Update CHANGES and UPGRADE.txt for 20.0.0

CHANGES

@@ -12,6 +12,500 @@
 ===
 ==============================================================================
 
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 19.0.0 to Asterisk 20.0.0 ------------
+------------------------------------------------------------------------------
+
+Applications
+------------------
+ * added support for Danish syntax, playing the correct plural sound file
+   dependen on where you have 1 or multipe messages
+   based on the existing SE/NO code
+
+ * added that we set DIALEDPEERNUMBER on the outgoing channels
+   so it is avalible in b(content^extension^line)
+   this add the same behaviour as Dial
+
+Channel-agnostic MF support
+------------------
+ * A SendMF application and PlayMF manager
+   application are now included to send
+   arbitrary standard R1 MF tones on the
+   current channel or another specified channel.
+
+Core
+------------------
+ * Bundled PJProject Build
+
+   The build process has been updated to make pjproject troubleshooting
+   and development easier. See third-party/pjproject/README-hacking.md or
+   https://wiki.asterisk.org/wiki/display/AST/Bundled+PJProject
+   for more info.
+
+Handle non-standard Meter metric type safely
+------------------
+ * A meter_support flag has been introduced that defaults to true to maintain current behaviour.
+   If disabled, a counter metric type will be used instead wherever a meter metric type was used,
+   the counter will have a "_meter" suffix appended to the metric name.
+
+MessageSend
+------------------
+ * The MessageSend AMI action has been updated to allow the Destination
+   and the To addresses to be provided separately. This brings the
+   MessageSend manager command in line with the capabilities of the
+   MessageSend dialplan application.
+
+ToneScan application
+------------------
+ * A new application, ToneScan, allows for
+   synchronous detection of call progress
+   signals such as dial tone, busy tone,
+   Special Information Tones, and modems.
+
+ami
+------------------
+ * An AMI event now exists for "Wink".
+
+ * AMI events can now be globally disabled using
+   the disabledevents [general] setting.
+
+app_confbridge
+------------------
+ * Added the hear_own_join_sound option to the confbridge user profile to
+   control who hears the sound_join audio file. When set to 'yes' the user
+   entering the conference and the participants already in the conference
+   will hear the sound_join audio file. When set to 'no' the user entering
+   the conference will not hear the sound_join audio file, but the
+   participants already in the conference will hear the sound_join audio file.
+
+ * Adds the CONFBRIDGE_CHANNELS function which can
+   be used to retrieve a list of channels in a ConfBridge,
+   optionally filtered by a particular category. This
+   list can then be used with functions like SHIFT, POP,
+   UNSHIFT, etc.
+
+app_dtmfstore
+------------------
+ * New application which collects digits
+   dialed and stores them into
+   a specified variable.
+
+app_mf
+------------------
+ * Adds MF receiver and sender applications to support
+   the R1 MF signaling protocol, including integration
+   with the Dial application.
+
+ * Adds an option to ReceiveMF to cap the
+   number of digits read at a user-specified
+   maximum.
+
+app_milliwatt
+------------------
+ * The Milliwatt application's existing behavior is
+   incorrect in that it plays a constant tone, which
+   is not how digital milliwatt test lines actually
+   work.
+
+   An option is added so that a proper milliwatt test
+   tone can be provided, including a 1 second silent
+   interval every 10 seconds. However, for compatability
+   reasons, the default behavior remains unchanged.
+
+app_morsecode
+------------------
+ * Extends the Morsecode application by adding support for
+   American Morse code and adds a configurable option
+   for the frequency used in off intervals.
+
+app_originate
+------------------
+ * Codecs can now be specified for dialplan-originated
+   calls, as with call files and the manager action.
+   By default, only the slin codec is now used, instead
+   of all the slin* codecs.
+
+app_playback
+------------------
+ * A new option 'mix' is added to the Playback application that 
+   will play by filename and say.conf. It will look on the format of the 
+   name, if it is like say format it will play with say.conf if not it 
+   will play the file name.
+
+app_queue
+------------------
+ * Reload behavior in app_queue has been changed so
+   queue and agent stats are not reset during full
+   app_queue module reloads. The queue reset stats
+   CLI command may still be used to reset stats while
+   Asterisk is running.
+
+ * Add field to save the time value when a member enter a queue.
+   Shows this time in seconds using 'queue show' command and the
+   field LoginTime for responses for AMI the events.
+
+   The output for the CLI command `queue show` is changed by added a
+   extra data field for the information of the time login time for each
+   member.
+
+ * added that we set DIALEDPEERNUMBER on the outgoing channels
+   so it is avalible in b(content^extension^line)
+   this add the same behaviour as Dial
+
+ * Load queues and members from Realtime for
+   AMI actions: QueuePause, QueueStatus and QueueSummary,
+   Applications: PauseQueueMember and UnpauseQueueMember.
+
+ * Added a new AMI action: QueueWithdrawCaller
+   This AMI action makes it possible to withdraw a caller from a queue
+   back to the dialplan. The call will be signaled to leave the queue
+   whenever it can, hence, it not guaranteed that the call will leave
+   the queue.
+
+   Optional custom data can be passed in the request, in the WithdrawInfo
+   parameter. If the call successfully withdrawn the queue,
+   it can be retrieved using the QUEUE_WITHDRAW_INFO variable.
+
+   This can be useful for certain uses, such as dispatching the call
+   to a specific extension.
+
+ * The m option now allows an override music on hold
+   class to be specified for the Queue application
+   within the dialplan.
+
+app_queue.c
+------------------
+ * Allow multiple files to be streamed for agent announcement.
+
+app_queues
+------------------
+ * adding support for playing the correct en/et for nordic languages
+
+ * Don't play sound_thanks if there is no leading hold_time message
+   When the only announcement is hold time, and there is no hold time (0 min, 0 sec), asterisk will say "thank you for your patience"
+
+app_read
+------------------
+ * A new option allows the digit '#' to be read literally,
+   rather than used exclusively as the input terminator
+   character.
+
+app_sendtext
+------------------
+ * A ReceiveText application has been added that can be
+   used in conjunction with the SendText application.
+
+app_voicemail
+------------------
+ * Add a new 'S' option to VoiceMail which prevents the instructions
+   (vm-intro) from being played if a busy/unavailable/temporary greeting
+   from the voicemail user is played. This is similar to the existing 's'
+   option except that instructions will still be played if no user
+   greeting is available.
+
+ * added support for Danish syntax, playing the correct plural sound file
+   dependen on where you have 1 or multipe messages
+   based on the existing SE/NO code
+
+ * The r option has been added, which prevents deletion
+   of messages from VoiceMailMain, which can be
+   useful for shared mailboxes.
+
+apps
+------------------
+ * A new option 'mix' is added to the Playback application that 
+   will play by filename and say.conf. It will look on the format of the 
+   name, if it is like say format it will play with say.conf if not it 
+   will play the file name.
+
+ari
+------------------
+ * Expose channel driver's unique id (which is the Call-ID for SIP/PJSIP)
+   to ARI channel resources as 'protocol_id'.
+
+   ASTERISK-30027
+
+ast_coredumper
+------------------
+ * New options:
+    --pid=<asterisk_pid>
+      Allows specification of an Asterisk instance when trying to
+      and the script can't determine it itself.
+    --libdir=<system library directory>
+      Allows specification of a non-standard installation directory
+      containing the Asterisk modules.
+    --(no-)rename
+      Renames the coredump and the output files with readable
+      timestamps. This is the default.
+   Removed unneeded or confusing options:
+    --append-coredumps
+    --conffile
+    --no-default-search
+    --tarball-uniqueid
+   Changed Variables:
+    COREDUMPS is now just "/tmp/core!(*.txt)"
+    DATEFORMAT is renamed to DATEOPTS and defaults to '-u +%FT%H-%M-%SZ'
+   Changed behavior:
+    If you use 'running' or 'RUNNING' you no longer need to specify
+    '--no-default-search' to ignore existing coredumps.
+
+cdr
+------------------
+ * A new CDR option, channeldefaultenabled, allows controlling
+   whether CDR is enabled or disabled by default on
+   newly created channels. The default behavior remains
+   unchanged from previous versions of Asterisk (new
+   channels will have CDR enabled, as long as CDR is
+   enabled globally).
+
+chan_dahdi
+------------------
+ * Previously, cadences were appended on dahdi restart,
+   rather than reloaded. This prevented cadences from
+   being updated and maxed out the available cadences
+   if reloaded multiple times. This behavior is fixed
+   so that reloading cadences is idempotent and cadences
+   can actually be reloaded.
+
+ * A POLARITY function is now available that allows
+   getting or setting the polarity on a channel
+   from the dialplan.
+
+chan_iax2
+------------------
+ * ANI2 (OLI) is now transmitted over IAX2 calls
+   as an information element.
+
+ * Both a secret and an outkey may be specified at dial time,
+   since encryption is possible with RSA authentication.
+
+chan_pjsip
+------------------
+ * Add function PJSIP_HEADERS() to get list of headers by pattern in the same way as SIP_HEADERS() do.
+
+   Add ability to read header by pattern using PJSIP_HEADER().
+
+ * added global config option "allow_sending_180_after_183"
+
+   Allow Asterisk to send 180 Ringing to an endpoint
+   after 183 Session Progress has been send.
+   If disabled Asterisk will instead send only a
+   183 Session Progress to the endpoint.
+
+ * Hook flash events can now be sent on a PJSIP channel
+   if requested to do so.
+
+chan_sip
+------------------
+ * Session timers get removed on UPDATE
+   Fix if Asterisk receives a SIP REFER with Session-Timers UAC
+   that Asterisk maintains Session-Timers when sending UPDATE request
+
+chan_sip.c
+------------------
+ * resolve issue with pickup on device that uses "183" and not "180"
+
+channel_internal_api
+------------------
+ * CHANNEL(lastcontext) and CHANNEL(lastexten)
+   are now available for use in the dialplan.
+
+cli
+------------------
+ * The "module refresh" command has been added,
+   which allows unloading and then loading a
+   module with a single command.
+
+ * A new CLI command 'dialplan eval function' has been
+   added which allows users to test the behavior of
+   dialplan function calls directly from the CLI.
+
+func_channel
+------------------
+ * Adds the CHANNEL_EXISTS function to check for the existence
+   of a channel by name or unique ID.
+
+func_db
+------------------
+ * The function DB_KEYCOUNT has been added, which
+   returns the cardinality of the keys at a specified
+   prefix in AstDB, i.e. the number of keys at a
+   given prefix.
+
+func_env.c
+------------------
+ * Two new functions, DIRNAME and BASENAME, are now
+   included which allow users to obtain the directory
+   or the base filename of any file.
+
+func_evalexten
+------------------
+ * This adds the EVAL_EXTEN function which may be
+   used to evaluate data at dialplan extensions.
+
+func_framedrop
+------------------
+ * New function to selectively drop specified frames
+   in either direction on a channel.
+
+func_json
+------------------
+ * The JSON_DECODE dialplan function can now be used
+   to parse JSON strings, such as in conjunction with
+   CURL for using API responses.
+
+func_odbc
+------------------
+ * A SQL_ESC_BACKSLASHES dialplan function has been added which
+   escapes backslashes. Usage of this is dependent on whether the
+   database in use can use backslashes to escape ticks or not. If
+   it can, then usage of this prevents a broken SQL query depending
+   on how the SQL query is constructed.
+
+func_scramble
+------------------
+ * Adds an audio scrambler function that may be used to
+   distort voice audio on a channel as a privacy
+   enhancement.
+
+func_strings
+------------------
+ * A new STRBETWEEN function is now included which
+   allows a substring to be inserted between characters
+   in a string. This is particularly useful for transforming
+   dial strings, such as adding pauses between digits
+   for a string of digits that are sent to another channel.
+
+func_vmcount
+------------------
+ * Multiple mailboxes may now be specified instead of just one.
+
+logger
+------------------
+ * Added the ability to define custom log levels in logger.conf
+   and use them in the Log dialplan application. Also adds a
+   logger show levels CLI command.
+
+res_agi
+------------------
+ * Agi command 'exec' can now be enabled
+   to evaluate dialplan functions and variables
+   by setting the variable AGIEXECFULL to yes.
+
+res_cliexec
+------------------
+ * A new CLI command, dialplan exec application, has
+   been added which allows dialplan applications to be
+   executed at the CLI, useful for some quick testing
+   without needing to write dialplan.
+
+res_fax_spandsp
+------------------
+ * Adds support for spandsp 3.0.0.
+
+res_geolocation
+------------------
+ * Added res_geolocation which creates the core capabilities
+   to manipulate Geolocation information on SIP INVITEs.
+
+res_parking
+------------------
+ * An m option to Park and ParkAndAnnounce now allows
+   specifying a music on hold class override.
+
+res_pjproject
+------------------
+ * In pjproject.conf you can now map pjproject log levels
+   to the Asterisk TRACE log level.  The default mappings
+   have therefore changed so that only pjproject levels
+   3 and 4 are mapped to DEBUG and 5 and 6 are now mapped
+   to TRACE.  Previously 3, 4, 5, and 6 were all mapped to
+   DEBUG.
+
+res_pjsip
+------------------
+ * A new transport option 'allow_wildcard_certs' has been added that when it
+   and 'verify_server' are both set to 'yes', enables verification against
+   wildcards, i.e. '*.' in certs for common, and subject alt names of type DNS
+   for TLS transport types. Names must start with the wildcard. Partial wildcards,
+   e.g. 'f*.example.com' and 'foo.*.com' are not allowed. As well, names only
+   match against a single level meaning '*.example.com' matches 'foo.example.com',
+   but not 'foo.bar.example.com'.
+
+res_pjsip_geolocation
+------------------
+ * Added res_pjsip_geolocation which gives chan_pjsip
+   the ability to use the core geolocation capabilities.
+
+res_pjsip_header_funcs
+------------------
+ * Add function PJSIP_RESPONSE_HEADERS() to get list of header names from 200 response, in the same way as PJSIP_HEADERS() from the request.
+
+   Add function PJSIP_RESPONSE_HEADER() to read header from 200 response, in the same way as PJSIP_HEADER() from the request.
+
+res_pjsip_pubsub
+------------------
+ * A new resource_list option, resource_display_name, indicates
+   whether display name of resource or the resource name being
+   provided for RLS entries.
+   If this option is enabled, the Display Name will be provided.
+   This option is disabled by default to remain the previous behavior.
+   If the 'event' set to 'presence' or 'dialog' the non-empty HINT name
+   will be set as the Display Name.
+   The 'message-summary' is not supported yet.
+
+ * The Resource List Subscriptions (RLS) is dynamic now.
+   The asterisk now updates current subscriptions to reflect the changes
+   to the list on subscription refresh. If list items are added,
+   removed, updated or do not exist anymore, the asterisk regenerates
+   the resource list.
+
+res_pjsip_registrar
+------------------
+ * Adds new PJSIP AOR option remove_unavailable to either
+   remove unavailable contacts when a REGISTER exceeds
+   max_contacts when remove_existing is disabled, or
+   prioritize unavailable contacts over other existing
+   contacts when remove_existing is enabled.
+
+res_pjsip_t38
+------------------
+ * In res_pjsip_sdp_rtp, the bind_rtp_to_media_address option and the
+   fallback use of the transport's bind address solve problems sending
+   media on systems that cannot send ipv4 packets on ipv6 sockets, and
+   certain other situations. This change extends both of these behaviors
+   to UDPTL sessions as well in res_pjsip_t38, to fix fax-specific
+   problems on these systems, introducing a new option
+   endpoint/t38_bind_udptl_to_media_address.
+
+res_rtp_asterisk
+------------------
+ * When the address of the STUN server (stunaddr) is a name resolved via DNS, the
+   stunaddr will be recurringly resolved when the DNS answer Time-To-Live (TTL)
+   expires. This allows the STUN server to change its IP address without having to
+   reload the res_rtp_asterisk module.
+
+res_tonedetect
+------------------
+ * Arbitrary tone detection is now available through a
+   WaitForTone application (blocking) and a TONE_DETECT
+   function (non-blocking).
+
+say.c
+------------------
+ * Adds SAYFILES function to retrieve the file names that would
+   be played by corresponding Say applications, such as
+   SayDigits, SayAlpha, etc.
+
+   Additionally adds SayMoney and SayOrdinal applications.
+
+stasis_channels
+------------------
+ * Expose channel driver's unique id (which is the Call-ID for SIP/PJSIP)
+   to ARI channel resources as 'protocol_id'.
+
+   ASTERISK-30027
+
 ------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 18.0.0 to Asterisk 19.0.0 ------------
 ------------------------------------------------------------------------------

UPGRADE.txt

@@ -18,6 +18,73 @@
 ===
 ===========================================================
 
+------------------------------------------------------------------------------
+--- 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 --------------------------
 ------------------------------------------------------------------------------

doc/CHANGES-staging/add_mix_option_to_playback.txt

@@ -1,7 +0,0 @@
-Subject: app_playback
-Subject: apps
-
-A new option 'mix' is added to the Playback application that 
-will play by filename and say.conf. It will look on the format of the 
-name, if it is like say format it will play with say.conf if not it 
-will play the file name.

doc/CHANGES-staging/allow_wildcard_certs.txt

@@ -1,9 +0,0 @@
-Subject: res_pjsip
-
-A new transport option 'allow_wildcard_certs' has been added that when it
-and 'verify_server' are both set to 'yes', enables verification against
-wildcards, i.e. '*.' in certs for common, and subject alt names of type DNS
-for TLS transport types. Names must start with the wildcard. Partial wildcards,
-e.g. 'f*.example.com' and 'foo.*.com' are not allowed. As well, names only
-match against a single level meaning '*.example.com' matches 'foo.example.com',
-but not 'foo.bar.example.com'.

doc/CHANGES-staging/ami_wink.txt

@@ -1,3 +0,0 @@
-Subject: ami
-
-An AMI event now exists for "Wink".

doc/CHANGES-staging/app_confbridge_channels.txt

@@ -1,7 +0,0 @@
-Subject: app_confbridge
-
-Adds the CONFBRIDGE_CHANNELS function which can
-be used to retrieve a list of channels in a ConfBridge,
-optionally filtered by a particular category. This
-list can then be used with functions like SHIFT, POP,
-UNSHIFT, etc.

doc/CHANGES-staging/app_confbridge_hear_join.txt

@@ -1,8 +0,0 @@
-Subject: app_confbridge
-
-Added the hear_own_join_sound option to the confbridge user profile to
-control who hears the sound_join audio file. When set to 'yes' the user
-entering the conference and the participants already in the conference
-will hear the sound_join audio file. When set to 'no' the user entering
-the conference will not hear the sound_join audio file, but the
-participants already in the conference will hear the sound_join audio file.

doc/CHANGES-staging/app_dtmfstore.txt

@@ -1,6 +0,0 @@
-Subject: app_dtmfstore
-
-New application which collects digits
-dialed and stores them into
-a specified variable.
-

doc/CHANGES-staging/app_mf_maxdigits.txt

@@ -1,5 +0,0 @@
-Subject: app_mf
-
-Adds an option to ReceiveMF to cap the
-number of digits read at a user-specified
-maximum.

doc/CHANGES-staging/app_mf_mf.txt

@@ -1,5 +0,0 @@
-Subject: app_mf
-
-Adds MF receiver and sender applications to support
-the R1 MF signaling protocol, including integration
-with the Dial application.

doc/CHANGES-staging/app_milliwatt.txt

@@ -1,11 +0,0 @@
-Subject: app_milliwatt
-
-The Milliwatt application's existing behavior is
-incorrect in that it plays a constant tone, which
-is not how digital milliwatt test lines actually
-work.
-
-An option is added so that a proper milliwatt test
-tone can be provided, including a 1 second silent
-interval every 10 seconds. However, for compatability
-reasons, the default behavior remains unchanged.

doc/CHANGES-staging/app_morsecode.txt

@@ -1,6 +0,0 @@
-Subject: app_morsecode
-
-Extends the Morsecode application by adding support for
-American Morse code and adds a configurable option
-for the frequency used in off intervals.
-

doc/CHANGES-staging/app_originate_codecs.txt

@@ -1,6 +0,0 @@
-Subject: app_originate
-
-Codecs can now be specified for dialplan-originated
-calls, as with call files and the manager action.
-By default, only the slin codec is now used, instead
-of all the slin* codecs.

doc/CHANGES-staging/app_queue.txt

@@ -1,4 +0,0 @@
-Subject: app_queue.c
-
-Allow multiple files to be streamed for agent announcement.
-

doc/CHANGES-staging/app_queue_DIALEDPEERNUMBER.txt

@@ -1,6 +0,0 @@
-Subject: app_queue
-Subject: Applications
-
-added that we set DIALEDPEERNUMBER on the outgoing channels
-so it is avalible in b(content^extension^line)
-this add the same behaviour as Dial

doc/CHANGES-staging/app_queue_logintime.txt

@@ -1,9 +0,0 @@
-Subject: app_queue
-
-Add field to save the time value when a member enter a queue.
-Shows this time in seconds using 'queue show' command and the
-field LoginTime for responses for AMI the events.
-
-The output for the CLI command `queue show` is changed by added a
-extra data field for the information of the time login time for each
-member.

doc/CHANGES-staging/app_queue_music.txt

@@ -1,5 +0,0 @@
-Subject: app_queue
-
-The m option now allows an override music on hold
-class to be specified for the Queue application
-within the dialplan.

doc/CHANGES-staging/app_queue_nordic_language.txt

@@ -1,3 +0,0 @@
-Subject: app_queues
-
-adding support for playing the correct en/et for nordic languages

doc/CHANGES-staging/app_queue_say_thanks.txt

@@ -1,4 +0,0 @@
-Subject: app_queues
-
-Don't play sound_thanks if there is no leading hold_time message
-When the only announcement is hold time, and there is no hold time (0 min, 0 sec), asterisk will say "thank you for your patience"

doc/CHANGES-staging/app_queue_stats.txt

@@ -1,7 +0,0 @@
-Subject: app_queue
-
-Reload behavior in app_queue has been changed so
-queue and agent stats are not reset during full
-app_queue module reloads. The queue reset stats
-CLI command may still be used to reset stats while
-Asterisk is running.

doc/CHANGES-staging/app_read.txt

@@ -1,5 +0,0 @@
-Subject: app_read
-
-A new option allows the digit '#' to be read literally,
-rather than used exclusively as the input terminator
-character.

doc/CHANGES-staging/app_sendtext.txt

@@ -1,4 +0,0 @@
-Subject: app_sendtext
-
-A ReceiveText application has been added that can be
-used in conjunction with the SendText application.

doc/CHANGES-staging/app_voicemail.txt

@@ -1,7 +0,0 @@
-Subject: app_voicemail
-
-Add a new 'S' option to VoiceMail which prevents the instructions
-(vm-intro) from being played if a busy/unavailable/temporary greeting
-from the voicemail user is played. This is similar to the existing 's'
-option except that instructions will still be played if no user
-greeting is available.

doc/CHANGES-staging/app_voicemail_danish_syntax.txt

@@ -1,6 +0,0 @@
-Subject: app_voicemail
-Subject: Applications
-
-added support for Danish syntax, playing the correct plural sound file
-dependen on where you have 1 or multipe messages
-based on the existing SE/NO code

doc/CHANGES-staging/app_voicemail_nodelete.txt

@@ -1,5 +0,0 @@
-Subject: app_voicemail
-
-The r option has been added, which prevents deletion
-of messages from VoiceMailMain, which can be
-useful for shared mailboxes.

doc/CHANGES-staging/ari_add_pvt_id_to_channel_resource.txt

@@ -1,7 +0,0 @@
-Subject: ari
-Subject: stasis_channels
-
-Expose channel driver's unique id (which is the Call-ID for SIP/PJSIP)
-to ARI channel resources as 'protocol_id'.
-
-ASTERISK-30027

doc/CHANGES-staging/ast_coredumper.txt

@@ -1,23 +0,0 @@
-Subject: ast_coredumper
-
-New options:
- --pid=<asterisk_pid>
-   Allows specification of an Asterisk instance when trying to
-   and the script can't determine it itself.
- --libdir=<system library directory>
-   Allows specification of a non-standard installation directory
-   containing the Asterisk modules.
- --(no-)rename
-   Renames the coredump and the output files with readable
-   timestamps. This is the default.
-Removed unneeded or confusing options:
- --append-coredumps
- --conffile
- --no-default-search
- --tarball-uniqueid
-Changed Variables:
- COREDUMPS is now just "/tmp/core!(*.txt)"
- DATEFORMAT is renamed to DATEOPTS and defaults to '-u +%FT%H-%M-%SZ'
-Changed behavior:
- If you use 'running' or 'RUNNING' you no longer need to specify
- '--no-default-search' to ignore existing coredumps.

doc/CHANGES-staging/bundled-pjproject-build.txt

@@ -1,8 +0,0 @@
-Subject: Core
-
-Bundled PJProject Build
-
-The build process has been updated to make pjproject troubleshooting
-and development easier. See third-party/pjproject/README-hacking.md or
-https://wiki.asterisk.org/wiki/display/AST/Bundled+PJProject
-for more info.

doc/CHANGES-staging/cdr_disable.txt

@@ -1,8 +0,0 @@
-Subject: cdr
-
-A new CDR option, channeldefaultenabled, allows controlling
-whether CDR is enabled or disabled by default on
-newly created channels. The default behavior remains
-unchanged from previous versions of Asterisk (new
-channels will have CDR enabled, as long as CDR is
-enabled globally).

doc/CHANGES-staging/chan_dahdi_cadences.txt

@@ -1,8 +0,0 @@
-Subject: chan_dahdi
-
-Previously, cadences were appended on dahdi restart,
-rather than reloaded. This prevented cadences from
-being updated and maxed out the available cadences
-if reloaded multiple times. This behavior is fixed
-so that reloading cadences is idempotent and cadences
-can actually be reloaded.

doc/CHANGES-staging/chan_dahdi_polarity.txt

@@ -1,5 +0,0 @@
-Subject: chan_dahdi
-
-A POLARITY function is now available that allows
-getting or setting the polarity on a channel
-from the dialplan.

doc/CHANGES-staging/chan_iax2_ani2.txt

@@ -1,4 +0,0 @@
-Subject: chan_iax2
-
-ANI2 (OLI) is now transmitted over IAX2 calls
-as an information element.

doc/CHANGES-staging/chan_iax2_dial.txt

@@ -1,4 +0,0 @@
-Subject: chan_iax2
-
-Both a secret and an outkey may be specified at dial time,
-since encryption is possible with RSA authentication.

doc/CHANGES-staging/chan_pjsip_180_sdp.txt

@@ -1,8 +0,0 @@
-Subject: chan_pjsip
-
-added global config option "allow_sending_180_after_183"
-
-Allow Asterisk to send 180 Ringing to an endpoint
-after 183 Session Progress has been send.
-If disabled Asterisk will instead send only a
-183 Session Progress to the endpoint.

doc/CHANGES-staging/chan_pjsip_flash.txt

@@ -1,4 +0,0 @@
-Subject: chan_pjsip
-
-Hook flash events can now be sent on a PJSIP channel
-if requested to do so.

doc/CHANGES-staging/chan_sip_pickup_AST_STATE_DOWN.txt

@@ -1,3 +0,0 @@
-Subject: chan_sip.c
-
-resolve issue with pickup on device that uses "183" and not "180"

doc/CHANGES-staging/chan_sip_session-timer_on_update.txt

@@ -1,6 +0,0 @@
-Subject: chan_sip
-
-Session timers get removed on UPDATE
-Fix if Asterisk receives a SIP REFER with Session-Timers UAC
-that Asterisk maintains Session-Timers when sending UPDATE request
-

doc/CHANGES-staging/channel_internal_api.txt

@@ -1,4 +0,0 @@
-Subject: channel_internal_api
-
-CHANNEL(lastcontext) and CHANNEL(lastexten)
-are now available for use in the dialplan.

doc/CHANGES-staging/cli_eval_function.txt

@@ -1,5 +0,0 @@
-Subject: cli
-
-A new CLI command 'dialplan eval function' has been
-added which allows users to test the behavior of
-dialplan function calls directly from the CLI.

doc/CHANGES-staging/cli_refresh.txt

@@ -1,5 +0,0 @@
-Subject: cli
-
-The "module refresh" command has been added,
-which allows unloading and then loading a
-module with a single command.

doc/CHANGES-staging/func_channel.txt

@@ -1,4 +0,0 @@
-Subject: func_channel
-
-Adds the CHANNEL_EXISTS function to check for the existence
-of a channel by name or unique ID.

doc/CHANGES-staging/func_db.txt

@@ -1,6 +0,0 @@
-Subject: func_db
-
-The function DB_KEYCOUNT has been added, which
-returns the cardinality of the keys at a specified
-prefix in AstDB, i.e. the number of keys at a
-given prefix.

doc/CHANGES-staging/func_env.txt

@@ -1,5 +0,0 @@
-Subject: func_env.c
-
-Two new functions, DIRNAME and BASENAME, are now
-included which allow users to obtain the directory
-or the base filename of any file.

doc/CHANGES-staging/func_evalexten.txt

@@ -1,4 +0,0 @@
-Subject: func_evalexten
-
-This adds the EVAL_EXTEN function which may be
-used to evaluate data at dialplan extensions.

doc/CHANGES-staging/func_framedrop.txt

@@ -1,5 +0,0 @@
-Subject: func_framedrop
-
-New function to selectively drop specified frames
-in either direction on a channel.
-

doc/CHANGES-staging/func_json.txt

@@ -1,5 +0,0 @@
-Subject: func_json
-
-The JSON_DECODE dialplan function can now be used
-to parse JSON strings, such as in conjunction with
-CURL for using API responses.

doc/CHANGES-staging/func_odbc_esc_backslashes.txt

@@ -1,7 +0,0 @@
-Subject: func_odbc
-
-A SQL_ESC_BACKSLASHES dialplan function has been added which
-escapes backslashes. Usage of this is dependent on whether the
-database in use can use backslashes to escape ticks or not. If
-it can, then usage of this prevents a broken SQL query depending
-on how the SQL query is constructed.

doc/CHANGES-staging/func_scramble.txt

@@ -1,5 +0,0 @@
-Subject: func_scramble
-
-Adds an audio scrambler function that may be used to
-distort voice audio on a channel as a privacy
-enhancement.

doc/CHANGES-staging/func_strings.txt

@@ -1,7 +0,0 @@
-Subject: func_strings
-
-A new STRBETWEEN function is now included which
-allows a substring to be inserted between characters
-in a string. This is particularly useful for transforming
-dial strings, such as adding pauses between digits
-for a string of digits that are sent to another channel.

doc/CHANGES-staging/func_vmcount.txt

@@ -1,3 +0,0 @@
-Subject: func_vmcount
-
-Multiple mailboxes may now be specified instead of just one.

doc/CHANGES-staging/load_realtime_queues.txt

@@ -1,5 +0,0 @@
-Subject: app_queue
-
-Load queues and members from Realtime for
-AMI actions: QueuePause, QueueStatus and QueueSummary,
-Applications: PauseQueueMember and UnpauseQueueMember.

doc/CHANGES-staging/logger.txt

@@ -1,5 +0,0 @@
-Subject: logger
-
-Added the ability to define custom log levels in logger.conf
-and use them in the Log dialplan application. Also adds a
-logger show levels CLI command.

doc/CHANGES-staging/manager_disable.txt

@@ -1,4 +0,0 @@
-Subject: ami
-
-AMI events can now be globally disabled using
-the disabledevents [general] setting.

doc/CHANGES-staging/manager_message_send.txt

@@ -1,6 +0,0 @@
-Subject: MessageSend
-
-The MessageSend AMI action has been updated to allow the Destination
-and the To addresses to be provided separately. This brings the
-MessageSend manager command in line with the capabilities of the
-MessageSend dialplan application.

doc/CHANGES-staging/mf.txt

@@ -1,6 +0,0 @@
-Subject: Channel-agnostic MF support
-
-A SendMF application and PlayMF manager
-application are now included to send
-arbitrary standard R1 MF tones on the
-current channel or another specified channel.

doc/CHANGES-staging/pjsip_read_headers.txt

@@ -1,5 +0,0 @@
-Subject: chan_pjsip
-
-Add function PJSIP_HEADERS() to get list of headers by pattern in the same way as SIP_HEADERS() do.
-
-Add ability to read header by pattern using PJSIP_HEADER().

doc/CHANGES-staging/queue_withdraw_caller.txt

@@ -1,14 +0,0 @@
-Subject: app_queue
-
-Added a new AMI action: QueueWithdrawCaller
-This AMI action makes it possible to withdraw a caller from a queue
-back to the dialplan. The call will be signaled to leave the queue
-whenever it can, hence, it not guaranteed that the call will leave
-the queue.
-
-Optional custom data can be passed in the request, in the WithdrawInfo
-parameter. If the call successfully withdrawn the queue,
-it can be retrieved using the QUEUE_WITHDRAW_INFO variable.
-
-This can be useful for certain uses, such as dispatching the call
-to a specific extension.

doc/CHANGES-staging/res_agi.txt

@@ -1,5 +0,0 @@
-Subject: res_agi
-
-Agi command 'exec' can now be enabled
-to evaluate dialplan functions and variables
-by setting the variable AGIEXECFULL to yes.

doc/CHANGES-staging/res_cliexec.txt

@@ -1,6 +0,0 @@
-Subject: res_cliexec
-
-A new CLI command, dialplan exec application, has
-been added which allows dialplan applications to be
-executed at the CLI, useful for some quick testing
-without needing to write dialplan.

doc/CHANGES-staging/res_fax_spandsp.txt

@@ -1,3 +0,0 @@
-Subject: res_fax_spandsp
-
-Adds support for spandsp 3.0.0.

doc/CHANGES-staging/res_geolocation.txt

@@ -1,4 +0,0 @@
-Subject: res_geolocation
-
-Added res_geolocation which creates the core capabilities
-to manipulate Geolocation information on SIP INVITEs.

doc/CHANGES-staging/res_parking_moh.txt

@@ -1,4 +0,0 @@
-Subject: res_parking
-
-An m option to Park and ParkAndAnnounce now allows
-specifying a music on hold class override.

doc/CHANGES-staging/res_pjproject.txt

@@ -1,8 +0,0 @@
-Subject: res_pjproject
-
-In pjproject.conf you can now map pjproject log levels
-to the Asterisk TRACE log level.  The default mappings
-have therefore changed so that only pjproject levels
-3 and 4 are mapped to DEBUG and 5 and 6 are now mapped
-to TRACE.  Previously 3, 4, 5, and 6 were all mapped to
-DEBUG.

doc/CHANGES-staging/res_pjsip_geolocation.txt

@@ -1,4 +0,0 @@
-Subject: res_pjsip_geolocation
-
-Added res_pjsip_geolocation which gives chan_pjsip
-the ability to use the core geolocation capabilities.

doc/CHANGES-staging/res_pjsip_header_funcs.txt

@@ -1,5 +0,0 @@
-Subject: res_pjsip_header_funcs
-
-Add function PJSIP_RESPONSE_HEADERS() to get list of header names from 200 response, in the same way as PJSIP_HEADERS() from the request.
-
-Add function PJSIP_RESPONSE_HEADER() to read header from 200 response, in the same way as PJSIP_HEADER() from the request.

doc/CHANGES-staging/res_pjsip_registrar.txt

@@ -1,7 +0,0 @@
-Subject: res_pjsip_registrar
-
-Adds new PJSIP AOR option remove_unavailable to either
-remove unavailable contacts when a REGISTER exceeds
-max_contacts when remove_existing is disabled, or
-prioritize unavailable contacts over other existing
-contacts when remove_existing is enabled.

doc/CHANGES-staging/res_pjsip_t38_bind_fixes.txt

@@ -1,9 +0,0 @@
-Subject: res_pjsip_t38
-
-In res_pjsip_sdp_rtp, the bind_rtp_to_media_address option and the
-fallback use of the transport's bind address solve problems sending
-media on systems that cannot send ipv4 packets on ipv6 sockets, and
-certain other situations. This change extends both of these behaviors
-to UDPTL sessions as well in res_pjsip_t38, to fix fax-specific
-problems on these systems, introducing a new option
-endpoint/t38_bind_udptl_to_media_address.

doc/CHANGES-staging/res_rtp_asterisk_stunaddr_recurring_resolution.txt

@@ -1,6 +0,0 @@
-Subject: res_rtp_asterisk
-
-When the address of the STUN server (stunaddr) is a name resolved via DNS, the
-stunaddr will be recurringly resolved when the DNS answer Time-To-Live (TTL)
-expires. This allows the STUN server to change its IP address without having to
-reload the res_rtp_asterisk module.

doc/CHANGES-staging/res_statsd.txt

@@ -1,5 +0,0 @@
-Subject: Handle non-standard Meter metric type safely
-
-A meter_support flag has been introduced that defaults to true to maintain current behaviour.
-If disabled, a counter metric type will be used instead wherever a meter metric type was used,
-the counter will have a "_meter" suffix appended to the metric name.

doc/CHANGES-staging/res_tonedetect.txt

@@ -1,5 +0,0 @@
-Subject: res_tonedetect
-
-Arbitrary tone detection is now available through a
-WaitForTone application (blocking) and a TONE_DETECT
-function (non-blocking).

doc/CHANGES-staging/rls_display_name.txt

@@ -1,10 +0,0 @@
-Subject: res_pjsip_pubsub
-
-A new resource_list option, resource_display_name, indicates
-whether display name of resource or the resource name being
-provided for RLS entries.
-If this option is enabled, the Display Name will be provided.
-This option is disabled by default to remain the previous behavior.
-If the 'event' set to 'presence' or 'dialog' the non-empty HINT name
-will be set as the Display Name.
-The 'message-summary' is not supported yet.

doc/CHANGES-staging/rls_refresh.txt

@@ -1,7 +0,0 @@
-Subject: res_pjsip_pubsub
-
-The Resource List Subscriptions (RLS) is dynamic now.
-The asterisk now updates current subscriptions to reflect the changes
-to the list on subscription refresh. If list items are added,
-removed, updated or do not exist anymore, the asterisk regenerates
-the resource list.

doc/CHANGES-staging/say.txt

@@ -1,7 +0,0 @@
-Subject: say.c
-
-Adds SAYFILES function to retrieve the file names that would
-be played by corresponding Say applications, such as
-SayDigits, SayAlpha, etc.
-
-Additionally adds SayMoney and SayOrdinal applications.

doc/CHANGES-staging/tonescan.txt

@@ -1,6 +0,0 @@
-Subject: ToneScan application
-
-A new application, ToneScan, allows for
-synchronous detection of call progress
-signals such as dial tone, busy tone,
-Special Information Tones, and modems.

doc/UPGRADE-staging/chan_iax2_rsa.txt

@@ -1,15 +0,0 @@
-Subject: 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.

doc/UPGRADE-staging/http-media-cache-lookup-order.txt

@@ -1,9 +0,0 @@
-Subject: 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.

doc/UPGRADE-staging/manager_amxml_attribute_fix.txt

@@ -1,8 +0,0 @@
-Subject: 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.

doc/UPGRADE-staging/res_monitor_disabled.txt

@@ -1,8 +0,0 @@
-Subject: res_monitor
-Master-Only: True
-
-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.

doc/UPGRADE-staging/res_pjsip_async_operations.txt

@@ -1,7 +0,0 @@
-Subject: 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.

doc/UPGRADE-staging/stir_shaken_option_split.txt

@@ -1,7 +0,0 @@
-Subject: 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.