansson-bundled`.

chan_dahdi:
 - Timeouts for reading digits from analog phones are now configurable in
   chan_dahdi.conf: firstdigit_timeout, interdigit_timeout, matchdigit_timeout.

cdr_syslog:
 - The cdr_syslog module is now deprecated and by default it is no longer
   built.

res_config_sqlite:
 - The res_config_sqlite module is now deprecated, users should migrate to the
   replacement module res_config_sqlite3.

res_monitor:
 - The res_monitor module is now deprecated, users should migrate to the
   replacement module app_mixmonitor.

Core:
 - libedit is no longer available as an embedded library and must be provided
   by the system.
 - The module loader now enforces inter-module dependencies.  This ensures that
   a module is not started before another it depends on, even if preload is used.
   If a dependency is not available or fails to startup this will block any
   dependants from startup.
 - Parts of the Asterisk core which can load configuration from realtime are now
   built-in modules.  It is no longer necessary to preload realtime drivers as
   they are always initialized before the built-in modules.

From 15.2.0 to 15.3.0:

res_pjsip
------------------
 * Users who are matching endpoints by SIP header need to reevaluate their
   global "endpoint_identifier_order" option in light of the "ip" endpoint
   identifier method split into the "ip" and "header" endpoint identifier
   methods.

res_pjsip_endpoint_identifier_ip
------------------
 * The endpoint identifier "ip" method previously recognized endpoints either
   by IP address or a matching SIP header.  The "ip" endpoint identifier method
   is now split into the "ip" and "header" endpoint identifier methods.  The
   "ip" endpoint identifier method only matches by IP address and the "header"
   endpoint identifier method only matches by SIP header.  The split allows the
   user to control the relative priority of the IP address and the SIP header
   identification methods in the global "endpoint_identifier_order" option.
   e.g., If you have two type=identify sections where one matches by IP address
   for endpoint alice and the other matches by SIP header for endpoint bob then
   you can now predict which endpoint is matched when a request comes in that
   matches both.

New in 15.0.0:

Build System:
 - '--with-pjproject-bundled' is now the default when running ./configure
   It can be disabled with '--without-pjproject-bundled'.

Core:
 - Multi-stream support has been added so a channel can have multiple
   streams of the same type such as audio and video.

 - The 'Data Retrieval API' has been removed. This API was not actively
   maintained, was not added to new modules (such as res_pjsip), and there
   exist better alternatives to acquire the same information, such as the
   ARI. As a result, the 'DataGet' AMI action as well as the 'data get'
   CLI command have been removed.

From 14.6.0 to 14.7.0:

Core:
 - ast_app_parse_timelen now returns an error if it encounters extra characters
   at the end of the string to be parsed.

From 14.4.0 to 14.5.0:

Core:
 - Support for embedded modules has been removed.  This has not worked in
   many years.  LOADABLE_MODULES menuselect option is also removed as
   loadable module support is now always enabled.

From 14.3.0 to 14.4.0:

res_rtp_asterisk:
 - The RTP layer of Asterisk now has support for RFC 5761: "Multiplexing RTP
   Data and Control Packets on a Single Port." For the PJSIP channel driver,
   chan_pjsip, you can set "rtcp_mux = yes" on a PJSIP endpoint in pjsip.conf
   to enable the feature. For chan_sip you can set "rtcp_mux = yes" either
   globally or on a per-peer basis in sip.conf.

New in 14.0.0

ARI:
 - The policy for when to send "Dial" events has changed. Previously, "Dial"
   events were sent on the calling channel's topic. However, starting in Asterisk
   14, if there is no calling channel on which to send the event, the event is
   instead sent on the called channel's topic. Note that for the ARI channels
   resource's dial operation, this means that the "Dial" events will always be
   sent on the called channel's topic.

Channel Drivers:

chan_dahdi:
 - For users using the FXO port (FXS signaling) distinctive ring detection
   feature, you will need to adjust the dringX count values.  The count
   values now only record ring end events instead of any DAHDI event.  A
   ring-ring-ring pattern would exceed the pattern limits and stop
   Caller-ID detection.

chan_sip:
 - The SIP dial string has been extended past the [!dnid] option by another
   exclamation mark: [!dnid[!fromuri].  An exclamation mark in the To-URI
   will now mean changes to the From-URI.

Core:
 - The REF_DEBUG compiler flag is now used to enable refdebug by default.
   The setting can be overridden in asterisk.conf by setting refdebug in
   the options category.  No recompile is required to enable/disable it.

 - 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.

AMI:
 - The 'ModuleCheck' Action's Version key will no longer show the module
   version. The value will always be blank.

CLI:
 - The 'core show file version' command has been removed. When Asterisk
   moved to Git, the source control version support was removed. As a
   result, the CLi command was no longer useful and was removed as well.

Logging:
 - The first callid created is now 1 instead of 0.  The value 0
   is now reserved to represent a lack of callid.

AMI:
 - The Command action now sends the output from the CLI command as a series
   of Output headers for each line instead of as a block of text with the
   --END COMMAND-- delimiter to match the output from other actions.

   Commands that fail to execute (no such command, invalid syntax etc.) now
   return an Error response instead of Success.

app_amd:
 - The 'maximum_number_of_words' configuration option and parameter to the AMD
   application previously did not match the documented functionality + variable
   name.  In Asterisk 13, a value of '3' would mean that if '3' words were detected,
   the result would be detection as a 'MACHINE'.  As of this version, the value
   reflects the maximum words that if EXCEEDED (rather than reached), would
   result in detection as a machine.  This means that you should update this
   value to be one higher than your previos value, if your previous value
   was working well for you.

From 12 to 13:

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

 - 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.

 - The asterisk compatibility options in asterisk.conf have been removed.
   These options enabled certain backwards compatibility features for
   pbx_realtime, res_agi, and app_set that made their behaviour similar to
   Asterisk 1.4. Users who used these backwards compatibility settings should
   update their dialplans to use ',' instead of '|' as a delimiter, and should
   use the Set dialplan application instead of the MSet dialplan application.

Build System:
 - Sample config files have been moved from configs/ to a subfolder 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.

 - 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.

Applications:

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.

 - 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.

Queue:
 - Queue rules provided in queuerules.conf can no longer be named "general".

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.

CDR Backends:
 - The cdr_sqlite module was deprecated and has been removed. Users of this
   module should use the cdr_sqlite3_custom module instead.

Channel Drivers:

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

 - 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.
   Default is now no.

chan_gtalk
 - This module was deprecated and has been removed. Users of chan_gtalk
   should use chan_motif.

chan_h323
 - This module was deprecated and has been removed. Users of chan_h323
   should use chan_ooh323.

chan_jingle
 - This module was deprecated and has been removed. Users of chan_jingle
   should use chan_motif.

chan_pjsip:
 - 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:
 - Made set SIPREFERREDBYHDR as inheritable for better chan_pjsip
   interoperability.

 - The SIPPEER dialplan function no longer supports using a colon as a
   delimiter for parameters. The parameters for the function should be
   delimited using a comma.

 - The SIPCHANINFO dialplan function was deprecated and has been removed. Users
   of the function should use the CHANNEL function instead.

 - 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.

 - The 'progressinband=never' option is now more zealous in the persecution of
   progress messages coming from Asterisk. Channels bridged with a SIP channel
   that has 'progressinband=never' set will not be able to forward their
   progress indications through to the SIP device. chan_sip will now turn such
   progress indications into a 180 Ringing (if a 180 has not yet been
   transmitted) if 'progressinband=never'.

  - The codec preference order in an SDP during an offer is slightly different
    than previous releases. Prior to Asterisk 13, the preference order of
    codecs used to be:
    (1) Our preferred codec
    (2) Our configured codecs
    (3) Any non-audio joint codecs

    One of the ways the new media format architecture in Asterisk 13 improves
    performance is by reference counting formats, such that they can be reused
    in many places without additional allocation. To not require a large
    amount of locking, an instance of a format is immutable by convention.
    This works well except for formats with attributes. Since a media format
    with an attribute is a different object than the same format without an
    attribute, we have to carry over the formats with attributes from an
    inbound offer so that the correct attributes are offered in an outgoing
    INVITE request. This requires some subtle tweaks to the preference order
    to ensure that the media format with attributes is offered to a remote
    peer, as opposed to the same media format (but without attributes) that
    may be stored in the peer object.

    All of this means that our offer offer list will now be:
    (1) Our preferred codec
    (2) Any joint codecs offered by the inbound offer
    (3) All other codecs that are not the preferred codec and not a joint
        codec offered by the inbound offer

chan_unistim:
 - The unistim.conf 'dateformat' has changed meaning of options values to conform
   values used inside Unistim protocol

 - Added 'dtmf_duration' option with changing default operation to disable
   received dtmf playback on unistim phone

Core:

Account Codes:
 - accountcode behavior changed somewhat to add functional peeraccount
   support.  The main change is that local channels now cross accountcode
   and peeraccount across the special bridge between the ;1 and ;2 channels
   just like channels between normal bridges.  See the CHANGES file for
   more information.

ARI:
 - The ARI version has been changed to 1.5.0. This is to reflect backwards
   compatible changes made since 12.0.0 was released.

 - 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.

 - 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.

AMI:
 - The AMI version has been changed to 2.5.0. This is to reflect backwards
   compatible changes made since 12.0.0 was released.

 - 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

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

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

 - The response to the PresenceState AMI action has historically contained two
   Message keys. The first of these is used as an informative message regarding
   the success/failure of the action; the second contains a Presence state
   specific message. Having two keys with the same unique name in an AMI
   message is cumbersome for some client; hence, the Presence specific Message
   has been deprecated. The message will now contain a PresenceMessage key
   for the presence specific information; the Message key containing presence
   information will be removed in the next major version of AMI.

 - The manager.conf 'eventfilter' now takes an "extended" regular expression
   instead of a "basic" one.

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.

Logging:
 - 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.

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

 - 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.

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', 'user_agent', 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
   'message_context' was added to let users configure how MESSAGE requests are
   routed to the dialplan.

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

 - 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.

 - A new upgrade script has been added that adds a 'queue_rules' table for
   app_queue. Users of app_queue can store queue rules in a database. It is
   important to note that app_queue only looks for this table on module load or
   module reload; for more information, see the CHANGES file.

Resources:

res_odbc:
- The compatibility setting, allow_empty_string_in_nontext, has been removed.
  Empty column values will be stored as empty strings during realtime updates.

res_jabber:
 - This module was deprecated and has been removed. Users of this module should
   use res_xmpp instead.

res_http_websocket:
 - Added a compatibility option to ari.conf, sip.conf, and pjsip.conf
   '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.
Scripts:

safe_asterisk:
 - 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)"

Utilities:
 - The refcounter program has been removed in favor of the refcounter.py script
   in contrib/scripts.

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.