--- /dev/null
+==============================================================================
+===
+=== THIS FILE IS AUTOMATICALLY GENERATED DURING THE RELEASE
+=== PROCESS. DO NOT MAKE CHANGES HERE. INSTEAD, REFER TO
+=== doc/CHANGES-staging/README.md FOR MORE DETAILS.
+===
+=== 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 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\r
+ to evaluate dialplan functions and variables\r
+ 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 ------------
+------------------------------------------------------------------------------
+
+AMI Flash event
+------------------
+ * Hook flash events are now exposed as AMI events.
+
+Add variable support to Originate
+------------------
+ * The Originate application now allows
+ variables to be set on the new channel
+ through a new option.
+
+Core
+------------------
+ * Added debug logging categories that allow a user to output debug information
+ based on a specified category. This lets the user limit, and filter debug
+ output to data relevant to a particular context, or topic. For instance the
+ following categories are now available for debug logging purposes:
+
+ dtls, dtls_packet, ice, rtcp, rtcp_packet, rtp, rtp_packet, stun, stun_packet
+
+ These debug categories can be enable/disable via an Asterisk CLI command:
+
+ core set debug category <category>[:<sublevel>] [category[:<sublevel] ...]
+ core set debug category off [<category> [<category>] ...]
+
+ If no sub-level is associated all debug statements for a given category are
+ output. If a sub-level is given then only those statements assigned a value
+ at or below the associated sub-level are output.
+
+ * The location where the media cache stores its temporary files
+ is no longer hardcoded to /tmp but can now be configured separately
+ via the astcachedir config variable in asterisk.conf.
+
+ The default location for astcachedir is now /var/cache/asterisk
+ instead of /tmp, please make sure to manually cleanup and/or
+ migrate the temporary files in /tmp after upgrading.
+
+MessageSend
+------------------
+ * The MessageSend dialplan application now takes an
+ optional third argument that can set the message's
+ "To" field on outgoing messages. It's an alternative
+ to using the MESSAGE(to) dialplan function.
+
+ To prevent confusion with the first argument, currently
+ named "to", it's been renamed to "destination".
+ Its function, creating the request URI, hasn't changed.
+
+ The online documentation has also been enhanced to
+ explain the behavior.
+
+ Despite the changes in this commit, there should be
+ no impact to current users of MessageSend.
+
+New ConfKick application
+------------------
+ * Adds a ConfKick() application, which allows
+ a specific channel, all users, or all non-admin
+ users to be kicked from a conference bridge.
+
+New Reload application
+------------------
+ * Adds an application to reload modules
+
+PlaybackFinished has a new error state
+------------------
+ * The PlaybackFinished event now has a new state "failed"
+ that is used when the sound file was not played due to an error.
+ Before the state on PlaybackFinished was always "done".
+
+ In case of multiple sound files to be played,
+ the PlaybackFinished is sent only once in the end of the list,
+ even in case of error.
+
+WaitForCondition application
+------------------
+ * This application provides a way to halt
+ dialplan execution until a provided
+ condition evaluates to true.
+
+app_confbridge
+------------------
+ * app_confbridge now has the ability to force the estimated bitrate on an SFU
+ bridge. To use it, set a bridge profile's remb_behavior to "force" and
+ set remb_estimated_bitrate to a rate in bits per second. The
+ remb_estimated_bitrate parameter is ignored if remb_behavior is something
+ other than "force".
+
+app_confbridge answer supervision control
+------------------
+ * app_confbridge now provides a user option to prevent
+ answer supervision if the channel hasn't been
+ answered yet. To use it, set a user profile's
+ answer_channel option to no.
+
+app_dial announcement option
+------------------
+ * The A option for Dial now supports
+ playing audio to the caller as well
+ as the called party.
+
+app_mixmonitor
+------------------
+ * app_mixmonitor now sends manager events MixMonitorStart, MixMonitorStop and
+ MixMonitorMute when the channel monitoring is started, stopped and muted (or
+ unmuted) respectively.
+
+app_voicemail
+------------------
+ * The VoiceMail application can now be configured to send greetings and
+ instructions via early media and only answering the channel when it is
+ time for the caller to record their message. This behavior can be
+ activated by passing the new 'e' option to VoiceMail.
+
+ * You can now customize the "beep" tone or omit it entirely.
+
+chan_iax2
+------------------
+ * You can now specify a default "auth" method in the
+ [general] section of iax.conf
+
+chan_pjsip
+------------------
+ * The PJSIP_SEND_SESSION_REFRESH dialplan function now issues a warning, and
+ returns unsuccessful if it's used on a channel prior to answering.
+
+chan_pjsip, app_transfer
+------------------
+ * Added TRANSFERSTATUSPROTOCOL variable. When transfer is performed,
+ transfers can pass a protocol specific error code.
+ Example, in SIP 3xx-6xx represent any SIP specific error received when
+ performing a REFER.
+
+func_math: Three new dialplan functions
+------------------
+ * Introduce three new functions, MIN, MAX, and ABS, which can be used to
+ obtain the minimum or maximum of up to two integers or absolute value.
+
+func_odbc
+------------------
+ * Introduce an ARGC variable for func_odbc functions, along with a minargs
+ per-function configuration option.
+
+ minargs enables enforcing of minimum count of arguments to pass to
+ func_odbc, so if you're unconditionally using ARG1 through ARG4 then
+ this should be set to 4. func_odbc will generate an error in this case,
+ so for example
+
+ [FOO]
+ minargs = 4
+
+ and ODBC_FOO(a,b,c) in dialplan will now error out instead of using a
+ potentially leaked ARG4 from Gosub().
+
+ ARGC is needed if you're using optional argument, to verify whether or
+ not an argument has been passed, else it's possible to use a leaked ARGn
+ from Gosub (app_stack). So now you can safely do
+ ${IF($[${ARGC}>3]?${ARGV}:default value)} kind of thing.
+
+func_volume now can be read
+------------------
+ * The VOLUME function can now also be used
+ to read existing values previously set.
+
+logger
+------------------
+ * Added a new log formatter called "plain" that always prints
+ file, function and line number if available (even for verbose
+ messages) and never prints color control characters. Most
+ suitable for file output but can be used for other channels
+ as well.
+
+ You use it in logger.conf like so:
+ debug => [plain]debug
+ console => [plain]error,warning,debug,notice,pjsip_history
+ messages => [plain]warning,error,verbose
+
+ * The dateformat option in logger.conf will now control the remote
+ console (asterisk -r -T) timestamp format. Previously, dateformat only
+ controlled the formatting of the timestamp going to log files and the
+ main console (asterisk -c) but only for non-verbose messages.
+
+ Internally, Asterisk does not send the logging timestamp with verbose
+ messages to console clients. It's up to the Asterisk remote consoles
+ to format verbose messages. Asterisk remote consoles previously did
+ not load dateformat from logger.conf.
+
+ Previously there was a non-configurable and hard-coded "%b %e %T"
+ dateformat that would be used no matter what on all verbose console
+ messages printed on remote consoles.
+
+ Example:
+ logger.conf
+ dateformat=%F %T.%3q
+
+ # asterisk -rvvv -T
+ [2021-03-19 09:54:19.760-0400] Loading res_stasis_answer.so.
+ [Mar 19 09:55:43] -- Goto (dialExten,s,1)
+
+ Given the following example configuration in logger.conf, Asterisk log
+ files and the console, will log verbose messages using the given
+ timestamp. Now ensuring that all remote console messages are logged
+ with the same dateformat as other log streams.
+
+ ---
+ [general]
+ dateformat=%F %T.%3q
+
+ [logfiles]
+ console => notice,warning,error,verbose
+ full => notice,warning,error,debug,verbose
+ ---
+
+ Now we have a globally-defined dateformat that will be used
+ consistently across the Asterisk main console, remote consoles, and
+ log files.
+
+ Now we have consistent logging:
+
+ # asterisk -rvvv -T
+ [2021-03-19 09:54:19.760-0400] Loading res_stasis_answer.so.
+ [2021-03-19 09:55:43.920-0400] -- Goto (dialExten,s,1)
+
+res_pjsip
+------------------
+ * PJSIP transports can now be partially reloaded safely. This allows the
+ local_net and external_* options to be updated without restarting Asterisk.
+
+ * PJSIP endpoints can now be configured to skip authentication when
+ handling OPTIONS requests by setting the allow_unauthenticated_options
+ configuration property to 'yes.'
+
+ * PJSIP support of registrations of endpoints in multidomain
+ scenarios, where the endpoint contains the domain info
+ in pjsip.conf.
+
+res_pjsip_dialog_info_body_generator
+------------------
+ * PJSIP now supports RFC 4235 Section 4.1.6 dialog-info+xml local and
+ remote elements by iterating through ringing channels and inserting
+ that info into NOTIFY packet sent to the endpoint.
+
+res_pjsip_messaging
+------------------
+ * Implemented the new "to" parameter of the MessageSend()
+ dialplan application. This allows a user to specify
+ a complete SIP "To" header separate from the Request URI.
+ We now also accept a destination in the same format
+ as Dial()... PJSIP/number@endpoint
+
+res_rtp_asterisk
+------------------
+ * By default Asterisk reports the PJSIP version in all
+ STUN packets it sends.
+
+ This behaviour may not be desired in a production
+ environment and can now be disabled by setting the
+ stun_software_attribute option to 'no' in rtp.conf.
+
+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 Streams API becomes the home for the core ACN capabilities.
+ These include...
+
+ * Parsing and formatting of codec negotiation preferences.
+ * Resolving pending streams and topologies with those configured
+ using configured preferences.
+ * Utility functions for creating string representations of
+ streams, topologies, and negotiation preferences.
+
+ For codec negotiation preferences:
+ * Added ast_stream_codec_prefs_parse() which takes a string
+ representation of codec negotiation preferences, which
+ may come from a pjsip endpoint for example, and populates
+ a ast_stream_codec_negotiation_prefs structure.
+ * Added ast_stream_codec_prefs_to_str() which does the reverse.
+ * Added many functions to parse individual parameter name
+ and value strings to their respective enum values, and the
+ reverse.
+
+ For streams:
+ * Added ast_stream_create_resolved() which takes a "live" stream
+ and resolves it with a configured stream and the negotiation
+ preferences to create a new stream.
+ * Added ast_stream_to_str() which create a string representation
+ of a stream suitable for debug or display purposes.
+
+ For topology:
+ * Added ast_stream_topology_create_resolved() which takes a "live"
+ topology and resolves it, stream by stream, with a configured
+ topology stream and the negotiation preferences to create a new
+ topology.
+ * Added ast_stream_topology_to_str() which create a string
+ representation of a topology suitable for debug or display
+ purposes.
+ * Renamed ast_format_caps_from_topology() to
+ ast_stream_topology_get_formats() to be more consistent with
+ the existing ast_stream_get_formats().
+
+ Additional changes:
+ * A new function ast_format_cap_append_names() appends the results
+ to the ast_str buffer instead of replacing buffer contents.
+
+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.
+
+res_pjsip
+------------------
+ * Two new options, incoming_call_offer_pref and outgoing_call_offer_pref
+ have been added to res_pjsip endpoints that specify the preferred order
+ of codecs to use between those received/sent in an SDP offer and those
+ set in the endpoint configuration.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 17.0.0 to Asterisk 18.0.0 ------------
+------------------------------------------------------------------------------
+
+AMI
+------------------
+ * You can now specify an optional 'Content-Type' as an argument for the Asterisk
+ SendText manager action.
+
+ARI
+------------------
+ * A new parameter 'inhibitConnectedLineUpdates' is now available in the
+ 'bridges.addChannel' call. This prevents the identity of the newly connected
+ channel from being presented to other bridge members.
+
+ARI Channels
+------------------
+ * The Channel resource has a new sub-resource "externalMedia".
+ This allows an application to create a channel for the sole purpose
+ of exchanging media with an external server. Once created, this
+ channel could be placed into a bridge with existing channels to
+ allow the external server to inject audio into the bridge or
+ receive audio from the bridge.
+ See https://wiki.asterisk.org/wiki/display/AST/External+Media+and+ARI
+ for more information.
+
+Core
+------------------
+ * H.265/HEVC is now a supported video codec and it can be used by
+ specifying "h265" in the allow line.
+ Please note however, that handling of the additional SDP parameters
+ described in RFC 7798 section 7.2 is not yet supported.
+
+Features
+------------------
+ * Adds support for AudioSocket, a very simple bidirectional audio streaming
+ protocol. There are both channel and application interfaces.
+
+ A description of the protocol can be found on the referenced wiki page. A
+ short talk about the reasons and implementation can be found on YouTube at
+ the link provided.
+
+ ARI support has also been added via the existing "externalMedia" ARI
+ functionality. The UUID is specified using the arbitrary "data" field.
+
+ Wiki: https://wiki.asterisk.org/wiki/display/AST/AudioSocket
+ YouTube: https://www.youtube.com/watch?v=tjduXbZZEgI
+
+Messaging
+------------------
+ * In order to reduce the amount of AMI and ARI events generated,
+ the global "Message/ast_msg_queue" channel can be set to suppress
+ it's normal channel housekeeping events such as "Newexten",
+ "VarSet", etc. This can greatly reduce load on the manager
+ and ARI applications when the Digium Phone Module for Asterisk
+ is in use. To enable, set "hide_messaging_ami_events" in
+ asterisk.conf to "yes" In Asterisk versions <18, the default
+ is "no" preserving existing behavior. Beginning with
+ Asterisk 18, the option will default to "yes".
+
+STIR/SHAKEN
+------------------
+ * STIR/SHAKEN support has been added to Asterisk. Configuration is done in
+ stir_shaken.conf. There is a sample configuration file to help you get
+ started (asterisk/configs/samples/stir_shaken.conf.sample). Once that's
+ set up, you can enable STIR/SHAKEN on any endpoint by setting stir_shaken
+ to yes on the endpoint configuration object. This will add an Identity
+ header on outgoing INVITEs, and check for an Identity header on incoming
+ INVITEs. This option has been added to Alembic as well.
+
+ The information received on an incoming INVITE can be checked using the
+ STIR_SHAKEN dialplan function. There are two variations:
+
+ STIR_SHAKEN(count)
+ STIR_SHAKEN(0, verify_result)
+
+ The first variation will tell you how many STIR/SHAKEN results are on the
+ channel. The second fetches information for a specific result. The first
+ parameter is the index, followed by what information you want to retrieve.
+ The available options are 'verify_result', 'identity', and 'attestation'.
+
+app_chanisavail
+------------------
+ * The ChanIsAvail application now tolerates empty positions in the supplied
+ device list. Dialplan can now be simplified by not having to check for
+ empty positions in the device list.
+
+app_confbridge
+------------------
+ * A new bridge profile option, maximum_sample_rate, has been added which sets
+ a maximum sample rate that the bridge will be mixed at. This allows the bridge
+ to move below the maximum sample rate as needed but caps it at the maximum.
+
+ * A new option, "text_messaging", has been added to the user profile
+ which allows control over whether text messaging is enabled or
+ disabled for a user. If enabled (the default) text messages
+ will be sent to the user. If disabled no text messages will be
+ sent to the user.
+
+app_dial
+------------------
+ * The Dial application now tolerates empty positions in the supplied
+ destination list. Dialplan can now be simplified by not having to check
+ for empty positions in the destination list. If there are no endpoints to
+ dial then DIALSTATUS is set to CHANUNAVAIL.
+
+app_mixmonitor
+------------------
+ * An option 'S' has been added to MixMonitor. If used in combination with
+ the r() and/or t() options, if a frame is available to write to one of
+ those files but not the other, a frame of silence if written to the file
+ that does not have an audio frame. This should prevent the two files
+ from "drifting" when mixed after the fact.
+
+ * If the 'filename' argument to MixMonitor() ended with '.wav49,'
+ Asterisk would silently convert the extension to '.WAV' when opening
+ the file for writing. This caused the MIXMONITOR_FILENAME variable to
+ reference the wrong file. The MIXMONITOR_FILENAME variable will now
+ reflect the name of the file that Asterisk actually used instead of
+ the filename that was passed to the application.
+
+app_page
+------------------
+ * The Page application now tolerates empty positions in the supplied
+ destination list. Dialplan can now be simplified by not having to check
+ for empty positions in the destination list.
+
+app_voicemail
+------------------
+ * A feature was added in Asterisk 13.27.0 and 16.4.0 that removed lock files from
+ the Asterisk voicemail directory on startup. Some users that store their
+ voicemails on network storage devices experienced slow startup times due to the
+ relative expense of traversing the voicemail directory structure looking for
+ orphaned lock files. This feature has now been removed.
+
+ Users who require the lock files to be removed at startup should modify their
+ startup scripts to do so before starting the asterisk process.
+
+chan_pjsip
+------------------
+ * A new dialplan function, PJSIP_MOH_PASSTHROUGH, has been added to chan_pjsip. This
+ allows the behaviour of the moh_passthrough endpoint option to be read or changed
+ in the dialplan. This allows control on a per-call basis.
+
+chan_rtp
+------------------
+ * The UnicastRTP channel driver provided by chan_rtp now accepts
+ "<hostname>:<port>" as an alternative to "<ip_address>:<port>" in the destination.
+ The first AAAA (preferred) or A record resolved will be used as the destination.
+ The lookup is synchronous so beware of possible dialplan delays if you specify a
+ hostname.
+
+func_curl
+------------------
+ * A new parameter, httpheader, has been added to CURLOPT function. This parameter
+ allows to set custom http headers for subsequent calls off CURL function.
+ Any setting of headers will replace the default curl headers
+ (e.g. "Content-type: application/x-www-form-urlencoded")
+
+ * A new option, followlocation, can now be enabled with the CURLOPT()
+ dialplan function. Setting this will instruct cURL to follow 3xx
+ redirects, which it does not by default.
+
+func_jitterbuffer
+------------------
+ * The JITTERBUFFER dialplan function now has an option to enable video synchronization
+ support. When enabled and used with a compatible channel driver (chan_sip, chan_pjsip)
+ the video is buffered according to the size of the audio jitterbuffer and is
+ synchronized to the audio.
+
+func_volume
+------------------
+ * Accept decimal number as argument.
+
+http
+------------------
+ * You can now disable the /httpstatus page served by Asterisk's built-in
+ HTTP server by setting 'enable_status' to 'no' in http.conf.
+
+minmemfree
+------------------
+ * The 'minmemfree' configuration option now counts memory allocated to
+ the filesystem cache as "free" because it is memory that is available
+ to the process.
+
+res_ari_channels
+------------------
+ * When creating a channel in ARI using the create call
+ you can now specify dialplan variables to be set as part
+ of the same operation.
+
+res_musiconhold
+------------------
+ * This fix allows a realtime moh class to be unregistered from the command
+ line. This is useful when the contents of a directory referenced by a
+ realtime moh class have changed.
+ The realtime moh class is then reloaded on the next request and uses the
+ new directory contents.
+
+ * A new mode - playlist - has been added to res_musiconhold. This mode allows the
+ user to specify the files (or URLs) to play explicitly by putting them directly
+ in musiconhold.conf.
+
+res_pjsip
+------------------
+ * Added a new PJSIP system setting called disable_rport.
+ Default is no to keep support working as before.
+
+ If it is false (default) it adds the 'rport' parameter in the outgoing request message.
+ If it is true it does not add the 'rport' parameter in the outgoing request message.
+
+ This is a system option, but working as a global option.
+
+res_pjsip_endpoint_identifier_ip
+------------------
+ * In 'type = identify' sections, the addresses specified for the 'match'
+ clause can now include a port number. For IP addresses, the port is
+ provided by including a colon after the address, followed by the
+ desired port number. If supplied, the netmask should follow the port
+ number. To specify a port for IPv6 addresses, the address itself must
+ be enclosed in brackets to be parsed correctly.
+
+res_pjsip_logger
+------------------
+ * The PJSIP packet logger now has the following CLI commands:
+
+ pjsip set logger pcap <filename>
+
+ When used this will create a pcap file containing the incoming
+ and outgoing SIP packets, in unencrypted form.
+
+ pjsip set logger console <on / off>
+
+ This allows you to toggle logging to console on and off.
+
+ pjsip set logger host <IP/subnet mask> add
+
+ This allows you to add an additional IP address or subnet
+ mask to logging, allowing you to log multiple instead of
+ just a single IP address or all traffic.
+
+ The normal "pjsip set logger host" CLI command has also been
+ expanded to allow subnet masks as well.
+
+res_pjsip_session
+------------------
+ * When placing an outgoing call to a PJSIP endpoint the intent
+ of any requested formats will now be respected. If only an audio
+ format is requested (such as ulaw) but the underlying endpoint
+ does not support the format the resulting SDP will still only
+ contain an audio stream, and not any additional streams such as
+ video.
+
+ * Two new options, incoming_call_offer_pref and outgoing_call_offer_pref
+ have been added to res_pjsip endpoints that specify the preferred order
+ of codecs to use between those received/sent in an SDP offer and those
+ set in the endpoint configuration.
+
+res_rtp_asterisk
+------------------
+ * This change include a new cli command 'rtp show settings'
+
+ The command display by general settings of rtp configuration. For this
+ point is added the fields: rtpstart, rtpend, dtmftimeout, rtpchecksum,
+ strictrtp, learning_min_sequential and icesupport.
+
+ * The blacklist mechanism in res_rtp_asterisk for ICE and STUN was converted to
+ an ACL mechanism.
+
+ As such six now options are now available:
+
+ ice_deny
+ ice_permit
+ ice_acl
+ stun_deny
+ stun_permit
+ stun_acl
+
+ These options have their obvious meanings as used elsewhere.
+
+ Backwards compatibility was maintained by adding {stun,ice}_blacklist as
+ aliases for {stun,ice}_deny.
+
+res_sorcery_memory_cache
+------------------
+ * The SorceryMemoryCacheExpireObject AMI action and CLI
+ command allow expiring of a specific object within the
+ sorcery memory cache. This is done by removing the
+ object from the cache with the expectation that the
+ cache will then re-populate the object when it is next
+ needed.
+
+ For full backend caching this does not occur. The cache
+ won't repopulate until an entire refresh is done resulting
+ in the possibility that objects are missing until that
+ time.
+
+ The AMI action and CLI command will now not allow
+ expiring of an object if the cache is configured as a
+ full backend cache. Instead you must use either the
+ SorceryMemoryCacheExpire or SorceryMemoryCachePopulate
+ AMI actions or their associated CLI commands.
+
+taskprocessor.c
+------------------
+ * Added two new CLI commands to reset stats for taskprocessors. You can
+ reset stats for a single, specific taskprocessor ('core reset
+ taskprocessor <taskprocessor>'), or you can reset all taskprocessors
+ ('core reset taskprocessors'). These commands will reset the counter for
+ the number of tasks processed as well as the max queue size.
+
+ * Added "like" support for 'core show taskprocessors'. Now you
+ can specify a specific set of taskprocessors (or just one) by
+ adding the keyword "like" to the above command, followed by
+ your search criteria.
+
+------------------------------------------------------------------------------
+--- New functionality introduced in Asterisk 17.0.0 --------------------------
+------------------------------------------------------------------------------
+
+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.
+
+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
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 16.0.0 to Asterisk 17.0.0 ------------
+------------------------------------------------------------------------------
+
+AttendedTransfer
+------------------
+ * A new application, this will queue up attended transfer to the given extension.
+
+BlindTransfer
+------------------
+ * A new application, this will redirect all channels currently
+ bridged to the caller channel to the specified destination.
+
+ConfBridge
+------------------
+ * Add "average_all", "highest_all", and "lowest_all" values for
+ the remb_behavior option. These values operate on a bridge
+ level instead of a per-source level. This means that a single
+ REMB value is calculated and sent to every sender, instead of
+ a REMB value that is unique for the specific sender..
+
+Dial
+------------------
+ * Add RINGTIME and RINGTIME_MS variables containing respectively seconds and
+ milliseconds between creation of the dialing channel and receiving the first
+ RINGING signal
+
+ Add PROGRESSTIME and PROGRESSTIME_MS variables analogous to the above with respect to
+ the PROGRESS signal. Shorter of these two times should be equivalent to
+ the PDD (Post Dial Delay) value
+
+ Add DIALEDTIME_MS and ANSWEREDTIME_MS variables to get millisecond resolution
+ versions of DIALEDTIME and ANSWEREDTIME
+
+RTP/ICE
+------------------
+ * You can now indicate that you'd like an ice_host_candidate's local address
+ to be published as well as the mapped address. See the sample rtp.conf
+ for more information.
+
+ReadExten
+------------------
+ * Add 'p' option to stop reading extension if user presses '#' key.
+
+pbx_dundi
+------------------
+ * The DUNDi PBX module now supports IPv4/IPv6 dual binding.
+
+res_pjsip
+------------------
+ * Added a new PJSIP global setting called norefersub.
+ Default is true to keep support working as before.
+
+ res_pjsip_refer configures PJSIP norefersub capability accordingly.
+
+ Checks the PJSIP global setting value.
+ If it is true (default) it adds the norefersub capability to PJSIP.
+ If it is false (disabled) it does not add the norefersub capability
+ to PJSIP.
+
+ This is useful for Cisco switches that do not follow RFC4488.
+
+res_rtp_asterisk
+------------------
+ * DTLS packets will now be fragmented according to the MTU as set in rtp.conf. This
+ allows larger certificates to be used for the DTLS negotiation. By default this value
+ is 1200.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 16.2.0 to Asterisk 16.3.0 ----------
+------------------------------------------------------------------------------
+
+ARI
+------------------
+ * Application event filtering is now supported. An application can now specify
+ an "allowed" and/or "disallowed" list(s) of event types. Only those types
+ indicated in the "allowed" list are sent to the application. Conversely, any
+ types defined in the "disallowed" list are not sent to the application. Note
+ that if a type is specified in both lists "disallowed" takes precedence.
+
+ * A new REST API call has been added: 'move'. It follows the format
+ 'channels/{channelId}/move' and can be used to move channels from one application
+ to another without needing to exit back into the dialplan. An application must be
+ specified, but the passing a list of arguments to the new application is optional.
+ An example call would look like this:
+
+ client.channels.move(channelId=chan.id, app='ari-example', appArgs='a,b,c')
+
+ If the channel was inside of a bridge when switching applications, it will
+ remain there. If the application specified cannot be moved to, then the channel
+ will remain in the current application and an event will be triggered named
+ "ApplicationMoveFailed", which will provide the destination application's name
+ and the channel information.
+
+res_pjsip
+------------------
+ * A new configuration parameter "taskprocessor_overload_trigger" has been
+ added to the pjsip.conf "globals" section. The distributor currently stops
+ accepting new requests when any taskprocessor overload is triggered. The
+ new option allows you to completely disable overload detection (NOT
+ RECOMMENDED), keep the current behavior, or trigger only on pjsip
+ taskprocessor overloads.
+
+chan_pjsip
+------------------
+ * A new configuration parameter 'ignore_183_without_sdp' has been added
+ to the pjsip.conf "endpoints" section. If enabled, will make chan_pjsip
+ discard 183s that do not contain an SDP body, which can resolve no
+ ringback tone issues as well as making the behavior match chan_sip.
+
+MWI
+------------------
+ * A new module "res_mwi_devstate" has been added that allows subscriptions
+ to voicemail boxes using "presence" events. This allows common BLF keys
+ to act as voicemail waiting indicators.
+
+app_queue
+------------------
+ * Added the ability to set the wrapuptime per-member using the AddQueueMember
+ application.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 16.1.0 to Asterisk 16.2.0 ------------
+------------------------------------------------------------------------------
+
+ARI
+------------------
+ * Whenever an ARI application is started, a context will be created for it
+ automatically as long as one does not already exist, following the format
+ 'stasis-<app_name>'. Two extensions are also added to this context: a match-all
+ extension, and the 'h' extension. Any phone that registers under this context
+ will place all calls to the corresponding Stasis application.
+
+res_pjsip
+------------------
+ * Added "send_contact_status_on_update_registration" global configuration option
+ to enable sending AMI ContactStatus event when a device refreshes its registration.
+
+Core
+------------------
+ * Reworked the media indexer so it doesn't cache the index. Testing revealed
+ that the cache added no benefit but that it could consume excessive memory.
+ Two new index related functions were created: ast_sounds_get_index_for_file()
+ and ast_media_index_update_for_file() which restrict index updating to
+ specific sound files. The original ast_sounds_get_index() and
+ ast_media_index_update() calls are still available but since they no longer
+ cache the results internally, developers should re-use an index they may
+ already have instead of calling ast_sounds_get_index() repeatedly. If
+ information for only a single file is needed, ast_sounds_get_index_for_file()
+ should be called instead of ast_sounds_get_index().
+
+Features
+------------------
+ * Before Asterisk 12, when using the automon or automixmon features defined
+ in features.conf, a channel variable (TOUCH_MIXMONITOR_OUTPUT) was set on
+ both channels, indicating the filename of the recording.
+
+ When bridging was overhauled in Asterisk 12, the behavior was changed such
+ that the variable was only set on the peer channel and not on the channel
+ that initiated the automon or automixmon.
+
+ The previous behavior has been restored so both channels receive the
+ channel variable when one of these features is invoked.
+
+app_voicemail
+------------------
+ * You can now specify a special context with the "aliasescontext" parameter
+ in voicemail.conf which will allow you to create aliases for physical
+ mailboxes.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 16.0.0 to Asterisk 16.1.0 ------------
+------------------------------------------------------------------------------
+
+pbx_config
+------------------
+ * pbx_config will now find and process multiple 'globals' sections from
+ extensions.conf. Variables are processed in the order they are found
+ and duplicate variables overwrite the previous value.
+
+chan_pjsip
+------------------
+ * New dialplan function PJSIP_PARSE_URI added to parse an URI and return
+ a specified part of the URI.
+
+Core
+------------------
+ * ast_bt_get_symbols() now returns a vector of strings instead of an
+ array of strings. This must be freed with ast_bt_free_symbols.
+
+res_pjsip
+------------------
+ * New options 'trust_connected_line' and 'send_connected_line' have been
+ added to the endpoint. The option 'trust_connected_line' is to control
+ if connected line updates are accepted from this endpoint.
+ The option 'send_connected_line' is to control if connected line updates
+ can be sent to this endpoint.
+ The default value is 'yes' for both options.
+
+res_rtp_asterisk
+------------------
+ * The existing strictrtp option in rtp.conf has a new choice availabe, called
+ 'seqno', which behaves the same way as setting strictrtp to 'yes', but will
+ ignore the time interval during learning so that bursts of packets can still
+ trigger learning our source.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 15 to Asterisk 16 --------------------
+------------------------------------------------------------------------------
+
+app_fax
+------------------
+ * The app_fax module is now deprecated, users should migrate to the
+ replacement module res_fax.
+
+app_originate
+------------------
+ * An 'a' option has been added to the Originate dialplan application which
+ will execute the originate in an asynchronous fashion. If set then the
+ application will return immediately without waiting for the originated
+ channel to answer.
+
+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`.
+
+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.
+
+app_setcallerid
+------------------
+ * The app_setcallerid module has been removed. The CALLERID dialplan function
+ should be used instead.
+
+chan_sip
+------------------
+ * New function SIP_HEADERS() enumerates all headers in the incoming INVITE.
+
+ * The variable GET_TRANSFERRER_DATA set in the peer channel causes matching
+ headers be retrieved from the REFER message and made accessible to the
+ dialplan in the hash TRANSFER_DATA.
+
+chan_dahdi
+------------------
+ * Timeouts for reading digits from analog phones are now configurable in
+ chan_dahdi.conf: firstdigit_timeout, interdigit_timeout, matchdigit_timeout.
+
+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 "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.
+
+app_queue
+------------------
+ * Added the ability to set the wrapuptime in the configuration of member.
+ When set the wrapuptime on the member is used instead of the wrapuptime
+ defined for the queue itself.
+
+ * Added predial handler support for caller and callee channels with the
+ B and b options respectively. This is similar to the predial support
+ in app_dial.
+
+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.
+
+res_pjsip
+------------------
+ * A new AMI action, PJSIPShowAors, has been added which displays information
+ about all configured PJSIP AORs.
+
+ * A new AMI action, PJSIPShowAuths, has been added which displays information
+ about all configured PJSIP Auths.
+
+ * A new AMI action, PJSIPShowContacts, has been added which displays information
+ about all configured PJSIP Contacts.
+
+res_pjsip_registrar_expire
+------------------
+ * The res_pjsip_registrar_expire module has been removed. The functionality has
+ been moved into res_pjsip_registrar.
+
+func_audiohookinherit
+------------------
+ * The func_audiohookinherit module has been removed. Due to architectural changes
+ in Asterisk 12, audiohook inheritance is performed automatically and this
+ function now lacks function.
+
+cdr_syslog
+------------------
+ * The cdr_syslog module is now deprecated and by default it is no longer
+ built.
+
+cdr_sqlite
+------------------
+ * The cdr_sqlite module has been removed. Users should move to using the
+ cdr_sqlite3_custom module instead.
+
+format_jpeg
+------------------
+ * The format_jpeg module has been removed.
+
+pbx_dundi
+------------------
+ * DUNDi now supports IPv6
+
+Core:
+------------------
+ * libedit is no longer available as an embedded library and must be provided
+ by the system.
+ * The STATIC_BUILD functionality has been removed as it has not been maintained
+ and has not worked in quite some time.
+ * 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.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 15.5.0 to Asterisk 15.6.0 ------------
+------------------------------------------------------------------------------
+
+res_pjsip
+------------------
+ * A new option 'suppress_q850_reason_headers' has been added to the endpoint
+ object. Some devices can't accept multiple Reason headers and get confused
+ when both 'SIP' and 'Q.850' Reason headers are received. This option allows
+ the 'Q.850' Reason header to be suppressed. The default value is 'no'.
+
+res_pjsip_endpoint_identifier_ip
+------------------
+ * Added regex support to the identify section match_header option. You
+ specify a regex instead of an explicit string by surrounding the header
+ value with slashes:
+ match_header = SIPHeader: /regex/
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 15.4.0 to Asterisk 15.5.0 ------------
+------------------------------------------------------------------------------
+
+Core
+------------------
+ * Core bridging and, more specifically, bridge_softmix have been enhanced to
+ relay received frames of type TEXT or TEXT_DATA to all participants in a
+ softmix bridge. res_pjsip_messaging and chan_pjsip have been enhanced to
+ take advantage of this so when res_pjsip_messaging receives an in-dialog
+ MESSAGE message from a user in a conference call, it's relayed to all
+ other participants in the call.
+
+app_sendtext
+------------------
+ * Support Enhanced Messaging. SendText now accepts new channel variables
+ that can be used to override the To and From display names and set the
+ Content-Type of a message. Since you can now set Content-Type, other
+ text/* content types are now valid.
+
+app_confbridge
+------------------
+ * ConfbridgeList now shows talking status. This utilizes the same voice
+ detection as the ConfbridgeTalking event, so bridges must be configured
+ with "talk_detection_events=yes" for this flag to have meaning.
+
+ * ConfBridge can now send events to participants via in-dialog MESSAGEs.
+ All current Confbridge events are supported, such as ConfbridgeJoin,
+ ConfbridgeLeave, etc. In addition to those events, a new event
+ ConfbridgeWelcome has been added that will send a list of all
+ current participants to a new participant.
+
+res_pjsip
+------------------
+ * Two new options have been added to the system and endpoint objects to
+ control whether, on outbound calls, Asterisk will accept updated SDP answers
+ during the initial INVITE transaction when 100rel is not in effect.
+ This usually happens when the INVITE is forked to multiple UASs and more
+ than one sends an SDP answer or when a single UAS needs to change a media
+ port to switch from custom ringback to the actual media destination.
+
+ The 'follow_early_media_forked' option sets whether Asterisk will accept
+ the updated SDP when the To tag on the subsequent response is different than
+ that on the the previous response. This usually occurs in the forked INVITE
+ scenario. The default value is "yes" which is the current behavior.
+
+ The 'accept_multiple_sdp_answers' flag sets whether Asterisk will accept the
+ updated SDP when the To tag on the subsequent response is the same as that
+ on the previous response. This can occur when a UAS needs to switch media
+ ports from custom ringback to the final media path. The default value is
+ "no" which is the current behavior.
+
+ These options have to be enabled system-wide in the system config section
+ of pjsip.conf as well as on individual endpoints that require the
+ functionality.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 15.3.0 to Asterisk 15.4.0 ------------
+------------------------------------------------------------------------------
+
+Core
+------------------
+ * A new configuration option "genericplc_on_equal_codecs" was added to the
+ "plc" section of codecs.conf to allow generic packet loss concealment even
+ if no transcoding was originally needed. Transcoding via SLIN is forced
+ in this case.
+
+res_pjproject
+------------------
+ * Added the "cache_pools" option to pjproject.conf. Disabling the option
+ helps track down pool content mismanagement when using valgrind or
+ MALLOC_DEBUG. The cache gets in the way of determining if the pool contents
+ are used after free and who freed it.
+
+res_pjsip_notify
+------------------
+ * Extend the PJSIPNotify AMI command to send an in-dialog notify on a
+ channel.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 15.2.0 to Asterisk 15.3.0 ------------
+------------------------------------------------------------------------------
+
+Core
+------------------
+ * During dialplan reload log messages are produced for each context,
+ extension and include. These messages are no longer printed by the
+ verbose loggers, they are now only logged as debug messages.
+
+app_confbridge
+------------------
+ * Added the Muted header to the ConfbridgeJoin AMI event to indicate the
+ participant's starting mute status.
+
+ * Made the AMI ConfbridgeList action's ConfbridgeList events output all
+ the standard channel snapshot headers instead of a few hand-coded channel
+ snapshot headers. The benefit is that the CallerIDName gets disruptive
+ characters like CR, LF, Tab, and a few others escaped. However, an empty
+ CallerIDName is now output as "<unknown>" instead of "<no name>".
+
+app_followme
+------------------
+ * Added a new prompt, connecting-prompt, which will be played
+ (if configured) to the "winner" callee before connecting the call.
+
+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.
+
+ * The pjsip_transport_event feature introduced in 15.1.0 has been refactored.
+ Any external modules that may have used that feature (highly unlikely) will
+ need to be changed as the API has been altered slightly.
+
+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.
+
+res_pjsip_pubsub
+------------------
+ * In an earlier release, inbound registrations on a reliable transport
+ were pruned on Asterisk restart since the TCP connection would have
+ been torn down and become unusable when Asterisk stopped. This same
+ process is now also applied to inbound subscriptions. Since this
+ required the addition of a new column to the ps_subscription_persistence
+ realtime table, users who store their subscriptions in a database will
+ need to run the "alembic upgrade head" process to add the column to
+ the schema.
+
+res_pjsip_transport_management
+------------------
+ * Since res_pjsip_transport_management provides several attack
+ mitigation features, its functionality moved to res_pjsip and
+ this module has been removed. This way the features will always
+ be available if res_pjsip is loaded.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 15.1.0 to Asterisk 15.2.0 ------------
+------------------------------------------------------------------------------
+
+Core
+------------------
+ * Added the "cache_media_frames" option to asterisk.conf. Disabling the option
+ helps track down media frame mismanagement when using valgrind or
+ MALLOC_DEBUG. The cache gets in the way of determining if the frame is
+ used after free and who freed it. NOTE: This option has no effect when
+ Asterisk is compiled with the LOW_MEMORY compile time option enabled because
+ the cache code does not exist.
+
+chan_sip
+------------------
+ * Calls to invalid extensions are now reported as an ACL failure security event
+ "no_extension_match".
+
+res_rtp_asterisk
+------------------
+ * The X.509 certificate used for DTLS negotiation can now be automatically
+ generated. This is supported by res_pjsip by specifying
+ "dtls_auto_generate_cert = yes" on a PJSIP endpoint. For chan_sip, you
+ would set "dtlsautogeneratecert = yes" either in the [general] section of
+ sip.conf or on a specific peer.
+
+res_pjsip
+------------------
+ * The "identify_by" on endpoints can now be set to "ip" to restrict an endpoint
+ being matched based only on IP address. To ensure no behavior change the
+ default has been changed to "username,ip".
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 15.0.0 to Asterisk 15.1.0 ------------
+------------------------------------------------------------------------------
+
+res_pjsip
+------------------
+ * The "remove_existing" option now allows a registration to succeed by
+ displacing any existing contacts that now exceed the "max_contacts" count.
+ Any removed contacts are the next to expire. The behaviour change is
+ beneficial when "rewrite_contact" is enabled and "max_contacts" is greater
+ than one. The removed contact is likely the old contact created by
+ "rewrite_contact" that the device is refreshing.
+
+AMI
+------------------
+ * Added a new CancelAtxfer action that cancels an attended transfer.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 14 to Asterisk 15 --------------------
+------------------------------------------------------------------------------
+
+app_queue
+------------------
+ * PAUSEALL/UNPAUSEALL now sets the pause reason in the queue_log if it has
+ been defined.
+
+ * A new option, "announce-position-only-up," has been added that, when set to
+ yes, causes position announcements to only be played when the caller's
+ queue position has improved since the last time that we announced their
+ position. This default is no.
+
+Build System
+------------------
+ * '--with-pjproject-bundled' is now the default when running ./configure
+ It can be disabled with '--without-pjproject-bundled'.
+
+ * A '--with-download-cache' option is now available which is equivalent to
+ setting '--with-sounds-cache' and '--with-externals-cache' to the same
+ value. The download cache can also be set via the AST_DOWNLOAD_CACHE
+ environment variable.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 14.6.0 to Asterisk 14.7.0 ------------
+------------------------------------------------------------------------------
+
+res_pjsip
+------------------
+ * The "external_media_address" on transports is now resolved using dnsmgr and
+ when dnsmgr refreshes are enabled will be automatically updated with the new
+ IP address of a given hostname.
+
+ * A new endpoint parameter "incoming_mwi_mailbox" allows Asterisk to receive
+ unsolicited MWI NOTIFY requests and make them available to other modules via
+ the stasis message bus.
+
+res_musiconhold
+------------------
+ * By default, when res_musiconhold reloads or unloads, it sends a HUP signal
+ to custom applications (and all descendants), waits 100ms, then sends a
+ TERM signal, waits 100ms, then finally sends a KILL signal. An application
+ which is interacting with an external device and/or spawns children of its
+ own may not be able to exit cleanly in the default times, expecially if sent
+ a KILL signal, or if it's children are getting signals directly from
+ res_musiconhoild. To allow extra time, the 'kill_escalation_delay'
+ class option can be used to set the number of milliseconds res_musiconhold
+ waits before escalating kill signals, with the default being the current
+ 100ms. To control to whom the signals are sent, the "kill_method"
+ class option can be set to "process_group" (the default, existing behavior),
+ which sends signals to the application and its descendants directly, or
+ "process" which sends signals only to the application itself.
+
+ * New dialplan function PJSIP_DTMF_MODE added to get or change the DTMF mode
+ of a channel on a per-call basis.
+
+res_xmpp
+-----------------
+ * OAuth 2.0 authentication is now supported when contacting Google. Follow the
+ instructions in xmpp.conf.sample to retrieve and configure the necessary
+ tokens.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 14.5.0 to Asterisk 14.6.0 ------------
+------------------------------------------------------------------------------
+
+app_voicemail
+------------------
+ * A new global option "imap_poll_logout" was added to specify whether need to
+ disconnect from the IMAP server after polling of mailboxes.
+ Default: no
+
+res_pjsip
+------------------
+ * A new endpoint option "refer_blind_progress" was added to turn off notifying
+ the progress details on Blind Transfer. If this option is not set then
+ the chan_pjsip will send NOTIFY "200 OK" immediately after "202 Accepted".
+ On default is enabled.
+ Some SIP phones like Mitel/Aastra or Snom keep the line busy until
+ receive "200 OK".
+
+ * A new endpoint option "notify_early_inuse_ringing" was added to control
+ whether to notify dialog-info state 'early' or 'confirmed' on Ringing
+ when already INUSE.
+
+ * The endpoint option 'dtmf_mode' has a new option 'auto_dtmf' added. This
+ mode works similar to 'auto' except uses DTMF INFO as fallback instead of
+ INBAND.
+
+res_agi
+------------------
+ * The EAGI() application will now look for a dialplan variable named
+ EAGI_AUDIO_FORMAT and use that format with the 'enhanced' audio pipe that
+ EAGI provides. If not specified, it will continue to use the default signed
+ linear (slin).
+
+chan_pjsip
+------------------
+ * When dialing an endpoint directly or using the PJSIP_DIAL_CONTACTS dialplan
+ function any contact which is considered unreachable due to qualify being
+ enabled will no longer be called.
+
+ * The asymmetric_rtp_codec option now also controls whether chan_pjsip will
+ send media as-is without transcoding if the codec has been negotiated in the
+ SDP. If set to "no" then Asterisk will only ever send the preferred codec
+ from the SDP, unless the remote side sends a different codec and we will
+ switch to match.
+
+Build System
+------------------
+ * Added a new PJPROJECT_CONFIGURE_OPTS environment variable which can be used
+ to pass arbitrary options to the bundled pjproject configure.
+
+ * Automatically set the bundled pjproject configure --host and --build
+ options to match those supplied for the asterisk configure.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 14.4.0 to Asterisk 14.5.0 ------------
+------------------------------------------------------------------------------
+
+res_rtp_asterisk
+------------------
+ * Added the stun_blacklist option to rtp.conf. Some multihomed servers have
+ IP interfaces that cannot reach the STUN server specified by stunaddr.
+ Blacklist those interface subnets from trying to send a STUN packet to find
+ the external IP address. Attempting to send the STUN packet needlessly
+ delays processing incoming and outgoing SIP INVITEs because we will wait
+ for a response that can never come until we give up on the response.
+ Multiple subnets may be listed.
+
+Logging
+-------------------
+ * Added logger_queue_limit to the configuration options.
+ All log messages go to a queue serviced by a single thread
+ which does all the IO. This setting controls how big that
+ queue can get (and therefore how much memory is allocated)
+ before new messages are discarded.
+ The default is 1000.
+
+res_pjsip_config_wizard
+------------------
+ * Two new parameters have been added to the pjsip config wizard.
+ Setting 'sends_line_with_registrations' to true will cause the wizard
+ to skip the creation of an identify object to match incoming requests
+ to the endpoint and instead add the line and endpoint parameters to
+ the outbound registration object.
+ Setting 'outbound_proxy' is a shortcut for adding individual
+ endpoint/outbound_proxy, aor/outbound_proxy and registration/outbound_proxy
+ parameters.
+
+res_hep_rtcp
+------------------
+ * If the 'call-id' value is specified for the uuid_type option and a
+ chan_sip channel is used the resulting HEP traffic will now contain the
+ SIP Call-ID instead of the Asterisk channel name.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 14.3.0 to Asterisk 14.4.0 ------------
+------------------------------------------------------------------------------
+
+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.
+
+ * RADIUS backends for CEL and CDR can now also be built using the radcli
+ client library, in addition to the existing support for building them
+ using either freeradius or radiusclient-ng.
+
+Core
+------------------
+ * ASTERISK_REGISTER_FILE was no longer useful and has been removed. Sources
+ which use mtx_prof must now manually declare and initialize the variable.
+
+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.
+
+cdr_radius
+------------------
+ * To fix a memory leak the syslog channel is now empty if it has not been set
+ and used by a syslog channel in the logger.
+
+cel_radius
+------------------
+ * To fix a memory leak the syslog channel is now empty if it has not been set
+ and used by a syslog channel in the logger.
+
+RTP
+------------------
+ * New setting "rtp_pt_dynamic = 35" in asterisk.conf:
+ Normally the Dynamic RTP Payload Type numbers are 96-127, which allow just 32
+ formats. To avoid the message "No Dynamic RTP mapping available", the range
+ was changed to 35-63,96-127. This is allowed by RFC 3551 section 3. However,
+ when you use more than 32 formats and calls are not accepted by a remote
+ implementation, please report this and go back to rtp_pt_dynamic = 96.
+
+ * A new setting, "rtp_use_dynamic", has been added in asterisk.conf". When set
+ to "yes" RTP dynamic payload types are assigned dynamically per RTP instance.
+ When set to "no" RTP dynamic payload types are globally initialized to pre-
+ designated numbers and function similar to static payload types.
+
+app_originate
+------------------
+ * Added support to gosub predial routines on both original channel and on the
+ created channel using options parameter (like app_dial) B() and b(). This
+ allows for adding variables to newly created channel or, e.g. setting callerid.
+
+CLI Commands
+------------------
+ * 'dialplan show' output will now show [config_file:line_number] instead of
+ [registrar] when that information is available. Currently only extensions
+ registered by pbx_config when loading/reloading will use this format.
+
+app_queue
+------------------
+ * Add 'QueueUpdate' application which can be used to track outbound calls
+ using app_queue.
+
+pbx_spool
+------------------
+ * Asterisk will now set the AST_OUTGOING_ATTEMPT channel variable so that
+ attempt-specific behavior is possible. This is a 1-based number that
+ simply increases by 1 for each attempt.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 14.3.0 to Asterisk 14.4.0 ------------
+------------------------------------------------------------------------------
+
+AMI
+------------------
+ * The 'PJSIPShowEndpoint' command's respone event of 'IdentifyDetail' now
+ contains a new optional parameter, 'MatchHeader', mapping to the new
+ configuration option 'match_header' for the corresponding 'identify' object.
+ It should be noted that since 'match_header' takes in a key: value pair, the
+ event parameter will contain a ':' as well.
+
+app_record
+------------------
+ * Added new 'u' option to Record() application which prevents Asterisk from
+ truncating silence from the end of recorded files.
+
+res_pjsip_outbound_registration
+------------------
+ * Outbound registrations are now refreshed when res_stun_monitor detects
+ a network change event has happened.
+ The 'pjsip send (un)register' CLI commands were updated to accept '*all'
+ as an argument to operate on all registrations.
+ The 'PJSIP(Un)Register' AMI commands were updated to also accept '*all'.
+
+app_voicemail
+------------------
+ * The 'Comedian Mail' prompts can now be overriden using the 'vm-login' and
+ 'vm-newuser' configuration options in voicemail.conf.
+
+ * Added 'fromstring' field to the voicemail boxes. If set, it will override
+ the global 'fromstring' field on a per-mailbox basis.
+
+func_channel
+------------------
+ * Added CHANNEL(callid) to retrieve the call log tag associated with the
+ channel. e.g., [C-00000000] Dialplan now has access to the call log
+ search key associated with the channel so it can be saved in case there
+ is a problem with the call.
+
+res_pjsip
+------------------
+ * A new transport parameter 'symmetric_transport' has been added.
+ When a request from a dynamic contact comes in on a transport with this
+ option set to 'yes', the transport name will be saved and used for
+ subsequent outgoing requests like OPTIONS, NOTIFY and INVITE. It's
+ saved as a contact uri parameter named 'x-ast-txp' and will display with
+ the contact uri in CLI, AMI, and ARI output. On the outgoing request,
+ if a transport wasn't explicitly set on the endpoint AND the request URI
+ is not a hostname, the saved transport will be used and the 'x-ast-txp'
+ parameter stripped from the outgoing packet. To facilitate recreation of
+ subscriptions on asterisk restart, a new column 'contact_uri' needed to be
+ added to the ps_subcsription_persistence table. Since new columns were
+ added to both transport and subscription_persistence, an alembic upgrade
+ should be run to bring the database tables up to date.
+
+ * A new option, allow_overlap, has been added to endpoints which allows
+ overlap dialing functionality to be enabled or disabled. The option defaults
+ to enabled.
+
+res_pjsip_transport_websocket
+------------------
+ * Removed non-secure websocket support. Firefox and Chrome have not allowed
+ non-secure websockets for quite some time so this shouldn't be an issue
+ for people. Attempting to use a non-secure websocket may or may not work
+ when Asterisk attempts to send SIP requests to do something like initiate
+ call hangup.
+
+res_pjsip_endpoint_identifier_ip
+------------------
+ * A new option has been added to the 'identify' configuration object,
+ 'match_header'. The 'match_header' attribute should contain a SIP
+ header: value pair that, When set, will cause inbound requests that contain
+ the matching SIP header/value pair to be associated with the corresponding
+ endpoint. This option is cumulative with the 'match' option, so that if
+ either option matches the request, the request is associated with the
+ endpoint.
+
+ In a future release, this module will be renamed to something more
+ appropriate, as it now matches inbound requests on more than just IP
+ address.
+
+res_rtp_asterisk
+-----------------
+ * The RTP layer of Asterisk now has support for RFC 5761: "Multiplexing RTP
+ Data and Control Packets on a Single Port." So far, the only channel driver
+ that supports this feature is chan_pjsip. You can set "rtcp_mux = yes" on
+ a PJSIP endpoint in pjsip.conf to enable the feature.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 14.2.0 to Asterisk 14.3.0 ------------
+------------------------------------------------------------------------------
+
+res_pjproject
+------------------
+ * Added new CLI command "pjproject set log level". The new command allows
+ the maximum PJPROJECT log levels to be adjusted dynamically and
+ independently from the set debug logging level like many other similar
+ module debug logging commands.
+
+ * Added new companion CLI command "pjproject show log level" to allow the
+ user to see the current maximum pjproject logging level.
+
+ * Added new pjproject.conf startup section "log_level' option to set the
+ initial maximum PJPROJECT logging level.
+
+res_pjsip_outbound_registration
+------------------
+ * Statsd no longer logs redundant status PJSIP.registrations.state changes
+ for internal state transitions that don't change the reported public status
+ state.
+
+res_pjsip_registrar
+------------------
+ * The PJSIPShowRegistrationInboundContactStatuses AMI command has been added
+ to return ContactStatusDetail events as opposed to
+ PJSIPShowRegistrationsInbound which just a dumps every defined AOR.
+
+res_pjsip
+------------------
+ * Six existing contact fields have been added to the end of the
+ ContactStatusDetail AMI event:
+ ID, AuthenticateQualify, OutboundProxy, Path, QualifyFrequency and
+ QualifyTimeout. Existing fields have not been disturbed.
+
+res_pjsip_endpoint_identifier_ip
+------------------
+ * SRV lookups can now be done on provided hostnames to determine additional
+ source IP addresses for requests. This is configurable using the
+ "srv_lookups" option on the identify and defaults to "yes".
+
+ARI
+------------------
+ * The 'ari set debug' command has been enhanced to accept 'all' as an
+ application name. This allows dumping of all apps even if an app
+ hasn't registered yet.
+
+ * 'ari set debug' now displays requests and responses as well as events.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 14.1.0 to Asterisk 14.2.0 ------------
+------------------------------------------------------------------------------
+
+AMI
+------------------
+ * Events that reference a bridge may now contain two new optional fields:
+ - 'BridgeVideoSourceMode': the video source mode for the bridge.
+ Can be one of 'none', 'talker', or 'single'.
+ - 'BridgeVideoSource': the unique ID of the channel that is the video
+ source in this bridge, if one exists.
+
+ * A new event, BridgeVideoSourceUpdate, has been added with a class
+ authorization of CALL. The event is raised when the video source changes
+ in a multi-party mixing bridge.
+
+ARI
+------------------
+ * The bridges resource now exposes two new operations:
+ - POST /bridges/{bridgeId}/videoSource/{channelId}: Set a video source in a
+ multi-party mixing bridge
+ - DELETE /bridges/{bridgeId}/videoSource: Remove the set video source,
+ reverting to talk detection for the video source
+
+ * The bridge model in any returned response or event now contains the following
+ optional fields:
+ - video_mode: the video source mode for the bridge. Can be one of 'none',
+ 'talker', or 'single'.
+ - video_source_id: the unique ID of the channel that is the video source
+ in this bridge, if one exists.
+
+ * A new event, BridgeVideoSourceChanged, has been added for bridges.
+ Applications subscribed to a bridge will receive this event when the source
+ of video changes in a mixing bridge.
+
+ * The ARI major version has been bumped. There are not any known breaking changes
+ in ARI. The major version has been bumped because otherwise we can end up with
+ overlapping version numbers between different Asterisk versions. Now each major
+ version of Asterisk will bring with it a change in the major version of ARI.
+ The ARI version in Asterisk 14 is now 2.0.0.
+
+res_pjsip
+------------------
+ * Automatic dual stack support is now implemented. Depending on DNS resolution
+ and the transport used for sending a message the SIP signaling and SDP will
+ be updated with the correct IP address and protocol version. This means that
+ the rtp_ipv6 and t38_udptl_ipv6 options no longer have any effect. The
+ res_pjsip_multihomed module has also been moved into core res_pjsip to ensure
+ that messages are updated with the correct address information in all cases.
+
+chan_pjsip
+------------------
+ * The default behavior for RTP codecs has been changed. The sending codec will
+ now match the receiving codec. This can be turned off and behavior reverted
+ to asymmetric using the "asymmetric_rtp_codec" endpoint option. If this
+ option is set then the sending and received codec are allowed to differ.
+
+CLI Commands
+------------------
+ * Three new CLI commands have been added for ARI:
+ - ari show apps:
+ Displays a listing of all registered ARI applications.
+ - ari show app <name>:
+ Display detailed information about a registered ARI application.
+ - ari set debug <name> <on|off>:
+ Enable/disable debugging of an ARI application. When debugged, verbose
+ information will be sent to the Asterisk CLI.
+
+
+Queue
+------------------
+ * A new dialplan variable, ABANDONED, is set when the call is not answered
+ by an agent.
+
+res_ari
+------------------
+ * The configuration file ari.conf now supports a channelvars option, which
+ specifies a list of channel variables to include in each channel-oriented
+ ARI event.
+
+------------------------------------------------------------------------------
+--- 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.
+
+res_rtp_asterisk
+------------------
+ * An option, ice_blacklist, has been added which allows certain subnets to be
+ excluded from local ICE candidates.
+
+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.
+
+res_ari
+------------------
+ * ARI events will all now include a new field in the root of the JSON message,
+ 'asterisk_id'. This will be the unique ID for the Asterisk system
+ transmitting the event. The value can be overridden using the 'entityid'
+ setting in asterisk.conf.
+
+------------------------------------------------------------------------------
+--- 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 alert is generated. A new CLI command
+ "pjsip show unidentified_requests" will list the current candidates.
+
+res_pjsip_history
+------------------
+ * A new module, res_pjsip_history, has been added that provides SIP history
+ viewing/filtering from the CLI. The module is intended to be used on systems
+ with busy SIP traffic, where existing forms of viewing SIP messages - such
+ as the res_pjsip_logger - may be inadequate. The module provides two new
+ CLI commands:
+ - 'pjsip set history {on|off|clear}' - this enables/disables SIP history
+ capturing, as well as clears an existing history capture. Note that SIP
+ packets captured are stored in memory until cleared. As a result, the
+ history capture should only be used for debugging/viewing purposes, and
+ should *NOT* be left permanently enabled on a system.
+ - 'pjsip show history' - displays the captured SIP history. When invoked
+ with no options, the entire captured history is displayed. Two options
+ are available:
+ -- 'entry <num>' - display a detailed view of a single SIP message in
+ the history
+ -- 'where ...' - filter the history based on some expression. For more
+ information on filtering, view the current CLI help for the
+ 'pjsip show history' command.
+
+Voicemail
+------------------
+ * app_voicemail and res_mwi_external can now be built together. The default
+ remains to build app_voicemail and not res_mwi_external but if they are
+ both built, the load order will cause res_mwi_external to load first and
+ app_voicemail will be skipped. Use 'preload=app_voicemail.so' in
+ modules.conf to force app_voicemail to be the voicemail provider.
+
+res_pjsip_sdp_rtp
+------------------
+ * A new option (bind_rtp_to_media_address) has been added to endpoint which
+ will cause res_pjsip_sdp_rtp to actually bind the RTP instance to the
+ media_address as well as using it in the SDP. If set, RTP packets will now
+ originate from the media address instead of the operating system's "primary"
+ ip address.
+
+res_rtp_asterisk
+------------------
+ * A new configuration section - ice_host_candidates - has been added to
+ rtp.conf, allowing automatically discovered ICE host candidates to be
+ overriden. This allows an Asterisk server behind a 1:1 NAT to send its
+ external IP as a host candidate rather than relying on STUN to discover it.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 13.6.0 to Asterisk 13.7.0 ------------
+------------------------------------------------------------------------------
+
+Codecs
+------------------
+ * Added format attribute negotiation for the VP8 video codec. Format attribute
+ negotiation is provided by the res_format_attr_vp8 module.
+
+ConfBridge
+------------------
+ * A new "timeout" user profile option has been added. This configures the number
+ of seconds that a participant may stay in the ConfBridge after joining. When
+ the time expires, the user is ejected from the conference and CONFBRIDGE_RESULT
+ is set to "TIMEOUT" on the channel.
+
+chan_sip
+------------------
+ * The websockets_enabled option has been added to the general section of
+ sip.conf. The option is enabled by default to match the previous behavior.
+ The option should be disabled when using res_pjsip_transport_websockets to
+ ensure chan_sip will not conflict with PJSIP websockets.
+
+Dialplan Functions
+------------------
+ * The HOLD_INTERCEPT dialplan function now actually exists in the source tree.
+ While support for the events was added in Asterisk 13.4.0, the function
+ accidentally never made it in. That function is now present, and will cause
+ the 'hold' raised by a channel to be intercepted and converted into an
+ event instead.
+
+res_pjsip_outbound_registration
+-------------------------------
+ * If res_statsd is loaded and a StatsD server is configured, basic statistics
+ regarding the state of outbound registrations will now be emitted. This
+ includes:
+ - A GAUGE statistic for the overall number of outbound registrations, i.e.:
+ PJSIP.registrations.count
+ - A GAUGE statistic for the overall number of outbound registrations in a
+ particular state, e.g.:
+ PJSIP.registrations.state.Registered
+
+res_pjsip
+------------------
+ * The ability to use "like" has been added to the pjsip list and show
+ CLI commands. For instance: CLI> pjsip list endpoints like abc
+
+ * If res_statsd is loaded and a StatsD server is configured, basic statistics
+ regarding the state of PJSIP contacts will now be emitted. This includes:
+ - A GAUGE statistic for the overall number of contacts in a particular
+ state, e.g.:
+ PJSIP.contacts.states.Reachable
+ - A TIMER statistic for the RTT time for each qualified contact, e.g.:
+ PJSIP.contacts.alice@@127.0.0.1:5061.rtt
+
+res_sorcery_memory_cache
+------------------------
+ * A new caching strategy, full_backend_cache, has been added which caches
+ all stored objects in the backend. When enabled all objects will be
+ expired or go stale according to the configuration. As well when enabled
+ all retrieval operations will be performed against the cache instead of
+ the backend.
+
+func_callerid
+-------------------
+ * CALLERID(pres) is now documented as a valid alternative to setting both
+ CALLERID(name-pres) and CALLERID(num-pres) at once. Some channel drivers,
+ like chan_sip, don't make a distinction between the two: they take the
+ least public value from name-pres and num-pres. By using CALLERID(pres)
+ for reading and writing, you touch the same combined value in the dialplan.
+ The same applies to CONNECTEDLINE(pres), REDIRECTING(orig-pres),
+ REDIRECTING(to-pres) and REDIRECTING(from-pres).
+
+res_endpoint_stats
+-------------------
+ * A new module that emits StatsD statistics regarding Asterisk endpoints.
+ This includes a total count of the number of endpoints, the count of the
+ number of endpoints in the technology agnostic state of the endpoint -
+ online or offline - as well as the number of channels associated with each
+ endpoint. These are recorded as three different GAUGE statistics:
+ - endpoints.count
+ - endpoints.state.{unknown|offline|online}
+ - endpoints.{tech}.{resource}.channels
+
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 13.5.0 to Asterisk 13.6.0 ------------
+------------------------------------------------------------------------------
+
+Dialplan Functions
+------------------
+ * The CHANNEL function, when used on a PJSIP channel, now exposes a 'call-id'
+ extraction option when using with the 'pjsip' signalling option. It will
+ return the SIP Call-ID associated with the INVITE request that established
+ the PJSIP channel.
+
+ARI
+------------------
+ * Two new endpoint related events are now available: PeerStatusChange and
+ ContactStatusChange. In particular, these events are useful when subscribing
+ to all event sources, as they provide additional endpoint related
+ information beyond the addition/removal of channels from an endpoint.
+
+ * Added the ability to subscribe to all ARI events in Asterisk, regardless
+ of whether the application 'controls' the resource. This is useful for
+ scenarios where an ARI application merely wants to observe the system,
+ as opposed to control it. There are two ways to accomplish this:
+ (1) Via the WebSocket connection URI. A new query paramter, 'subscribeAll',
+ has been added that, when present and True, will subscribe all
+ specified applications to all ARI event sources in Asterisk.
+ (2) Via the applications resource. An ARI client can, at any time, subscribe
+ to all resources in an event source merely by not providing an explicit
+ resource. For example, subscribing to an event source of 'channels:'
+ as opposed to 'channels:12345' will subscribe the application to all
+ channels.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 13.4.0 to Asterisk 13.5.0 ------------
+------------------------------------------------------------------------------
+
+AMI
+------------------
+ * A new ContactStatus event has been added that reflects res_pjsip contact
+ lifecycle changes: Created, Removed, Reachable, Unreachable, Unknown.
+
+ * Added the Linkedid header to the common channel headers listed for each
+ channel in AMI events.
+
+ARI
+------------------
+ * A new feature has been added that enables the retrieval of modules and
+ module information through an HTTP request. Information on a single module
+ can be also be retrieved. Individual modules can be loaded to Asterisk, as
+ well as unloaded and reloaded.
+
+* A new resource has been added to the 'asterisk' resource, 'config/dynamic'.
+ This resource allows for push configuration of sorcery derived objects
+ within Asterisk. The resource supports creation, retrieval, updating, and
+ deletion. Sorcery derived objects that are manipulated by this resource
+ must have a sorcery wizard that supports the desired operations.
+
+ * A new feature has been added that allows for the rotation of log channels
+ through HTTP requests.
+
+
+res_pjsip
+------------------
+* A new 'g726_non_standard' endpoint option has been added that, when set to
+ 'yes' and g.726 audio is negotiated, forces the codec to be treated as if it
+ is AAL2 packed on the channel.
+
+* A new 'rtp_keepalive' endpoint option has been added. This option specifies
+ an interval, in seconds, at which we will send RTP comfort noise packets to
+ the endpoint. This functions identically to chan_sip's "rtpkeepalive" option.
+
+* New 'rtp_timeout' and 'rtp_timeout_hold' endpoint options have been added.
+ These options specify the amount of time, in seconds, that Asterisk will wait
+ before terminating the call due to lack of received RTP. These are identical
+ to chan_sip's rtptimeout and rtpholdtimeout options.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 13.3.0 to Asterisk 13.4.0 ------------
+------------------------------------------------------------------------------
+
+chan_pjsip
+------------------
+ * New 'rpid_immediate' option to control if connected line update information
+ goes to the caller immediately or waits for another reason to send the
+ connected line information update. See the online option documentation for
+ more information. Defaults to 'no' as setting it to 'yes' can result in
+ many unnecessary messages being sent to the caller.
+
+ * The configuration setting 'progressinband' now defaults to 'no', which
+ matches the actual behavior of previous versions.
+
+res_pjsip
+------------------
+ * A new CLI command has been added: "pjsip show settings", which shows
+ both the global and system configuration settings.
+
+ * A new aor option has been added: "qualify_timeout", which sets the timeout
+ in seconds for a qualify. The default is 3 seconds. This overrides the
+ hard coded 32 seconds in pjproject.
+
+ * Endpoint status will now change to "Unreachable" when all contacts are
+ unavailable. When any contact becomes available, the endpoint will status
+ will change back to "Reachable".
+
+ * A new global option has been added: "max_initial_qualify_time", which
+ sets the maximum amount of time from startup that qualifies should be
+ attempted on all contacts.
+
+res_ari_channels
+------------------
+ * Two new events, 'ChannelHold' and 'ChannelUnhold', have been added to the
+ events data model. These events are raised when a channel indicates a hold
+ or unhold, respectively.
+
+func_holdintercept
+------------------
+ * A new dialplan function, HOLD_INTERCEPT, has been added. This function, when
+ placed on a channel, intercepts hold/unhold indications signalled by the
+ channel and prevents them from moving on to other channels in a bridge with
+ the hold initiator. Instead, AMI or ARI events are raised indicating that
+ the channel wanted to place someone on hold. This allows external
+ applications to implement their own custom hold/unhold logic.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 13.2.0 to Asterisk 13.3.0 ------------
+------------------------------------------------------------------------------
+
+chan_pjsip/app_transfer
+------------------
+ * The Transfer application, when used with chan_pjsip, now supports using
+ a PJSIP endpoint as the transfer destination. This is in addition to
+ explicitly specifying a SIP URI to transfer to.
+
+res_ari_channels
+------------------
+ * The ARI /channels resource now supports a new operation, 'redirect'. The
+ redirect operation will perform a technology and state specific redirection
+ on the channel to a specified endpoint or destination. In the case of SIP
+ technologies, this is either a 302 Redirect response to an on-going INVITE
+ dialog or a SIP REFER request.
+
+res_pjsip
+------------------
+ * A new 'endpoint_identifier_order' option has been added that allows one to
+ set the order by which endpoint identifiers are processed and checked. This
+ option is specified under the 'global' type configuration section.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 13.1.0 to Asterisk 13.2.0 ------------
+------------------------------------------------------------------------------
+
+ * New 'PJSIP_AOR' and 'PJSIP_CONTACT' dialplan functions have been added which
+ allow examining PJSIP AORs or contacts from the dialplan.
+
+res_pjsip_outbound_registration
+------------------
+ * The 'pjsip send unregister' command now stops further registrations.
+
+ * A new command 'pjsip send register' has been added which allows you to
+ start or restart periodic registration. It can be used after a
+ 'send unregister' or after a 401 permanent error.
+
+res_pjsip_config_wizard
+------------------
+ * This is a new module that adds streamlined configuration capability for
+ chan_pjsip. It's targeted at users who have lots of basic configuration
+ scenarios like 'phone' or 'agent' or 'trunk'. Additional information
+ can be found in the sample configuration file at
+ config/samples/pjsip_wizard.conf.sample.
+
+res_fax
+-----------
+ * The T.38 negotiation timeout was previously hard coded at 5000 milliseconds
+ and is now configurable via the 't38timeout' configuration option in
+ res_fax.conf and via the fax options dialplan function 'FAXOPT(t38timeout)'.
+ The default remains at 5000 milliseconds.
+
+PJSIP Transports
+----------
+ * The ca_list_path transport parameter has been added for TLS transports. This
+ option behaves similarly to the old sip.conf option "tlscapath". In order to
+ use this, you must be using PJProject version 2.4 or higher.
+
+ARI
+------------------
+ * The Originate operation now takes in an originator channel. The linked ID of
+ this originator channel is applied to the newly originated outgoing channel.
+ If using CEL this allows an association to be established between the two so
+ it can be recognized that the originator is dialing the originated channel.
+
+ * "language" (the default spoken language for the channel) is now included in
+ the standard channel state output for suitable events.
+
+ * The POST channels/{id} operation and the POST channels/{id}/continue operation
+ now have a new "label" parameter. This allows for origination or continuation
+ to a labeled priority in the dialplan instead of requiring a specific priority
+ number. The ARI version has been bumped to 1.7.0 as a result.
+
+AMI
+------------------
+ * "Language" (the default spoken language for the channel) is now included in
+ the standard channel state output for suitable events.
+
+ * AMI actions that return a list of events have been made to return consistent
+ headers for the action response event starting the list and the list complete
+ event. The AMI version has been bumped to 2.7.0 as a result.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 13.0.0 to Asterisk 13.1.0 ------------
+------------------------------------------------------------------------------
+
+AMI
+------------------
+ * Event NewConnectedLine is emitted when the connected line information on
+ a channel changes.
+
+ARI
+------------------
+ * Event ChannelConnectedLine is emitted when the connected line information
+ on a channel changes.
+
+Core Transfers
+-----------------
+
+The features.conf general section has three new configurable options:
+ * transferdialattempts
+ * transferretrysound
+ * transferinvalidsound
+For more information on what these options do, see the Asterisk wiki:
+ https://wiki.asterisk.org/wiki/x/W4fAAQ
+
+Channel Drivers
+------------------
+
+chan_pjsip
+------------------
+ * New 'media_encryption_optimistic' endpoint setting. This will use SRTP
+ when possible but does not consider lack of it a failure.
+
+res_pjsip_endpoint_identifer_ip
+------------------
+ * New CLI commands have been added: "pjsip show identif(y|ies)", which lists
+ all configured PJSIP identify objects
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 12 to Asterisk 13 --------------------
+------------------------------------------------------------------------------
+
+Overview
+------------------
+
+Asterisk 13 is the next Long Term Support (LTS) release of Asterisk. As such,
+the focus of development for this release of Asterisk was on improving the
+usability and features developed in the previous Standard release, Asterisk 12.
+Beyond a general refinement of end user features, development focussed heavily
+on the Asterisk APIs - the Asterisk Manager Interface (AMI) and the Asterisk
+REST Interface (ARI) - and the PJSIP stack in Asterisk. Some highlights of the
+new features include:
+
+* Asterisk security events are now provided via AMI, allowing end users to
+ monitor their Asterisk system in real time for security related issues.
+* External control of Message Waiting Indicators (MWI) through both AMI and ARI.
+* Reception/transmission of out of call text messages using any supported
+ channel driver/protocol stack through ARI.
+* Resource List Server support in the PJSIP stack, providing subscriptions to
+ lists of resources and batched delivery of NOTIFY requests.
+* Inter-Asterisk distributed device state and mailbox state using the PJSIP
+ stack.
+
+It is important to note that Asterisk 13 is built on the architecture developed
+during the previous Standard release, Asterisk 12. Users upgrading to
+Asterisk 13 should read about the new features in Asterisk 12 later in this file
+(see Functionality changes from Asterisk 11 to Asterisk 12), as well as the
+UPGRADE-12.txt delivered with this release. In particular, users upgrading to
+Asterisk 13 from a release prior to Asterisk 12 should read the specifications
+on AMI, CDRs, and CEL on the Asterisk wiki:
+ * AMI - https://wiki.asterisk.org/wiki/x/dAFRAQ
+ * CEL - https://wiki.asterisk.org/wiki/x/4ICLAQ
+ * CDRs - https://wiki.asterisk.org/wiki/x/pwpRAQ
+
+Many new featuers in Asterisk 13 were introduced in point releases of
+Asterisk 12. Following this section - which documents the changes from all
+versions of Asterisk 12 to Asterisk 13 - users should examine the new features
+that were introduced in the point releases of Asterisk 12, as they are also
+included in Asterisk 13.
+
+Finally, all users upgrading to Asterisk 13 should read the UPGRADE.txt file
+delivered with this release.
+
+
+Build System
+------------------
+ * Sample config files have been moved from configs/ to a sub-folder of that
+ directory, samples.
+
+ * The menuselect utility has been pulled into the Asterisk repository. As a
+ result, the libxml2 development library is now a required dependency for
+ Asterisk.
+
+ * A new Compiler Flag, REF_DEBUG, has been added. When enabled, reference
+ counted objects will emit additional debug information to the refs log file
+ located in the standard Asterisk log file directory. This log file is useful
+ in tracking down object leaks and other reference counting issues. Prior to
+ this version, this option was only available by modifying the source code
+ directly. This change also includes a new script, refcounter.py, in the
+ contrib folder that will process the refs log file. Note that this replaces
+ the refcounter utility that could be built from the utils directory.
+
+
+Applications
+------------------
+
+DahdiBarge
+------------------
+ * This module was deprecated and has been removed. Users of app_dahdibarge
+ should use ChanSpy instead.
+
+MixMonitor
+------------------
+ * New options to play a beep when starting a recording and stopping a recording
+ have been added. The option "p" will play a beep to the channel that starts
+ the recording. The option "P" will play a beep to the channel that stops the
+ recording.
+
+Queue
+------------------
+ * Queue rules can now be stored in a database table, queue_rules. Unlike other
+ RealTime tables, the queue_rules table is only examined on module load or
+ module reload. A new general setting has been added to queuerules.conf,
+ 'realtime_rules', which, when set to 'yes', will cause app_queue to look in
+ RealTime for additional queue rules to parse. Note that both the file and
+ the database can be used as a provide of queue rules when 'realtime_rules'
+ is set to 'yes'.
+
+ When app_queue is reloaded, all rules are re-parsed and loaded into memory.
+ There is no caching of RealTime queue rules.
+
+ReadFile
+------------------
+ * This module was deprecated and has been removed. Users of app_readfile
+ should use func_env's FILE function instead.
+
+Say
+------------------
+ * The 'say' family of dialplan applications now support the Japanese
+ language. The 'language' parameter in say.conf now recognizes a setting of
+ 'ja', which will enable Japanese language specific mechanisms for playing
+ back numbers, dates, and other items.
+ * Counting, enumeration and dates now supports Icelandic grammar with the
+ 'language' parameter set to 'is'.
+
+SayCountPL
+------------------
+ * This module was deprecated and has been removed. Users of app_saycountpl
+ should use the Say family of applications.
+
+SetMusicOnHold
+------------------
+ * The SetMusicOnHold dialplan application was deprecated and has been removed.
+ Users of the application should use the CHANNEL function's musicclass
+ setting instead.
+
+WaitMusicOnHold
+------------------
+ * The WaitMusicOnHold dialplan application was deprecated and has been
+ removed. Users of the application should use MusicOnHold with a duration
+ parameter instead.
+
+VoiceMail
+------------------
+ * VoiceMail and VoiceMailMain now support the Japanese language. The
+ 'language' parameter in voicemail.conf now recognizes a setting of 'ja',
+ which will enable prompts to be played back using a Japanese grammatical
+ structure. Additional prompts are necessary for this functionality,
+ including:
+ - jb-arimasu: there is
+ - jb-arimasen: there is not
+ - jb-oshitekudasai: please press
+ - jb-ni: article ni
+ - jb-ga: article ga
+ - jb-wa: article wa
+ - jb-wo: article wo
+
+ * Add the ability to specify multiple email addresses in configuration,
+ separated by a |.
+
+
+CDR Backends
+------------------
+
+cdr_sqlite
+-----------------
+ * This module was deprecated and has been removed. Users of cdr_sqlite
+ should use cdr_sqlite3_custom.
+
+cdr_pgsql
+------------------
+ * Added the ability to support PostgreSQL application_name on connections.
+ This allows PostgreSQL to display the configured name in the
+ pg_stat_activity view and CSV log entries. This setting is configurable
+ for cdr_pgsql via the appname configuration setting in cdr_pgsql.conf.
+
+
+CEL Backends
+------------------
+
+cel_pgsql
+------------------
+ * Added the ability to support PostgreSQL application_name on connections.
+ This allows PostgreSQL to display the configured name in the
+ pg_stat_activity view and CSV log entries. This setting is configurable
+ for cel_pgsql via the appname configuration setting in cel_pgsql.conf.
+
+
+Channel Drivers
+------------------
+
+chan_dahdi
+------------------
+ * SS7 support now requires libss7 v2.0 or later.
+
+ * Added SS7 support for connected line and redirecting.
+
+ * Most SS7 CLI commands are reworked as well as new SS7 commands added.
+ See online CLI help.
+
+ * Added several SS7 config option parameters described in
+ chan_dahdi.conf.sample.
+
+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 the CLI command 'pjsip list ciphers' so a user can know what
+ OpenSSL names are available on their system for the pjsip.conf cipher
+ option.
+
+chan_sip
+------------------
+ * 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.
+
+
+Core
+------------------
+
+Account Codes
+------------------
+ * Added functional peeraccount support. Except for Queue, the
+ accountcode propagation is now consistently propagated to outgoing
+ channels before dialing. The channel accountcode can change from its
+ original non-empty value on channel creation for the following specific
+ reasons. One, dialplan sets it using CHANNEL(accountcode). Two, an
+ originate method that can specify an accountcode value. Three, the
+ calling channel propagates its peeraccount or accountcode to the
+ outgoing channel's accountcode before dialing. The change has two
+ visible effects. One, local channels now cross accountcode and
+ peeraccount across the special bridge between the ;1 and ;2 channels
+ just like channels between normal bridges. Two, the
+ CHANNEL(peeraccount) value can now be set before Dial and FollowMe to
+ set the accountcode on the outgoing channel(s).
+
+ For Queue, an outgoing channel's non-empty accountcode will not change
+ unless explicitly set by CHANNEL(accountcode). The change has three
+ visible effects. One, local channels now cross accountcode and
+ peeraccount across the special bridge between the ;1 and ;2 channels
+ just like channels between normal bridges. Two, the queue member will
+ get an accountcode if it doesn't have one and one is available from the
+ calling channel's peeraccount. Three, accountcode propagation includes
+ local channel members where the accountcodes are propagated early
+ enough to be available on the ;2 channel.
+
+AMI
+------------------
+ * New DeviceStateChanged and PresenceStateChanged AMI events have been added.
+ These events are emitted whenever a device state or presence state change
+ occurs. The events are controlled by res_manager_device_state.so and
+ res_manager_presence_state.so. If the high frequency of these events is
+ problematic for you, do not load these modules.
+
+ * Added DialplanExtensionAdd and DialplanExtensionRemove AMI commands. They
+ work in basically the same way as the 'dialplan add extension' and
+ 'dialplan remove extension' CLI commands respectively.
+
+ * New AMI action LoggerRotate reloads and rotates logger in the same manner
+ as CLI command 'logger rotate'
+
+ * New AMI Actions FAXSessions, FAXSession, and FAXStats replicate the
+ functionality of CLI commands 'fax show sessions', 'fax show session',
+ and fax show stats' respectively.
+
+ * New AMI actions PRIDebugSet, PRIDebugFileSet, and PRIDebugFileUnset
+ enable manager control over PRI debugging levels and file output.
+
+ * AMI action PJSIPNotify may now send to a URI instead of only to a PJSIP
+ endpoint as long as a default outbound endpoint is set. This also applies
+ to the equivalent CLI command (pjsip send notify)
+
+ * The AMI action PJSIPShowEndpoint now includes ContactStatusDetail sections
+ that give information on Asterisk's attempts to qualify the endpoint.
+
+ * The DialEnd event will now contain a Forward header if the dial is ending
+ due to the call being forwarded. The contents of the Forward header is the
+ extension in the number to which the call is being forwarded.
+
+CEL
+------------------
+ * The "bridge_technology" extra field key has been added to BRIDGE_ENTER
+ and BRIDGE_EXIT events.
+
+Features
+------------------
+ * Channel variables are now substituted in arguments passed to applications
+ run by using dynamic features.
+
+TLS
+------------------
+ * The TLS core 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:
+ tlscipher=AES128-SHA:DES-CBC3-SHA
+ - Ephemeral DH (DHE) is disabled by default. To enable it, add DH parameters
+ into the private key file, e.g., sip.conf tlsprivatekey. For example, the
+ default dh2048.pem - see
+ http://www.opensource.apple.com/source/OpenSSL098/OpenSSL098-35.1/src/apps/dh2048.pem?txt
+ - 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:
+ tlscipher=AES128+kEECDH:AES128+kEDH:3DES+kEDH:AES128-SHA:DES-CBC3-SHA:-ADH:-AECDH
+ will use PFS when offered by the client. Clients which do not offer PFS
+ fall-back to AES-128 (or even 3DES, as recommended by RFC 3261).
+
+
+Functions
+------------------
+
+JACK_HOOK
+------------------
+ * The JACK_HOOK function now supports audio with a sample rate higher than
+ 8kHz.
+
+
+Resources
+------------------
+
+res_config_pgsql
+------------------
+ * Added the ability to support PostgreSQL application_name on connections.
+ This allows PostgreSQL to display the configured name in the
+ pg_stat_activity view and CSV log entries. This setting is configurable
+ for res_config_pgsql via the dbappname configuration setting in
+ res_pgsql.conf.
+
+res_pjsip_outbound_publish
+------------------
+ * A new module, res_pjsip_outbound_publish provides the mechanisms for sending
+ PUBLISH requests for specific event packages to another SIP User Agent.
+
+res_pjsip_pubsub
+------------------
+ * The publish/subscribe core module has been updated to support RFC 4662
+ Resource Lists, allowing Asterisk to act as a Resource List Server (RLS).
+ Resource lists are configured in pjsip.conf under a new object type,
+ resource_list. Resource lists can contain either message-summary or presence
+ events, and can be composed of specific resources that provide the event or
+ other resource lists.
+
+ * Inbound publication support is provided by a new object, inbound-publication.
+ This configures res_pjsip_pubsub to accept PUBLISH requests from a particular
+ resource. Which events are accepted is constructed dynamically; see
+ res_pjsip_publish_asterisk for more information.
+
+res_pjsip_publish_asterisk
+------------------
+ * A new module, res_pjsip_publish_asterisk adds support for PUBLISH requests of
+ Asterisk information to other Asterisk servers. This module is intended only
+ for Asterisk to Asterisk exchanges of information. Currently, this includes
+ both mailbox state and device state information.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 12.4.0 to Asterisk 12.5.0 ------------
+------------------------------------------------------------------------------
+
+ARI
+------------------
+ * Stored recordings now support a new operation, copy. This will take an
+ existing stored recording and copy it to a new location in the recordings
+ directory.
+
+ * LiveRecording objects now have three additional fields that can be reported
+ in a RecordingFinished ARI event:
+ - total_duration: the duration of the recording
+ - talking_duration: optional. The duration of talking detected in the
+ recording. This is only available if max_silence_seconds was specified
+ when the recording was started.
+ - silence_duration: optional. The duration of silence detected in the
+ recording. This is only available if max_silence_seconds was specified
+ when the recording was started.
+ Note that all duration values are reported in seconds.
+
+ * Users of ARI can now send and receive out of call text messages. Messages
+ can be sent directly to a particular endpoint, or can be sent to the
+ endpoints resource directly and inferred from the URI scheme. Text
+ messages are passed to ARI clients as TextMessageReceived events. ARI
+ clients can choose to receive text messages by subscribing to the particular
+ endpoint technology or endpoints that they are interested in.
+
+ * The applications resource now supports subscriptions to all endpoints of
+ a particular channel technology. For example, subscribing to an eventSource
+ of 'endpoint:PJSIP' will subscribe to all PJSIP endpoints.
+
+res_pjsip
+------------------
+ * The endpoint configuration object now supports 'accountcode'. Any channel
+ created for an endpoint with this setting will have its accountcode set
+ to the specified value.
+
+res_hep_rtcp
+------------------
+ * A new module, res_hep_rtcp, has been added that will forward RTCP call
+ statistics to a HEP capture server. See res_hep for more information.
+
+Functions
+------------------
+ * Function AUDIOHOOK_INHERIT has been deprecated. Audiohooks are now
+ unconditionally inherited through masquerades. As a side benefit, more
+ than one audiohook of a given type may persist through a masquerade now.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 12.3.0 to Asterisk 12.4.0 ------------
+------------------------------------------------------------------------------
+
+AgentRequest
+------------------
+ * Returns new AGENT_STATUS value "NOT_CONNECTED" if the agent fails to
+ connect with an incoming caller after being alerted to the presence
+ of the incoming caller. The most likely reason this would happen is
+ the agent did not acknowledge the call in time.
+
+AMI
+------------------
+ * New events have been added for the TALK_DETECT function. When the function
+ is used on a channel, ChannelTalkingStart/ChannelTalkingStop events will be
+ emitted to connected AMI clients indicating the start/stop of talking on
+ the channel.
+
+ARI
+------------------
+ * New event models have been aded for the TALK_DETECT function. When the
+ function is used on a channel, ChannelTalkingStarted/ChannelTalkingFinished
+ events will be emitted to connected WebSockets subscribed to the channel,
+ indicating the start/stop of talking on the channel.
+
+Functions
+------------------
+ * A new function, TALK_DETECT, has been added. When set on a channel, this
+ fucntion causes events indicating the starting/stoping of talking on said
+ channel to be emitted to both AMI and ARI clients.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 12.2.0 to Asterisk 12.3.0 ------------
+------------------------------------------------------------------------------
+
+ARI
+------------------
+ * A new Playback URI 'tone' has been added. Tones are specified either as
+ an indication name (e.g. 'tone:busy') from indications.conf or as a tone
+ pattern (e.g. 'tone:240/250,0/250'). Tones differ from normal playback
+ URIs in that they must be stopped manually and will continue to occupy
+ a channel's ARI control queue until they are stopped. They also can not
+ be rewound or fastforwarded.
+
+ * User events can now be generated from ARI. Events can be signalled with
+ arbitrary json variables, and include one or more of channel, bridge, or
+ endpoint snapshots. An application must be specified which will receive
+ the event message (other applications can subscribe to it). The message
+ will also be delivered via AMI provided a channel is attached. Dialplan
+ generated user event messages are still transmitted via the channel, and
+ will only be received by a stasis application they are attached to or if
+ the channel is subscribed to.
+
+chan_sip
+-----------
+ * SIP peers can now specify 'trust_id_outbound' which affects RPID/PAI
+ fields for prohibited callingpres information. Values are legacy, no, and
+ yes. By default, legacy is used.
+ trust_id_outbound=legacy - behavior remains the same as 1.8.26.1. When
+ dealing with prohibited callingpres and sendrpid=pai/rpid, RPID/PAI
+ headers are appended to outbound SIP messages just as they are with
+ allowed callingpres values, but data about the remote party's identity is
+ anonymized.
+ When sendrpid=rpid, only the remote party's domain is anonymized.
+ trust_id_outbound=no - when dealing with prohibited callingpres, RPID/PAI
+ headers are not sent.
+ trust_id_outbound=yes - RPID/PAI headers are applied with the full remote
+ party information in tact even for prohibited callingpres information.
+ In the case of PAI, a Privacy: id header will be appended for prohibited
+ calling information to communicate that the private information should
+ not be relayed to untrusted parties.
+
+res_parking
+------------------
+ * Manager action 'Park' now takes an additional argument 'AnnounceChannel'
+ which can be used to announce the parked call's location to an arbitrary
+ channel in a bridge. If 'Channel' and 'TimeoutChannel' are now the two
+ parties in a one to one bridge, 'TimeoutChannel' is treated as having
+ parked 'Channel' like with the Park Call DTMF feature and will receive
+ announcements prior to being hung up.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 12.1.0 to Asterisk 12.2.0 ------------
+------------------------------------------------------------------------------
+
+Record
+------------------
+ * Record application now has an option 'o' which allows 0 to act as an exit
+ key setting the RECORD_STATUS variable to 'OPERATOR' instead of 'DTMF'
+
+ChanSpy
+--------------------------
+ * ChanSpy now accepts a channel uniqueid or a fully specified channel name
+ as the chanprefix parameter if the 'u' option is specified.
+
+ConfBridge
+--------------------------
+ * CONFBRIDGE dialplan function is now capable of creating/modifying dynamic
+ conference user menus.
+
+ * CONFBRIDGE dialplan function is now capable of removing dynamic conference
+ menus, bridge settings, and user settings that have been applied by the
+ CONFBRIDGE dialplan function.
+
+ * The ConfBridge dialplan application now sets a channel variable,
+ CONFBRIDGE_RESULT, upon exiting. This variable can be used to determine
+ how a channel exited the conference.
+
+ * Added conference user option 'announce_join_leave_review'. This option
+ implies 'announce_join_leave' with the added effect that the user will
+ be asked if they want to confirm or re-record the recording of their
+ name when entering the conference
+
+Directory
+--------------------------
+ * At exit, the Directory application now sets a channel variable
+ DIRECTORY_RESULT to one of the following based on the reason for exiting:
+ OPERATOR user requested operator by pressing '0' for operator
+ ASSISTANT user requested assistant by pressing '*' for assistant
+ TIMEOUT user pressed nothing and Directory stopped waiting
+ HANGUP user's channel hung up
+ SELECTED user selected a user from the directory and is routed
+ USEREXIT user pressed '#' from the selection prompt to exit
+ FAILED directory failed in a way that wasn't accounted for. Dang.
+
+Monitor
+------------------
+ * Monitor() - A new option, B(), has been added that will turn on a periodic
+ beep while the call is being recorded.
+
+MusicOnHold
+--------------------------
+ * MusicOnHold streams (all modes other than "files") now support wide band
+ audio too.
+
+Page
+--------------------------
+ * Added options 'b' and 'B' to apply predial handlers for outgoing calls
+ and for the channel executing Page respectively.
+
+PickupChan
+--------------------------
+ * PickupChan now accepts channel uniqueids of channels to pickup.
+
+Say
+--------------------------
+ * If a channel variable SAY_DTMF_INTERRUPT is present on a channel and set
+ to 'true' (case insensitive), then any Say application (SayNumber,
+ SayDigits, SayAlpha, SayAlphaCase, SayUnixTime, and SayCounted) will
+ anticipate DTMF. If DTMF is received, these applications will behave like
+ the background application and jump to the received extension once a match
+ is established or after a short period of inactivity.
+
+MixMonitor
+-------------------------
+ * A new function, MIXMONITOR, has been added to allow access to individual
+ instances of MixMonitor on a channel.
+
+ * A new option, B(), has been added that will turn on a periodic beep while the
+ call is being recorded.
+
+
+Channel Drivers
+-------------------------
+
+chan_sip
+-------------------------
+ * TEL URI support for inbound INVITE requests has been added. chan_sip will
+ now handle TEL schemes in the Request and From URIs. The phone-context in
+ the Request URI will be stored in the SIPURIPHONECONTEXT channel variable on
+ the inbound channel.
+
+Core
+------------------
+ * Exposed sorcery-based configuration files like pjsip.conf to dialplans via
+ the new AST_SORCERY diaplan function.
+
+ * Core Show Locks output now includes Thread/LWP ID if the platform
+ supports this feature.
+
+ * New "logger add channel" and "logger remove channel" CLI commands have
+ been added to allow creation and deletion of dynamic logger channels
+ without configuration changes. These dynamic logger channels will only
+ exist until the next restart of asterisk.
+
+ARI
+------------------
+ * The live recording object on recording events now contains a target_uri
+ field which contains the URI of what is being recorded.
+
+ * The bridge type used when creating a bridge is now a comma separated list of
+ bridge properties. Valid options are: mixing, holding, dtmf_events, and
+ proxy_media.
+
+ * A channelId can now be provided when creating a channel, either in the
+ uri (POST channels/my-channel-id) or as query parameter. A local channel
+ will suffix the second channel id with ';2' unless provided as query
+ parameter otherChannelId.
+
+ * A bridgeId can now be provided when creating a bridge, either in the uri
+ (POST bridges/my-bridge-id) or as a query parameter.
+
+ * A playbackId can be provided when starting a playback, either in the uri
+ (POST channels/my-channel-id/play/my-playback-id /
+ POST bridges/my-bridge-id/play/my-playback-id) or as a query parameter.
+
+ * A snoop channel can be started with a snoopId, in the uri or query.
+
+AMI
+------------------
+ * Originate now takes optional parameters ChannelId and OtherChannelId,
+ used to set the UniqueId on creation. The other id is assigned to the
+ second channel when dialing LOCAL, or defaults to appending ;2 if only
+ the single Id is given.
+
+ * The Mixmonitor action now has a "Command" header that can be used to
+ indicate a post-process command to run once recording finishes.
+
+RealTime
+------------------
+ * 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.
+
+
+Functions
+------------------
+ * A new function was added: PERIODIC_HOOK. This allows running a periodic
+ dialplan hook on a channel. Any audio generated by this hook will be
+ injected into the call.
+
+
+Resources
+------------------
+
+res_hep
+------------------
+ * A new module, res_hep, has been added, that acts as a generic packet
+ capture agent for the Homer Encapsulation Protocol (HEP) version 3.
+ It can be configured via hep.conf. Other modules can use res_hep to send
+ message traffic to a HEP capture server.
+
+res_hep_pjsip
+------------------
+ * A new module, res_hep_pjsip, has been added that will forward PJSIP
+ message traffic to a HEP capture server. See res_hep for more
+ information.
+
+res_pjsip
+------------------
+ * transport and endpoint ToS options (tos, tos_audio, and tos_video) may now
+ be set as the named set of ToS values (cs0-cs7, af11-af43, ef).
+
+ * Added the following new CLI commands:
+ - "pjsip show contacts" - list all current PJSIP contacts.
+ - "pjsip show contact" - show specific information about a current PJSIP
+ contact.
+ - "pjsip show channel" - show detailed information about a PJSIP channel.
+
+res_pjsip_multihomed
+------------------
+ * A new module, res_pjsip_multihomed handles situations where the system
+ Asterisk is running out has multiple interfaces. res_pjsip_multihomed
+ determines which interface should be used during message sending.
+
+res_pjsip_pidf_digium_body_supplement
+------------------
+ * A new module, res_pjsip_pidf_digium_body_supplement provides NOTIFY
+ request body formatting for presence support in Digium phones.
+
+res_pjsip_send_to_voicemail
+------------------
+ * A new module, res_pjsip_send_to_voicemail allows for REFER requests with
+ particular headers to transfer a PJSIP channel directly to a particular
+ extension that has VoiceMail. This is intended to be used with Digium
+ phones that support this feature.
+
+res_pjsip_outbound_registration
+------------------
+ * A new CLI command has been added: "pjsip show registrations", which lists
+ all configured PJSIP registrations
+
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 12.0.0 to Asterisk 12.1.0 ------------
+------------------------------------------------------------------------------
+
+AMI
+------------------
+ * Added a new module that provides AMI control over MWI within Asterisk,
+ res_mwi_external_ami. Note that this module depends on res_mwi_external;
+ for more information on enabling this module, see res_mwi_external.
+ This module provides the MWIGet/MWIUpdate/MWIDelete actions, as well as
+ the MWIGet/MWIGetComplete events.
+
+ * The DialStatus field in the DialEnd event can now contain additional
+ statuses that convey how the dial operation terminated. This includes
+ ABORT, CONTINUE, and GOTO.
+
+ * AMI will now emit security events. A new class authorization has been
+ added in manager.conf for the security events, 'security'. The new events
+ are:
+ - FailedACL - raised when a request violates an ACL check
+ - InvalidAccountID - raised when a request fails an authentication
+ check due to an invalid account ID
+ - SessionLimit - raised when a request fails due to exceeding the
+ number of allowed concurrent sessions for a service
+ - MemoryLimit - raised when a request fails due to an internal memory
+ allocation failure
+ - LoadAverageLimit - raised when a request fails because a configured
+ load average limit has been reached
+ - RequestNotAllowed - raised when a request is not allowed by
+ the service
+ - AuthMethodNotAllowed - raised when a request used an authentication
+ method not allowed by the service
+ - RequestBadFormat - raised when a request is received with bad formatting
+ - SuccessfulAuth - raised when a request successfully authenticates
+ - UnexpectedAddress - raised when a request has a different source address
+ then what is expected for a session already in progress with a service
+ - ChallengeResponseFailed - raised when a request's attempt to authenticate
+ has been challenged, and the request failed the authentication challenge
+ - InvalidPassword - raised when a request provides an invalid password
+ during an authentication attempt
+ - ChallengeSent - raised when an Asterisk service send an authentication
+ challenge to a request
+ - InvalidTransport - raised when a request attempts to use a transport not
+ allowed by the Asterisk service
+
+ * 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
+
+ARI
+------------------
+ * The Bridge data model now contains the additional fields 'name' and
+ 'creator'. The 'name' field conveys a descriptive name for the bridge;
+ the 'creator' field conveys the name of the entity that created the bridge.
+ This affects all responses to HTTP requests that return a Bridge data model
+ as well as all event derived data models that contain a Bridge data model.
+ The POST /bridges operation may now optionally specify a name to give to
+ the bridge being created.
+
+ * 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. For more
+ information on external MWI control, see res_mwi_external.
+
+ * 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.
+
+HTTP
+------------------
+ * Asterisk's HTTP server now supports chunked Transfer-Encoding. This will be
+ automatically handled by the HTTP server if a request is received with a
+ Transfer-Encoding type of "chunked".
+
+res_pjsip
+------------------
+ * Path support has been added with the 'support_path' option in registration
+ and aor sections.
+
+ * A 'debug' option has been added to the globals section that will allow
+ sip messages to be logged.
+
+ * A 'set_var' option has been added to endpoints that will automatically
+ set the desired variable(s) on a channel created for that endpoint.
+
+ * Several new tables and columns have been added to the realtime schema for
+ the res_pjsip related modules. See the UPGRADE.txt notes for updating
+ the database schema.
+
+res_mwi_external
+------------------
+ * A new module, res_mwi_external, has been added to Asterisk. This module
+ acts as a base framework that other modules can build on top of to allow
+ an external system to control MWI within Asterisk. For implementations
+ that make use of res_mwi_external, see res_mwi_external_ami and
+ res_ari_mailboxes. Note that res_mwi_external conflicts with other modules
+ that may produce MWI themselves, such as app_voicemail. res_mwi_external
+ and other modules that depend on it cannot be built or loaded with
+ app_voicemail present.
+
+res_pjsip
+------------------
+ * DNS functionality will now automatically be enabled if the system configured
+ nameservers can be retrieved. If the system configured nameservers can not be
+ retrieved the functionality will resort to using system resolution. Functionality
+ such as SRV records and failover will not be available if system resolution
+ is in use.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 11 to Asterisk 12 --------------------
+------------------------------------------------------------------------------
+
+Overview
+------------------
+
+Asterisk 12 is a standard release of the Asterisk project. As such, the
+focus of development for this release was on core architectural changes and
+major new features. This includes:
+ * A more flexible bridging core based on the Bridging API
+ * A new internal message bus, Stasis
+ * Major standardization and consistency improvements to AMI
+ * Addition of the Asterisk RESTful Interface (ARI)
+ * A new SIP channel driver, chan_pjsip
+In addition, as the vast majority of bridging in Asterisk was migrated to the
+Bridging API used by ConfBridge, major changes were made to most of the
+interfaces in Asterisk. This includes not only AMI, but also CDRs and CEL.
+
+Specifications have been written for the affected interfaces. These
+specifications are available on the Asterisk wiki:
+ * AMI - https://wiki.asterisk.org/wiki/x/dAFRAQ
+ * CEL - https://wiki.asterisk.org/wiki/x/4ICLAQ
+ * CDRs - https://wiki.asterisk.org/wiki/x/pwpRAQ
+
+It is *highly* recommended that anyone migrating to Asterisk 12 read the
+information regarding its release both in this file and in the accompanying
+UPGRADE.txt file. More detailed information on the major changes can be found
+on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/0YCLAQ.
+
+
+Build System
+------------------
+ * Added build option DISABLE_INLINE. This option can be used to work around a
+ bug in gcc. For more information, see
+ http://gcc.gnu.org/bugzilla/show_bug.cgi?id=47816
+
+ * Removed the CHANNEL_TRACE development mode build option. Certain aspects of
+ the CHANNEL_TRACE build option were incompatible with the new bridging
+ architecture.
+
+ * Asterisk now optionally uses libxslt to improve XML documentation generation
+ and maintainability. If libxslt is not available on the system, some XML
+ documentation will be incomplete.
+
+ * Asterisk now depends on libjansson. If a package of libjansson is not
+ available on your distro, please see http://www.digip.org/jansson/.
+
+ * Asterisk now depends on libuuid and, optionally, uriparser. It is
+ recommended that you install uriparser, even if it is optional.
+
+ * The new SIP stack and channel driver uses a particular version of PJSIP.
+ Please see https://wiki.asterisk.org/wiki/x/J4GLAQ for more information on
+ configuring and installing PJSIP for usage with Asterisk.
+
+ * Optional API was re-implemented to be more portable, and no longer requires
+ weak reference support from the compiler. The build option OPTIONAL_API may
+ be disabled to disable Optional API support.
+
+Applications
+------------------
+
+AgentLogin
+------------------
+ * Along with AgentRequest, this application has been modified to be a
+ replacement for chan_agent. The act of a channel calling the AgentLogin
+ application places the channel into a pool of agents that can be
+ requested by the AgentRequest application. Note that this application, as
+ well as all other agent related functionality, is now provided by the
+ app_agent_pool module. See chan_agent and AgentRequest for more information.
+
+ * This application no longer performs agent authentication. If authentication
+ is desired, the dialplan needs to perform this function using the
+ Authenticate or VMAuthenticate application or through an AGI script before
+ running AgentLogin.
+
+ * If this application is called and the agent is already logged in, the
+ dialplan will continue execution with the AGENT_STATUS channel variable set
+ to ALREADY_LOGGED_IN.
+
+ * The agents.conf schema has changed. Rather than specifying agents on a
+ single line in comma delineated fashion, each agent is defined in a separate
+ context. This allows agents to use the power of context templates in their
+ definition.
+
+ * A number of parameters from agents.conf have been removed. This includes
+ maxloginretries, autologoffunavail, updatecdr, goodbye, group, recordformat,
+ urlprefix, and savecallsin. These options were obsoleted by the move from
+ a channel driver model to the bridging/application model provided by
+ app_agent_pool.
+
+AgentRequest
+------------------
+ * A new application, this will request a logged in agent from the pool and
+ bridge the requested channel with the channel calling this application.
+ Logged in agents are those channels that called the AgentLogin application.
+ If an agent cannot be requested from the pool, the AGENT_STATUS dialplan
+ application will be set with an appropriate error value.
+
+AgentMonitorOutgoing
+------------------
+ * This application has been removed. It was a holdover from when
+ AgentCallbackLogin was removed.
+
+AlarmReceiver
+------------------
+ * Added support for additional Ademco DTMF signalling formats, including
+ Express 4+1, Express 4+2, High Speed and Super Fast.
+
+ * Added channel variable ALARMRECEIVER_CALL_LIMIT. This sets the maximum
+ call time, in milliseconds, to run the application.
+
+ * Added channel variable ALARMRECEIVER_RETRIES_LIMIT. This sets the
+ maximum number of times to retry the call.
+
+ * Added a new configuration option answait. If set, the AlarmReceiver
+ application will wait the number of milliseconds specified by answait
+ after the channel has answered. Valid values range between 500
+ milliseconds and 10000 milliseconds.
+
+ * Added configuration option no_group_meta. If enabled, grouping of metadata
+ information in the AlarmReceiver log file will be skipped.
+
+Answer
+------------------
+ * It is now no longer possible to bypass updating the CDR on the channel
+ when answering. CDRs reflect the state of the channel and will always
+ reflect the time they were Answered.
+
+BridgeWait
+------------------
+ * A new application in Asterisk, this will place the calling channel
+ into a holding bridge, optionally entertaining them with some form of
+ media. Channels participating in a holding bridge do not interact with
+ other channels in the same holding bridge. Optionally, however, a channel
+ may join as an announcer. Any media passed from an announcer channel is
+ played to all channels in the holding bridge. Channels leave a holding
+ bridge either when an optional timer expires, or via the ChannelRedirect
+ application or AMI Redirect action.
+
+ConfBridge
+------------------
+ * All participants in a bridge can now be kicked out of a conference room
+ by specifying the channel parameter as 'all' in the ConfBridge kick CLI
+ command, i.e., 'confbridge kick <conference> all'
+
+ * CLI output for the 'confbridge list' command has been improved. When
+ displaying information about a particular bridge, flags will now be shown
+ for the participating users indicating properties of that user.
+
+ * The ConfbridgeList event now contains the following fields: WaitMarked,
+ EndMarked, and Waiting. This displays additional properties about the
+ user's profile, as well as whether or not the user is waiting for a
+ Marked user to enter the conference.
+
+ * Added a new option for conference recording, record_file_append. If enabled,
+ when the recording is stopped and then re-started, the existing recording
+ will be used and appended to.
+
+ * ConfBridge now has the ability to set the language of announcements to the
+ conference. The language can be set on a bridge profile in confbridge.conf
+ or by the dialplan function CONFBRIDGE(bridge,language)=en.
+
+ControlPlayback
+------------------
+ * The channel variable CPLAYBACKSTATUS may now return the value
+ 'REMOTESTOPPED'. This occurs when playback is stopped by a remote interface,
+ such as AMI. See the AMI action ControlPlayback for more information.
+
+Directory
+------------------
+ * Added the 'a' option, which allows the caller to enter in an additional
+ alias for the user in the directory. This option must be used in conjunction
+ with the 'f', 'l', or 'b' options. Note that the alias for a user can be
+ specified in voicemail.conf.
+
+DumpChan
+------------------
+ * The output of DumpChan no longer includes the DirectBridge or IndirectBridge
+ fields. Instead, if a channel is in a bridge, it includes a BridgeID field
+ containing the unique ID of the bridge that the channel happens to be in.
+
+ForkCDR
+------------------
+ * ForkCDR no longer automatically resets the forked CDR. See the 'r' option
+ for more information.
+
+ * Variables are no longer purged from the original CDR. See the 'v' option for
+ more information.
+
+ * The 'A' option has been removed. The Answer time on a CDR is never updated
+ once set.
+
+ * The 'd' option has been removed. The disposition on a CDR is a function of
+ the state of the channel and cannot be altered.
+
+ * The 'D' option has been removed. Who the Party B is on a CDR is a function
+ of the state of the respective channels involved in the CDR and cannot be
+ altered.
+
+ * The 'r' option has been changed. Previously, ForkCDR always reset the CDR
+ such that the start time and, if applicable, the answer time was updated.
+ Now, by default, ForkCDR simply forks the CDR, maintaining any times. The
+ 'r' option now triggers the Reset, setting the start time (and answer time
+ if applicable) to the current time. Note that the 'a' option still sets
+ the answer time to the current time if the channel was already answered.
+
+ * The 's' option has been removed. A variable can be set on the original CDR
+ if desired using the CDR function, and removed from a forked CDR using the
+ same function.
+
+ * The 'T' option has been removed. The concept of DONT_TOUCH and LOCKED no
+ longer applies in the CDR engine.
+
+ * The 'v' option now prevents the copy of the variables from the original CDR
+ to the forked CDR. Previously the variables were always copied but were
+ removed from the original. This was changed as removing variables from a CDR
+ can have unintended side effects - this option allows the user to prevent
+ propagation of variables from the original to the forked without modifying
+ the original.
+
+MeetMe
+-------------------
+ * Added the 'n' option to MeetMe to prevent application of the DENOISE
+ function to a channel joining a conference. Some channel drivers that vary
+ the number of audio samples in a voice frame will experience significant
+ quality problems if a denoiser is attached to the channel; this option gives
+ them the ability to remove the denoiser without having to unload func_speex.
+
+MixMonitor
+------------------
+ * The 'b' option now includes conferences as well as sounds played to the
+ participants.
+
+ * The AUDIOHOOK_INHERIT function is no longer needed to keep a MixMonitor
+ running during a transfer. If a MixMonitor is started on a channel,
+ the MixMonitor will continue to record the audio passing through the
+ channel even in the presence of transfers.
+
+NoCDR
+------------------
+ * The NoCDR application is deprecated. Please use the CDR_PROP function to
+ disable CDRs.
+
+ * While the NoCDR application will prevent CDRs for a channel from being
+ propagated to registered CDR backends, it will not prevent that data from
+ being collected. Hence, a subsequent call to ResetCDR or the CDR_PROP
+ function that enables CDRs on a channel will restore those records that have
+ not yet been finalized.
+
+ParkAndAnnounce
+-------------------
+ * The app_parkandannounce module has been removed. The application
+ ParkAndAnnounce is now provided by the res_parking module. See the
+ res_parking changes for more information.
+
+Queue
+-------------------
+ * Added queue available hint. The hint can be added to the dialplan using the
+ following syntax: exten,hint,Queue:{queue_name}_avail
+ For example, if the name of the queue is 'markq':
+ exten => 8501,hint,Queue:markq_avail
+ This will report 'InUse' if there are no logged in agents or no free agents.
+ It will report 'Idle' when an agent is free.
+
+ * Queues now support a hint for member paused state. The hint uses the form
+ 'Queue:{queue_name}_pause_{member_name}', where {queue_name} and {member_name}
+ are the name of the queue and the name of the member to subscribe to,
+ respectively. For example: exten => 8501,hint,Queue:sales_pause_mark.
+ Members will show as In Use when paused.
+
+ * The configuration options eventwhencalled and eventmemberstatus have been
+ removed. As a result, the AMI events QueueMemberStatus, AgentCalled,
+ AgentConnect, AgentComplete, AgentDump, and AgentRingNoAnswer will always be
+ sent. The "Variable" fields will also no longer exist on the Agent* events.
+ These events can be filtered out from a connected AMI client using the
+ eventfilter setting in manager.conf.
+
+ * The queue log now differentiates between blind and attended transfers. A
+ blind transfer will result in a BLINDTRANSFER message with the destination
+ context and extension. An attended transfer will result in an
+ ATTENDEDTRANSFER message. This message will indicate the method by which
+ the attended transfer was completed: "BRIDGE" for a bridge merge, "APP"
+ for running an application on a bridge or channel, or "LINK" for linking
+ two bridges together with local channels. The queue log will also now detect
+ externally initiated blind and attended transfers and record the transfer
+ status accordingly.
+
+ * When performing queue pause/unpause on an interface without specifying an
+ individual queue, the PAUSEALL/UNPAUSEALL event will only be logged if at
+ least one member of any queue exists for that interface.
+
+ * Added the 'queue_log_realtime_use_gmt' option to have timestamps in GMT
+ for realtime queue log entries.
+
+ResetCDR
+------------------
+ * The 'e' option has been deprecated. Use the CDR_PROP function to re-enable
+ CDRs when they were previously disabled on a channel.
+
+ * The 'w' and 'a' options have been removed. Dispatching CDRs to registered
+ backends occurs on an as-needed basis in order to preserve linkedid
+ propagation and other needed behavior.
+
+SayAlphaCase
+------------------
+ * A new application, this is similar to SayAlpha except that it supports
+ case sensitive playback of the specified characters. For example,
+ SayAlphaCase(u,aBc) will result in 'a uppercase b c'.
+
+SetAMAFlags
+------------------
+ * This application is deprecated in favor of CHANNEL(amaflags).
+
+SendDTMF
+------------------
+ * The SendDTMF application will now accept 'W' as valid input. This will cause
+ the application to delay one second while streaming DTMF.
+
+Stasis
+------------------
+ * A new application in Asterisk 12, this hands control of the channel calling
+ the application over to an external system. Currently, external systems
+ manipulate channels in Stasis through the Asterisk RESTful Interface (ARI).
+
+UserEvent
+------------------
+ * UserEvent will now handle duplicate keys by overwriting the previous value
+ assigned to the key.
+
+ * In addition to AMI, UserEvent invocations will now be distributed to any
+ interested Stasis applications.
+
+VoiceMail
+------------------
+ * Mailboxes defined by app_voicemail MUST be referenced by the rest of the
+ system as mailbox@context. The rest of the system cannot add @default
+ to mailbox identifiers for app_voicemail that do not specify a context
+ any longer. It is a mailbox identifier format that should only be
+ interpreted by app_voicemail.
+
+ * The voicemail.conf configuration file now has an 'alias' configuration
+ parameter for use with the Directory application. The voicemail realtime
+ database table schema has also been updated with an 'alias' column.
+
+
+Codecs
+------------------
+ * Pass through support has been added for both VP8 and Opus.
+
+ * Added format attribute negotiation for the Opus codec. Format attribute
+ negotiation is provided by the res_format_attr_opus module.
+
+
+Core
+------------------
+ * Masquerades as an operation inside Asterisk have been effectively hidden
+ by the migration to the Bridging API. As such, many 'quirks' of Asterisk
+ no longer occur. This includes renaming of channels, "<ZOMBIE>" channels,
+ dropping of frame/audio hooks, and other internal implementation details
+ that users had to deal with. This fundamental change has large implications
+ throughout the changes documented for this version. For more information
+ about the new core architecture of Asterisk, please see the Asterisk wiki.
+
+ * Multiple parties in a bridge may now be transferred. If a participant in a
+ multi-party bridge initiates a blind transfer, a Local channel will be used
+ to execute the dialplan location that the transferer sent the parties to. If
+ a participant in a multi-party bridge initiates an attended transfer,
+ several options are possible. If the attended transfer results in a transfer
+ to an application, a Local channel is used. If the attended transfer results
+ in a transfer to another channel, the resulting channels will be merged into
+ a single bridge.
+
+ * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is no longer channel
+ driver specific. If the channel variable is set on the transferrer channel,
+ the sound will be played to the target of an attended transfer.
+
+ * The channel variable BRIDGEPEER becomes a comma separated list of peers in
+ a multi-party bridge. The BRIDGEPEER value can have a maximum of 10 peers
+ listed. Any more peers in the bridge will not be included in the list.
+ BRIDGEPEER is not valid in holding bridges like parking since those channels
+ do not talk to each other even though they are in a bridge.
+
+ * The channel variable BRIDGEPVTCALLID is only valid for two party bridges
+ and will contain a value if the BRIDGEPEER's channel driver supports it.
+
+ * A channel variable ATTENDEDTRANSFER is now set which indicates which channel
+ was responsible for an attended transfer in a similar fashion to
+ BLINDTRANSFER.
+
+ * Modules using the Configuration Framework or Sorcery must have XML
+ configuration documentation. This configuration documentation is included
+ with the rest of Asterisk's XML documentation, and is accessible via CLI
+ commands. See the CLI changes for more information.
+
+AMI (Asterisk Manager Interface)
+------------------
+ * Major changes were made to both the syntax as well as the semantics of the
+ AMI protocol. In particular, AMI events have been substantially improved
+ in this version of Asterisk. For more information, please see the AMI
+ specification at https://wiki.asterisk.org/wiki/x/dAFRAQ
+
+ * AMI events that reference a particular channel or bridge will now always
+ contain a standard set of fields. When multiple channels or bridges are
+ referenced in an event, fields for at least some subset of the channels
+ and bridges in the event will be prefixed with a descriptive name to avoid
+ name collisions. See the AMI event documentation on the Asterisk wiki for
+ more information.
+
+ * The CLI command 'manager show commands' no longer truncates command names
+ longer than 15 characters and no longer shows authorization requirement
+ for commands. 'manager show command' now displays the privileges needed
+ for using a given manager command instead.
+
+ * The SIPshowpeer action will now include a 'SubscribeContext' field for a
+ peer in its response if the peer has a subscribe context set.
+
+ * The SIPqualifypeer action now acknowledges the request once it has
+ established that the request is against a known peer. It also issues a new
+ event, 'SIPQualifyPeerDone', once the qualify action has been completed.
+
+ * The PlayDTMF action now supports an optional 'Duration' parameter. This
+ specifies the duration of the digit to be played, in milliseconds.
+
+ * Added VoicemailRefresh action to allow an external entity to trigger mailbox
+ updates when changes occur instead of requiring the use of pollmailboxes.
+
+ * Added a new action 'ControlPlayback'. The ControlPlayback action allows an
+ AMI client to manipulate audio currently being played back on a channel. The
+ supported operations depend on the application being used to send audio to
+ the channel. When the audio playback was initiated using the ControlPlayback
+ application or CONTROL STREAM FILE AGI command, the audio can be paused,
+ stopped, restarted, reversed, or skipped forward. When initiated by other
+ mechanisms (such as the Playback application), the audio can be stopped,
+ reversed, or skipped forward.
+
+ * Channel related events now contain a snapshot of channel state, adding new
+ fields to many of these events.
+
+ * The AMI event 'Newexten' field 'Extension' is deprecated, and may be removed
+ in a future release. Please use the common 'Exten' field instead.
+
+ * The AMI event 'UserEvent' from app_userevent now contains the channel state
+ fields. The channel state fields will come before the body fields.
+
+ * The AMI events 'ParkedCall', 'ParkedCallTimeOut', 'ParkedCallGiveUp', and
+ 'UnParkedCall' have changed significantly in the new res_parking module.
+
+ The 'Channel' and 'From' headers are gone. For the channel that was parked
+ or is coming out of parking, a 'Parkee' channel snapshot is issued and it
+ has a number of fields associated with it. The old 'Channel' header relayed
+ the same data as the new 'ParkeeChannel' header.
+
+ The 'From' field was ambiguous and changed meaning depending on the event.
+ for most of these, it was the name of the channel that parked the call
+ (the 'Parker'). There is no longer a header that provides this channel name,
+ however the 'ParkerDialString' will contain a dialstring to redial the
+ device that parked the call.
+
+ On UnParkedCall events, the 'From' header would instead represent the
+ channel responsible for retrieving the parkee. It receives a channel
+ snapshot labeled 'Retriever'. The 'from' field is is replaced with
+ 'RetrieverChannel'.
+
+ Lastly, the 'Exten' field has been replaced with 'ParkingSpace'.
+
+ * The AMI event 'Parkinglot' (response to 'Parkinglots' command) in a similar
+ fashion has changed the field names 'StartExten' and 'StopExten' to
+ 'StartSpace' and 'StopSpace' respectively.
+
+ * The deprecated use of | (pipe) as a separator in the channelvars setting in
+ manager.conf has been removed.
+
+ * Channel Variables conveyed with a channel no longer contain the name of the
+ channel as part of the key field, i.e., ChanVariable(SIP/foo): bar=baz is now
+ ChanVariable: bar=baz. When multiple channels are present in a single AMI
+ event, the various ChanVariable fields will contain a suffix that specifies
+ which channel they correspond to.
+
+ * The NewPeerAccount AMI event is no longer raised. The NewAccountCode AMI
+ event always conveys the AMI event for a particular channel.
+
+ * All 'Reload' events have been consolidated into a single event type. This
+ event will always contain a Module field specifying the name of the module
+ and a Status field denoting the result of the reload. All modules now issue
+ this event when being reloaded.
+
+ * The 'ModuleLoadReport' event has been removed. Most AMI connections would
+ fail to receive this event due to being connected after modules have loaded.
+ AMI connections that want to know when Asterisk is ready should listen for
+ the 'FullyBooted' event.
+
+ * app_fax now sends the same send fax/receive fax events as res_fax. The
+ 'FaxSent' event is now the 'SendFAX' event, and the 'FaxReceived' event is
+ now the 'ReceiveFAX' event.
+
+ * The 'MusicOnHold' event is now two events: 'MusicOnHoldStart' and
+ 'MusicOnHoldStop'. The sub type field has been removed.
+
+ * The 'JabberEvent' event has been removed. It is not AMI's purpose to be a
+ carrier for another protocol.
+
+ * The Bridge Manager action's 'Playtone' header now accepts more fine-grained
+ options. 'Channel1' and 'Channel2' may be specified in order to play a tone
+ to the specific channel. 'Both' may be specified to play a tone to both
+ channels. The old 'yes' option is still accepted as a way of playing the
+ tone to Channel2 only.
+
+ * The AMI 'Status' response event to the AMI Status action replaces the
+ 'BridgedChannel' and 'BridgedUniqueid' headers with the 'BridgeID' header to
+ indicate what bridge the channel is currently in.
+
+ * The AMI 'Hold' event has been moved out of individual channel drivers, into
+ core, and is now two events: 'Hold' and 'Unhold'. The status field has been
+ removed.
+
+ * The AMI events in app_queue have been made more consistent with each other.
+ Events that reference channels (QueueCaller* and Agent*) will show
+ information about each channel. The (infamous) 'Join' and 'Leave' AMI
+ events have been changed to 'QueueCallerJoin' and 'QueueCallerLeave'.
+
+ * The 'MCID' AMI event now publishes a channel snapshot when available and
+ its non-channel-snapshot parameters now use either the "MCallerID" or
+ 'MConnectedID' prefixes with Subaddr*, Name*, and Num* suffixes instead
+ of 'CallerID' and 'ConnectedID' to avoid confusion with similarly named
+ parameters in the channel snapshot.
+
+ * The AMI events 'Agentlogin' and 'Agentlogoff' have been renamed
+ 'AgentLogin' and 'AgentLogoff' respectively.
+
+ * The 'Channel' key used in the 'AlarmClear', 'Alarm', and 'DNDState' has been
+ renamed "DAHDIChannel" since it does not convey an Asterisk channel name.
+
+ * 'ChannelUpdate' events have been removed.
+
+ * All AMI events now contain a 'SystemName' field, if available.
+
+ * Local channel optimization is now conveyed in two events:
+ 'LocalOptimizationBegin' and 'LocalOptimizationEnd'. The Begin event is sent
+ when the Local channel driver begins attempting to optimize itself out of
+ the media path; the End event is sent after the channel halves have
+ successfully optimized themselves out of the media path.
+
+ * Local channel information in events is now prefixed with 'LocalOne' and
+ 'LocalTwo'. This replaces the suffix of '1' and '2' for the two halves of
+ the Local channel. This affects the 'LocalBridge', 'LocalOptimizationBegin',
+ and 'LocalOptimizationEnd' events.
+
+ * The option 'allowmultiplelogin' can now be set or overriden in a particular
+ account. When set in the general context, it will act as the default
+ setting for defined accounts.
+
+ * The 'BridgeAction' event was removed. It technically added no value, as the
+ Bridge Action already receives confirmation of the bridge through a
+ successful completion Event.
+
+ * The 'BridgeExec' events were removed. These events duplicated the events that
+ occur in the Bridging API, and are conveyed now through BridgeCreate,
+ BridgeEnter, and BridgeLeave events.
+
+ * The 'RTCPSent'/'RTCPReceived' events have been significantly modified from
+ previous versions. They now report all SR/RR packets sent/received, and
+ have been restructured to better reflect the data sent in a SR/RR. In
+ particular, the event structure now supports multiple report blocks.
+
+ * Added 'BlindTransfer' and 'AttendedTransfer' events. These events are
+ raised when a blind transfer/attended transfer completes successfully.
+ They contain information about the transfer that just completed, including
+ the location of the transfered channel.
+
+ * Added a 'security' class to AMI which outputs the required fields for
+ security messages similar to the log messages from res_security_log
+
+ * The AMI event 'ExtensionStatus' now contains a 'StatusText' field
+ that describes the status value in a human readable string.
+
+CDR (Call Detail Records)
+------------------
+ * Significant changes have been made to the behavior of CDRs. The CDR engine
+ was effectively rewritten and built on the Stasis message bus. For a full
+ definition of CDR behavior in Asterisk 12, please read the specification
+ on the Asterisk wiki (wiki.asterisk.org).
+
+ * CDRs will now be created between all participants in a bridge. For each
+ pair of channels in a bridge, a CDR is created to represent the path of
+ communication between those two endpoints. This lets an end user choose who
+ to bill for what during bridge operations with multiple parties.
+
+ * The duration, billsec, start, answer, and end times now reflect the times
+ associated with the current CDR for the channel, as opposed to a cumulative
+ measurement of all CDRs for that channel.
+
+ * When a CDR is dispatched, user defined CDR variables from both parties are
+ included in the resulting CDR. If both parties have the same variable, only
+ the Party A value is provided.
+
+ * Added a new option to cdr.conf, 'debug'. When enabled, significantly more
+ information regarding the CDR engine is logged as verbose messages. This
+ option should only be used if the behavior of the CDR engine needs to be
+ debugged.
+
+ * Added CLI command 'cdr set debug {on|off}'. This toggles the 'debug' setting
+ normally configured in cdr.conf.
+
+ * Added CLI command 'cdr show active {channel}'. When {channel} is not
+ specified, this command provides a summary of the channels with CDR
+ information and their statistics. When {channel} is specified, it shows
+ detailed information about all records associated with {channel}.
+
+CEL (Channel Event Logging)
+------------------
+ * CEL has undergone significant rework in Asterisk 12, and is now built on the
+ Stasis message bus. Please see the specification for CEL on the Asterisk
+ wiki at https://wiki.asterisk.org/wiki/x/4ICLAQ for more detailed
+ information.
+
+ * The 'extra' field of all CEL events that use it now consists of a JSON blob
+ with key/value pairs which are defined in the Asterisk 12 CEL documentation.
+
+ * BLINDTRANSFER events now report the transferee bridge unique
+ identifier, extension, and context in a JSON blob as the extra string
+ instead of the transferee channel name as the peer.
+
+ * ATTENDEDTRANSFER events now report the peer as NULL and additional
+ information in the 'extra' string as a JSON blob. For transfers that occur
+ between two bridged channels, the 'extra' JSON blob contains the primary
+ bridge unique identifier, the secondary channel name, and the secondary
+ bridge unique identifier. For transfers that occur between a bridged channel
+ and a channel running an app, the 'extra' JSON blob contains the primary
+ bridge unique identifier, the secondary channel name, and the app name.
+
+ * LOCAL_OPTIMIZE events have been added to convey local channel
+ optimizations with the record occurring for the semi-one channel and
+ the semi-two channel name in the peer field.
+
+ * BRIDGE_START, BRIDGE_END, BRIDGE_UPDATE, 3WAY_START, 3WAY_END, CONF_ENTER,
+ CONF_EXIT, CONF_START, and CONF_END events have all been removed. These
+ events have been replaced by BRIDGE_ENTER/BRIDGE_EXIT. The BRIDGE_ENTER
+ and BRIDGE_EXIT events are raised when a channel enters/exits any bridge,
+ regardless of whether or not that bridge happens to contain multiple
+ parties.
+
+CLI
+-------------------
+ * When compiled with '--enable-dev-mode', the astobj2 library will now add
+ several CLI commands that allow for inspection of ao2 containers that
+ register themselves with astobj2. The CLI commands are 'astobj2 container
+ dump', 'astobj2 container stats', and 'astobj2 container check'.
+
+ * Added specific CLI commands for bridge inspection. This includes 'bridge
+ show all', which lists all bridges in the system, and 'bridge show {id}',
+ which provides specific information about a bridge.
+
+ * Added CLI command 'bridge destroy'. This will destroy the specified bridge,
+ ejecting the channels currently in the bridge. If the channels cannot
+ continue in the dialplan or application that put them in the bridge, they
+ will be hung up.
+
+ * Added command 'bridge kick'. This will eject a single channel from a bridge.
+
+ * Added commands to inspect and manipulate the registered bridge technologies.
+ This include 'bridge technology show', which lists the registered bridge
+ technologies, as well as 'bridge technology {suspend|unsuspend} {tech}',
+ which controls whether or not a registered bridge technology can be used
+ during smart bridge operations. If a technology is suspended, it will not
+ be used when a bridge technology is picked for channels; when unsuspended,
+ it can be used again.
+
+ * The command 'config show help {module} {type} {option}' will show
+ configuration documentation for modules with XML configuration
+ documentation. When {module}, {type}, and {option} are omitted, a listing
+ of all modules with registered documentation is displayed. When {module}
+ is specified, a listing of all configuration types for that module is
+ displayed, along with their synopsis. When {module} and {type} are
+ specified, a listing of all configuration options for that type are
+ displayed along with their synopsis. When {module}, {type}, and {option}
+ are specified, detailed information for that configuration option is
+ displayed.
+
+ * Added 'core show sounds' and 'core show sound' CLI commands. These display
+ a listing of all installed media sounds available on the system and
+ detailed information about a sound, respectively.
+
+ * 'xmldoc dump' has been added. This CLI command will dump the XML
+ documentation DOM as a string to the specified file. The Asterisk core
+ will populate certain XML elements pulled from the source files with
+ additional run-time information; this command lets a user produce the
+ XML documentation with all information.
+
+Features
+-------------------
+ * Parking has been pulled from core and placed into a separate module called
+ res_parking. See Parking changes below for more details. Configuration for
+ parking should now be performed in res_parking.conf. Configuration for
+ parking in features.conf is now unsupported.
+
+ * Core attended transfers now have several new options. While performing an
+ attended transfer, the transferer now has the following options:
+ - *1 - cancel the attended transfer (configurable via atxferabort)
+ - *2 - complete the attended transfer, dropping out of the call
+ (configurable via atxfercomplete)
+ - *3 - complete the attended transfer, but stay in the call. This will turn
+ the call into a multi-party bridge (configurable via atxferthreeway)
+ - *4 - swap to the other party. Once an attended transfer has begun, this
+ options may be used multiple times (configurable via atxferswap)
+
+ * For DTMF blind and attended transfers, the channel variable TRANSFER_CONTEXT
+ must be on the channel initiating the transfer to have any effect.
+
+ * The BRIDGE_FEATURES channel variable would previously only set features for
+ the calling party and would set this feature regardless of whether the
+ feature was in caps or in lowercase. Use of a caps feature for a letter
+ will now apply the feature to the calling party while use of a lowercase
+ letter will apply that feature to the called party.
+
+ * Add support for automixmon to the BRIDGE_FEATURES channel variable.
+
+ * The channel variable DYNAMIC_PEERNAME is redundant with BRIDGEPEER and is
+ removed. The more useful DYNAMIC_WHO_ACTIVATED gives the channel name that
+ activated the dynamic feature.
+
+ * The channel variables DYNAMIC_FEATURENAME and DYNAMIC_WHO_ACTIVATED are set
+ only on the channel executing the dynamic feature. Executing a dynamic
+ feature on the bridge peer in a multi-party bridge will execute it on all
+ peers of the activating channel.
+
+ * You can now have the settings for a channel updated using the FEATURE()
+ and FEATUREMAP() functions inherited to child channels by setting
+ FEATURE(inherit)=yes.
+
+ * automixmon now supports additional channel variables from automon including:
+ TOUCH_MIXMONITOR_PREFIX, TOUCH_MIXMONITOR_MESSAGE_START,
+ and TOUCH_MIXMONITOR_MESSAGE_STOP
+
+ * A new general features.conf option 'recordingfailsound' has been added which
+ allowssetting a failure sound for a user tries to invoke a recording feature
+ such as automon or automixmon and it fails.
+
+ * It is no longer necessary (or possible) to define the ATXFER_NULL_TECH in
+ features.c for atxferdropcall=no to work properly. This option now just
+ works.
+
+Logging
+-------------------
+ * Added log rotation strategy 'none'. If set, no log rotation strategy will
+ be used. Given that this can cause the Asterisk log files to grow quickly,
+ this option should only be used if an external mechanism for log management
+ is preferred.
+
+Realtime
+------------------
+ * Dynamic realtime tables for SIP Users can now include a 'path' field. This
+ will store the path information for that peer when it registers. Realtime
+ tables can also use the 'supportpath' field to enable Path header support.
+
+ * LDAP realtime configurations for SIP Users now have the AstAccountPathSupport
+ objectIdentifier. This maps to the supportpath option in sip.conf.
+
+Sorcery
+------------------
+ * Sorcery is a new data abstraction and object persistence API in Asterisk. It
+ provides modules a useful abstraction on top of the many storage mechanisms
+ in Asterisk, including the Asterisk Database, static configuration files,
+ static Realtime, and dynamic Realtime. It also provides a caching service.
+ Users can configure a hierarchy of data storage layers for specific modules
+ in sorcery.conf.
+
+ * All future modules which utilize Sorcery for object persistence must have a
+ column named "id" within their schema when using the Sorcery realtime module.
+ This column must be able to contain a string of up to 128 characters in length.
+
+Security Events Framework
+------------------
+ * Security Event timestamps now use ISO 8601 formatted date/time instead of
+ the "seconds-microseconds" format that it was using previously.
+
+Stasis Message Bus
+------------------
+ * The Stasis message bus is a publish/subscribe message bus internal to
+ Asterisk. Many services in Asterisk are built on the Stasis message bus,
+ including AMI, ARI, CDRs, and CEL. Parameters controlling the operation of
+ Stasis can be configured in stasis.conf. Note that these parameters operate
+ at a very low level in Asterisk, and generally will not require changes.
+
+Channel Drivers
+------------------
+ * When a channel driver is configured to enable jiterbuffers, they are now
+ applied unconditionally when a channel joins a bridge. If a jitterbuffer
+ is already set for that channel when it enters, such as by the JITTERBUFFER
+ function, then the existing jitterbuffer will be used and the one set by
+ the channel driver will not be applied.
+
+chan_agent
+------------------
+ * chan_agent has been removed and replaced with AgentLogin and AgentRequest
+ dialplan applications provided by the app_agent_pool module. Agents are
+ connected with callers using the new AgentRequest dialplan application.
+ The Agents:<agent-id> device state is available to monitor the status of an
+ agent. See agents.conf.sample for valid configuration options.
+
+ * The updatecdr option has been removed. Altering the names of channels on a
+ CDR is not supported - the name of the channel is the name of the channel,
+ and pretending otherwise helps no one. The AGENTUPDATECDR channel variable
+ has also been removed, for the same reason.
+
+ * The endcall and enddtmf configuration options are removed. Use the
+ dialplan function CHANNEL(dtmf_features) to set DTMF features on the agent
+ channel before calling AgentLogin.
+
+chan_bridge
+------------------
+ * chan_bridge has been removed. Its functionality has been incorporated
+ directly into the ConfBridge application itself.
+
+chan_dahdi
+------------------
+ * Added the CLI command 'pri destroy span'. This will destroy the D-channel
+ of the specified span and its B-channels. Note that this command should
+ only be used if you understand the risks it entails.
+
+ * The CLI command 'dahdi destroy channel' is now 'dahdi destroy channels'.
+ A range of channels can be specified to be destroyed. Note that this command
+ should only be used if you understand the risks it entails.
+
+ * Added the CLI command 'dahdi create channels'. A range of channels can be
+ specified to be created, or the keyword 'new' can be used to add channels
+ not yet created.
+
+ * The script specified by the chan_dahdi.conf mwimonitornotify option now gets
+ the exact configured mailbox name. For app_voicemail mailboxes this is
+ mailbox@context.
+
+ * Added mwi_vm_boxes that also must be configured for ISDN MWI to be enabled.
+
+chan_iax2
+------------------
+ * IPv6 support has been added. We are now able to bind to and
+ communicate using IPv6 addresses.
+
+chan_local
+------------------
+ * The /b option has been removed.
+
+ * chan_local moved into the system core and is no longer a loadable module.
+
+chan_mobile
+------------------
+ * Added general support for busy detection.
+
+ * Added ECAM command support for Sony Ericsson phones.
+
+chan_pjsip
+------------------
+ * A new SIP channel driver for Asterisk, chan_pjsip is built on the PJSIP
+ SIP stack. A collection of resource modules provides the bulk of the SIP
+ functionality. For more information on the new SIP channel driver, see
+ https://wiki.asterisk.org/wiki/x/JYGLAQ
+
+chan_sip
+------------------
+ * Added support for RFC 3327 "Path" headers. This can be enabled in sip.conf
+ using the 'supportpath' setting, either on a global basis or on a peer basis.
+ This setting enables Asterisk to route outgoing out-of-dialog requests via a
+ set of proxies by using a pre-loaded route-set defined by the Path headers in
+ the REGISTER request. See Realtime updates for more configuration information.
+
+ * The SIP_CODEC family of variables may now specify more than one codec. Each
+ codec must be separated by a comma. The first codec specified is the
+ preferred codec for the offer. This allows a dialplan writer to specify both
+ audio and video codecs, e.g., Set(SIP_CODEC=ulaw,h264)
+
+ * The 'callevents' parameter has been removed. Hold AMI events are now raised
+ in the core, and can be filtered out using the 'eventfilter' parameter
+ in manager.conf.
+
+ * Added 'ignore_requested_pref'. When enabled, this will use the preferred
+ codecs configured for a peer instead of the requested codec.
+
+ * The option "register_retry_403" has been added to chan_sip to work around
+ servers that are known to erroneously send 403 in response to valid
+ REGISTER requests and allows Asterisk to continue attepmting to connect.
+
+chan_skinny
+------------------
+ * Added the 'immeddialkey' parameter. If set, when the user presses the
+ configured key the already entered number will be immediately dialed. This
+ is useful when the dialplan allows for variable length pattern matching.
+ Valid options are '*' and '#'.
+
+ * Added the 'callfwdtimeout' parameter. This configures the amount of time (in
+ milliseconds) before a call forward is considered to not be answered.
+
+ * The 'serviceurl' parameter allows Service URLs to be attached to line
+ buttons.
+
+
+Functions
+------------------
+
+AGENT
+------------------
+ * The password option has been disabled, as the AgentLogin application no
+ longer provides authentication.
+
+AUDIOHOOK_INHERIT
+------------------
+ * Due to changes in the Asterisk core, this function is no longer needed to
+ preserve a MixMonitor on a channel during transfer operations and dialplan
+ execution. It is effectively obsolete.
+
+CDR (function)
+------------------
+ * The 'amaflags' and 'accountcode' attributes for the CDR function are
+ deprecated. Use the CHANNEL function instead to access these attributes.
+
+ * The 'l' option has been removed. When reading a CDR attribute, the most
+ recent record is always used. When writing a CDR attribute, all non-finalized
+ CDRs are updated.
+
+ * The 'r' option has been removed, for the same reason as the 'l' option.
+
+ * The 's' option has been removed, as LOCKED semantics no longer exist in the
+ CDR engine.
+
+CDR_PROP
+------------------
+ * A new function CDR_PROP has been added. This function lets you set properties
+ on a channel's active CDRs. This function is write-only. Properties accept
+ boolean values to set/clear them on the channel's CDRs. Valid properties
+ include:
+ - 'party_a' - make this channel the preferred Party A in any CDR between two
+ channels. If two channels have this property set, the creation time of the
+ channel is used to determine who is Party A. Note that dialed channels are
+ never Party A in a CDR.
+ - 'disable' - disable CDRs on this channel. This is analogous to the NoCDR
+ application when set to True, and analogous to the 'e' option in ResetCDR
+ when set to False.
+
+CHANNEL
+------------------
+ * Added the argument 'dtmf_features'. This sets the DTMF features that will be
+ enabled on a channel when it enters a bridge. Allowed values are 'T', 'K',
+ 'H', 'W', and 'X', and are analogous to the parameters passed to the Dial
+ application.
+
+ * Added the argument 'after_bridge_goto'. This can be set to a parseable Goto
+ string, i.e., [[context],extension],priority. If set on a channel, if a
+ channel leaves a bridge but is not hung up it will resume dialplan execution
+ at that location.
+
+JITTERBUFFER
+------------------
+ * JITTERBUFFER now accepts an argument of 'disabled' which can be used
+ to remove jitterbuffers previously set on a channel with JITTERBUFFER.
+ The value of this setting is ignored when disabled is used for the argument.
+
+PJSIP_DIAL_CONTACTS
+------------------
+ * A new function provided by chan_pjsip, this function can be used in
+ conjunction with the Dial application to construct a dial string that will
+ dial all contacts on an Address of Record associated with a chan_pjsip
+ endpoint.
+
+PJSIP_MEDIA_OFFER
+------------------
+ * Provided by chan_pjsip, this function sets the codecs to be offered on the
+ outbound channel prior to dialing.
+
+REDIRECTING
+------------------
+ * Redirecting reasons can now be set to arbitrary strings. This means
+ that the REDIRECTING dialplan function can be used to set the redirecting
+ reason to any string. It also allows for custom strings to be read as the
+ redirecting reason from SIP Diversion headers.
+
+SPEECH_ENGINE
+------------------
+ * The SPEECH_ENGINE function now supports read operations. When read from, it
+ will return the current value of the requested attribute.
+
+VMCOUNT:
+------------------
+ * Mailboxes defined by app_voicemail MUST be referenced by the rest of the
+ system as mailbox@context. The rest of the system cannot add @default
+ to mailbox identifiers for app_voicemail that do not specify a context
+ any longer. It is a mailbox identifier format that should only be
+ interpreted by app_voicemail.
+
+
+Resources
+------------------
+
+res_agi (Asterisk Gateway Interface)
+------------------
+ * The manager event AGIExec has been split into AGIExecStart and AGIExecEnd.
+
+ * The manager event AsyncAGI has been split into AsyncAGIStart, AsyncAGIExec,
+ and AsyncAGIEnd.
+
+ * The CONTROL STREAM FILE command now accepts an offsetms parameter. This
+ will start the playback of the audio at the position specified. It will
+ also return the final position of the file in 'endpos'.
+
+ * The CONTROL STREAM FILE command will now populate the CPLAYBACKSTATUS
+ channel variable if the user stopped the file playback or if a remote
+ entity stopped the playback. If neither stopped the playback, it will
+ indicate the overall success/failure of the playback. If stopped early,
+ the final offset of the file will be set in the CPLAYBACKOFFSET channel
+ variable.
+
+ * The SAY ALPHA command now accepts an additional parameter to control
+ whether it specifies the case of uppercase, lowercase, or all letters to
+ provide functionality similar to SayAlphaCase.
+
+res_ari (Asterisk RESTful Interface) (and others)
+------------------
+ * The Asterisk RESTful Interface (ARI) provides a mechanism to expose and
+ control telephony primitives in Asterisk by remote client. This includes
+ channels, bridges, endpoints, media, and other fundamental concepts. Users
+ of ARI can develop their own communications applications, controlling
+ multiple channels using an HTTP RESTful interface and receiving JSON events
+ about the objects via a WebSocket connection. ARI can be configured in
+ Asterisk via ari.conf. For more information on ARI, see
+ https://wiki.asterisk.org/wiki/x/0YCLAQ
+
+res_parking
+-------------------
+ * Parking has been extracted from the Asterisk core as a loadable module,
+ res_parking. Configuration for parking is now provided by res_parking.conf.
+ Configuration through features.conf is no longer supported.
+
+ * res_parking uses the configuration framework. If an invalid configuration is
+ supplied, res_parking will fail to load or fail to reload. Previously,
+ invalid configurations would generally be accepted, with certain errors
+ resulting in individually disabled parking lots.
+
+ * Parked calls are now placed in bridges. While this is largely an
+ architectural change, it does have implications on how channels in a parking
+ lot are viewed. For example, commands that display channels in bridges will
+ now also display the channels in a parking lot.
+
+ * The order of arguments for the new parking applications have been modified.
+ Timeout and return context/exten/priority are now implemented as options,
+ while the name of the parking lot is now the first parameter. See the
+ application documentation for Park, ParkedCall, and ParkAndAnnounce for more
+ in-depth information as well as syntax.
+
+ * Extensions are by default no longer automatically created in the dialplan to
+ park calls or pickup parked calls. Generation of dialplan extensions can be
+ enabled using the 'parkext' configuration option.
+
+ * ADSI functionality for parking is no longer supported. The 'adsipark'
+ configuration option has been removed as a result.
+
+ * The PARKINGSLOT channel variable has been deprecated in favor of
+ PARKING_SPACE to match the naming scheme of the new system.
+
+ * PARKING_SPACE and PARKEDLOT channel variables will now be set for a parked
+ channel even when the configuration option 'comebactoorigin' is enabled.
+
+ * A new CLI command 'parking show' has been added. This allows a user to
+ inspect the parking lots that are currently in use.
+ 'parking show <parkinglot>' will also show the parked calls in a specific
+ parking lot.
+
+ * The CLI command 'parkedcalls' is now deprecated in favor of
+ 'parking show <parkinglot>'.
+
+ * The AMI command 'ParkedCalls' will now accept a 'ParkingLot' argument which
+ can be used to get a list of parked calls for a specific parking lot.
+
+ * The AMI command 'Park' field 'Channel2' has been deprecated and replaced
+ with 'TimeoutChannel'. If both 'Channel2' and 'TimeoutChannel' are
+ specified, 'TimeoutChannel' will be used. The field 'TimeoutChannel' is no
+ longer a required argument.
+
+ * The ParkAndAnnounce application is now provided through res_parking instead
+ of through the separate app_parkandannounce module.
+
+ * ParkAndAnnounce will no longer go to the next position in dialplan on timeout
+ by default. Instead, it will follow the timeout rules of the parking lot. The
+ old behavior can be reproduced by using the 'c' option.
+
+ * Dynamic parking lots will now fail to be created under the following
+ conditions:
+ - if the parking lot specified by PARKINGDYNAMIC does not exist
+ - if they require exclusive park and parkedcall extensions which overlap
+ with existing parking lots.
+
+ * Dynamic parking lots will be cleared on reload for dynamic parking lots that
+ currently contain no calls. Dynamic parking lots containing parked calls
+ will persist through the reloads without alteration.
+
+ * If 'parkext_exclusive' is set for a parking lot and that extension is
+ already in use when that parking lot tries to register it, this is now
+ considered a parking system configuration error. Configurations which do
+ this will be rejected.
+
+ * Added channel variable PARKER_FLAT. This contains the name of the extension
+ that would be used if 'comebacktoorigin' is enabled. This can be useful when
+ comebacktoorigin is disabled, but the dialplan or an external control
+ mechanism wants to use the extension in the park-dial context that was
+ generated to re-dial the parker on timeout.
+
+res_pjsip (and many others)
+------------------
+ * A large number of resource modules make up the SIP stack based on pjsip.
+ The chan_pjsip channel driver users these resource modules to provide
+ various SIP functionality in Asterisk. The majority of configuration for
+ these modules is performed in pjsip.conf. Other modules may use their
+ own configuration files.
+
+ * Added 'set_var' option for an endpoint. For each variable specified that
+ variable gets set upon creation of a channel involving the endpoint.
+
+res_rtp_asterisk
+------------------
+ * ICE/STUN/TURN support in res_rtp_asterisk has been made optional. To enable
+ them, an Asterisk-specific version of PJSIP needs to be installed.
+ Tarballs are available from https://github.com/asterisk/pjproject/tags/.
+
+res_statsd/res_chan_stats
+------------------
+ * A new resource module, res_statsd, has been added, which acts as a statsd
+ client. This module allows Asterisk to publish statistics to a statsd
+ server. In conjunction with res_chan_stats, it will publish statistics about
+ channels to the statsd server. It can be configured via res_statsd.conf.
+
+res_xmpp
+------------------
+ * Device state for XMPP buddies is now available using the following format:
+ XMPP/<client name>/<buddy address>
+ If any resource is available the device state is considered to be not in use.
+ If no resources exist or all are unavailable the device state is considered
+ to be unavailable.
+
+
+Scripts
+------------------
+
+Realtime/Database Scripts
+------------------
+ * Asterisk previously included example db schemas in the contrib/realtime/
+ directory of the source tree. This has been replaced by a set of database
+ migrations using the Alembic framework. This allows you to use alembic to
+ initialize the database for you. It will also serve as a database migration
+ tool when upgrading Asterisk in the future.
+
+ See contrib/ast-db-manage/README.md for more details.
+
+sip_to_res_pjsip.py
+-------------------
+ * A new script has been added in the contrib/scripts/sip_to_res_pjsip folder.
+ This python script will convert an existing sip.conf file to a
+ pjsip.conf file, for use with the chan_pjsip channel driver. This script
+ is meant to be an aid in converting an existing chan_sip configuration to
+ a chan_pjsip configuration, but it is expected that configuration beyond
+ what the script provides will be needed.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 10 to Asterisk 11 --------------------
+------------------------------------------------------------------------------
+
+Build System
+-------------------
+ * The Asterisk build system will now build and install a shared library
+ (libasteriskssl.so) used to wrap various initialization and shutdown functions
+ from the libssl and libcrypto libraries provided by OpenSSL. This is done so
+ that Asterisk can ensure that these functions do *not* get called by any
+ modules that are loaded into Asterisk, since they should only be called once
+ in any single process. If desired, this feature can be disabled by supplying
+ the "--disable-asteriskssl" option to the configure script.
+
+ * A new make target, 'full', has been added to the Makefile. This performs
+ the same compilation actions as make all, but will also scan the entirety of
+ each source file for documentation. This option is needed to generate AMI
+ event documentation. Note that your system must have Python in order for
+ this make target to succeed.
+
+ * The optimization portion of the build system has been reworked to avoid
+ broken builds on certain architectures. All architecture-specific
+ optimization has been removed in favor of using -march=native to allow gcc
+ to detect the environment in which it is running when possible. This can
+ be toggled as BUILD_NATIVE under "Compiler Flags" in menuselect.
+
+ * BUILD_CFLAGS and BUILD_LDFLAGS can now be passed to menuselect, e.g.,
+ make BUILD_CFLAGS="whatever" BUILD_LDFLAGS="whatever"
+
+ * Remove "asterisk/version.h" in favor of "asterisk/ast_version.h". If you
+ previously parsed the header file to obtain the version of Asterisk, you
+ will now have to go through Asterisk to get the version information.
+
+
+Applications
+-------------------
+
+Bridge
+-------------------
+ * Added 'F()' option. Similar to the dial option, this can be supplied with
+ arguments indicating where the callee should go after the caller is hung up,
+ or without options specified, the priority after the Queue will be used.
+
+
+ConfBridge
+-------------------
+ * Added menu action admin_toggle_mute_participants. This will mute / unmute
+ all non-admin participants on a conference. The confbridge configuration
+ file also allows for the default sounds played to all conference users when
+ this occurs to be overriden using sound_participants_unmuted and
+ sound_participants_muted.
+
+ * Added menu action participant_count. This will playback the number of
+ current participants in a conference.
+
+ * Added announcement configuration option to user profile. If set the sound
+ file will be played to the user, and only the user, upon joining the
+ conference bridge.
+
+ * Added record_file_append option that defaults to "yes", but if set to no
+ will create a new file between each start/stop recording.
+
+
+Dial
+-------------------
+ * Added 'b' and 'B' options to Dial that execute a Gosub on callee and caller
+ channels respectively before the callee channels are called.
+
+
+ExternalIVR
+-------------------
+ * Added support for IPv6.
+
+ * Add interrupt ('I') command to ExternalIVR. Sending this command from an
+ external process will cause the current playlist to be cleared, including
+ stopping any audio file that is currently playing. This is useful when you
+ want to interrupt audio playback only when specific DTMF is entered by the
+ caller.
+
+
+FollowMe
+-------------------
+ * A new option, 'I' has been added to app_followme. By setting this option,
+ Asterisk will not update the caller with connected line changes when they
+ occur. This is similar to app_dial and app_queue.
+
+ * The 'N' option is now ignored if the call is already answered.
+
+ * Added 'b' and 'B' options to FollowMe that execute a Gosub on callee
+ and caller channels respectively before the callee channels are called.
+
+ * The winning FollowMe outgoing call is now put on hold if the caller put it on
+ hold.
+
+
+MixMonitor
+------------------
+ * MixMonitor hooks now have IDs associated with them which can be used to
+ assign a target to StopMixMonitor. Use of MixMonitor's i(variable) option
+ will allow storage of the MixMonitor ID in a channel variable. StopMixmonitor
+ now accepts that ID as an argument.
+
+ * Added 'm' option, which stores a copy of the recording as a voicemail in the
+ indicated mailboxes.
+
+
+MySQL
+-------------------
+ * The connect action in app_mysql now allows you to specify a port number to
+ connect to. This is useful if you run a MySQL server on a non-standard
+ port number.
+
+
+OSP Applications
+-------------------
+ * Increased the default number of allowed destinations from 5 to 12.
+
+
+Page
+-------------------
+ * The app_page application now no longer depends on DAHDI or app_meetme. It
+ has been re-architected to use app_confbridge internally.
+
+
+Queue
+-------------------
+ * Added queue options autopausebusy and autopauseunavail for automatically
+ pausing a queue member when their device reports busy or congestion.
+
+ * The 'ignorebusy' option for queue members has been deprecated in favor of
+ the option 'ringinuse. Also a 'queue set ringinuse' CLI command has been
+ added as well as an AMI action 'QueueMemberRingInUse' to set this variable on a
+ per interface basis. Individual ringinuse values can now be set in
+ queues.conf via an argument to member definitions. Lastly, the queue
+ 'ringinuse' setting now only determines defaults for the per member
+ 'ringinuse' setting and does not override per member settings like it does
+ in earlier versions.
+
+ * Added 'F()' option. Similar to the dial option, this can be supplied with
+ arguments indicating where the callee should go after the caller is hung up,
+ or without options specified, the priority after the Queue will be used.
+
+ * Added new option log_member_name_as_agent, which will cause the membername to
+ be logged in the agent field for ADDMEMBER and REMOVEMEMBER queue events if a
+ state_interface has been set.
+
+ * Add queue monitoring hints. exten => 8501,hint,Queue:markq.
+
+ * App_queue will now play periodic announcements for the caller that
+ holds the first position in the queue while waiting for answer.
+
+SayUnixTime
+------------------
+ * Added 'j' option to SayUnixTime. SayUnixTime no longer auto jumps to extension
+ when receiving DTMF. Use the 'j' option to enable extension jumping. Also
+ changed arguments to SayUnixTime so that every option is truly optional even
+ when using multiple options (so that j option could be used without having to
+ manually specify timezone and format) There are other benefits, e.g., format
+ can now be used without specifying time zone as well.
+
+
+Voicemail
+------------------
+ * Addition of the VM_INFO function - see Function changes.
+
+ * The imapserver, imapport, and imapflags configuration options can now be
+ overriden on a user by user basis.
+
+ * When voicemail plays a message's envelope with saycid set to yes, when
+ reaching the caller id field it will play a recording of a file with the same
+ base name as the sender's callerid if there is a similarly named file in
+ <astspooldir>/recordings/callerids/
+
+ * Voicemails now contains a unique message identifier "msg_id", which is stored
+ in the message envelope with the sound files. IMAP backends will now store
+ the message identifiers with a header of "X-Asterisk-VM-Message-ID". ODBC
+ backends will store the message identifier in a "msg_id" column. See
+ UPGRADE.txt for more information.
+
+ * Added VoiceMailPlayMsg application. This application will play a single
+ voicemail message from a mailbox. The result of the application, SUCCESS or
+ FAILED, is stored in the channel variable VOICEMAIL_PLAYBACKSTATUS.
+
+
+Functions
+------------------
+ * Hangup handlers can be attached to channels using the CHANNEL() function.
+ Hangup handlers will run when the channel is hung up similar to the h
+ extension. The hangup_handler_push option will push a GoSub compatible
+ location in the dialplan onto the channel's hangup handler stack. The
+ hangup_handler_pop option will remove the last added location, and optionally
+ replace it with a new GoSub compatible location. The hangup_handler_wipe
+ option will remove all locations on the stack, and optionally add a new
+ location.
+
+ * The expression parser now recognizes the ABS() absolute value function,
+ which will convert negative floating point values to positive values.
+
+ * FAXOPT(faxdetect) will enable a generic fax detect framehook for dialplan
+ control of faxdetect.
+
+ * Addition of the VM_INFO function that can be used to retrieve voicemail
+ user information, such as the email address and full name.
+ The MAILBOX_EXISTS dialplan function has been deprecated in favour of
+ VM_INFO.
+
+ * The REDIRECTING function now supports the redirecting original party id
+ and reason.
+
+ * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
+ lets you set some of the configuration options from the [general] section
+ of features.conf on a per-channel basis. FEATUREMAP() lets you customize
+ the key sequence used to activate built-in features, such as blindxfer,
+ and automon. See the built-in documentation for details.
+
+ * MESSAGE(from) for incoming SIP messages now returns "display-name" <uri>
+ instead of simply the uri. This is the format that MessageSend() can use
+ in the from parameter for outgoing SIP messages.
+
+ * Added the PRESENCE_STATE function. This allows retrieving presence state
+ information from any presence state provider. It also allows setting
+ presence state information from a CustomPresence presence state provider.
+ See AMI/CLI changes for related commands.
+
+ * Added the AMI_CLIENT function to make manager account attributes available
+ to the dialplan. It currently supports returning the current number of
+ active sessions for a given account.
+
+ * Added support for private party ID information to CALLERID, CONNECTEDLINE,
+ and the REDIRECTING functions.
+
+
+Channel Drivers
+------------------
+
+chan_local
+------------------
+ * Added a manager event "LocalBridge" for local channel call bridges between
+ the two pseudo-channels created.
+
+
+chan_dahdi
+------------------
+ * Added dialtone_detect option for analog ports to disconnect incoming
+ calls when dialtone is detected.
+
+ * Added option colp_send to send ISDN connected line information. Allowed
+ settings are block, to not send any connected line information; connect, to
+ send connected line information on initial connect; and update, to send
+ information on any update during a call. Default is update.
+
+ * Add options namedcallgroup and namedpickupgroup to support installations
+ where a higher number of groups (>64) is required.
+
+ * Added support to use private party ID information with PRI calls.
+
+
+chan_motif
+------------------
+ * A new channel driver named chan_motif has been added which provides support for
+ Google Talk and Jingle in a single channel driver. This new channel driver includes
+ support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk,
+ hold, unhold, and ringing notification. It is also compliant with the current Jingle
+ specification, current Google Jingle specification, and the original Google Talk
+ protocol.
+
+
+chan_ooh323
+------------------
+ * Added NAT support for RTP. Setting in config is 'nat', which can be set
+ globally and overriden on a peer by peer basis.
+
+ * Direct media functionality has been added. Options in config are:
+ directmedia (directrtp) and directrtpsetup (earlydirect)
+
+ * ChannelUpdate events now contain a CallRef header.
+
+
+chan_sip
+------------------
+ * Asterisk will no longer substitute CID number for CID name in the display
+ name field if CID number exists without a CID name. This change improves
+ compatibility with certain device features such as Avaya IP500's directory
+ lookup service.
+
+ * A new setting for autocreatepeer (autocreatepeer=persistent) allows peers
+ created using that setting to not be removed during SIP reload.
+
+ * Added settings recordonfeature and recordofffeature. When receiving an INFO
+ request with a "Record:" header, this will turn the requested feature on/off.
+ Allowed values are 'automon', 'automixmon', and blank to disable. Note that
+ dynamic features must be enabled and configured properly on the requesting
+ channel for this to function properly.
+
+ * Add support to realtime for the 'callbackextension' option.
+
+ * When multiple peers exist with the same address, but differing
+ callbackextension options, incoming requests that are matched by address
+ will be matched to the peer with the matching callbackextension if it is
+ available.
+
+ * Two new NAT options, auto_force_rport and auto_comedia, have been added
+ which set the force_rport and comedia options automatically if Asterisk
+ detects that an incoming SIP request crossed a NAT after being sent by
+ the remote endpoint.
+
+ * The default global nat setting in sip.conf has been changed from force_rport
+ to auto_force_rport.
+
+ * NAT settings are now a combinable list of options. The equivalent of the
+ deprecated nat=yes is nat=force_rport,comedia. nat=no behaves as before.
+
+ * Adds an option send_diversion which can be disabled to prevent
+ diversion headers from automatically being added to INVITE requests.
+
+ * Add support for lightweight NAT keepalive. If enabled a blank packet will
+ be sent to the remote host at a given interval to keep the NAT mapping open.
+ This can be enabled using the keepalive configuration option.
+
+ * Add option 'tonezone' to specify country code for indications. This option
+ can be set both globally and overridden for specific peers.
+
+ * The SIP Security Events Framework now supports IPv6.
+
+ * Add a new setting for directmedia, 'outgoing', to alleviate INVITE glares
+ between multiple user agents. When set, for directmedia reinvites,
+ Asterisk will not send an immediate reinvite on an incoming call leg. This
+ option is useful when peered with another SIP user agent that is known to
+ send immediate direct media reinvites upon call establishment.
+
+ * Add support for WebSocket transport. This can be configured using 'ws' or 'wss'
+ as the transport.
+
+ * Add options subminexpiry and submaxexpiry to set limits of subscription
+ timer independently from registration timer settings. The setting of the
+ registration timer limits still is done by options minexpiry, maxexpiry
+ and defaultexpiry. For backwards compatibility the setting of minexpiry
+ and maxexpiry also is used to configure the subscription timer limits if
+ subminexpiry and submaxexpiry are not set in sip.conf.
+
+ * Set registration timer limits to default values when reloading sip
+ configuration and values are not set by configuration.
+
+ * Add options namedcallgroup and namedpickupgroup to support installations
+ where a higher number of groups (>64) is required.
+
+ * When a MESSAGE request is received, the address the request was received from
+ is now saved in the SIP_RECVADDR variable.
+
+ * Add ANI2/OLI parsing for SIP. The "From" header in INVITE requests is now
+ parsed for the presence of "isup-oli", "ss7-oli", or "oli" tags. If present,
+ the ANI2/OLI information is set on the channel, which can be retrieved using
+ the CALLERID function.
+
+ * Peers can now be configured to support negotiation of ICE candidates using
+ the setting icesupport. See res_rtp_asterisk changes for more information.
+
+ * Added support for format attribute negotiation. See the Codecs changes for
+ more information.
+
+ * Extra headers specified with SIPAddHeader are sent with the REFER message
+ when using Transfer application. See refer_addheaders in sip.conf.sample.
+
+ * Added support to use private party ID information with calls.
+
+ * Adds an option discard_remote_hold_retrieval that when set stops telling
+ the peer to start music on hold.
+
+
+chan_skinny
+------------------
+ * Added skinny version 17 protocol support.
+
+
+chan_unistim
+--------------------
+ * Added option 'dtmf_duration' allowing playback time of DTMF tones to be set
+
+ * Modified option 'date_format' to allow options to display date in 31Jan and Jan31
+ formats as options 0 and 1. The previous options 0 and 1 now map to options 2 and 3
+ as per the UNISTIM protocol.
+
+ * Fixed issues with dialtone not matching indications.conf and mute stopping rx
+ as well as tx. Also fixed issue with call "Timer" displaying as French "Dur\E9e"
+
+ * Added ability to use multiple lines for a single phone. This allows multiple
+ calls to occur on a single phone, using callwaiting and switching between calls.
+
+ * Added option 'sharpdial' allowing end dialing by pressing # key
+
+ * Added option 'interdigit_timer' to control phone dial timeout
+
+ * Added options 'cwstyle', 'cwvolume' controlling callwaiting appearance
+
+ * Added global 'debug' option, that enables debug in channel driver
+
+ * Added ability to translate on-screen menu in multiple languages. Tested on
+ Russian languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO 8859-4,
+ ISO 8859-5, ISO 2022-JP. Language controlled by 'language' and on-screen
+ menu of phone
+
+ * In addition to English added French and Russian languages for on-screen menus
+
+ * Reworked dialing number input: added dialing by timeout, immediate dial on
+ on dialplan compare, phone number length now not limited by screen size
+
+ * Added ability to pickup a call using features.conf defined value and
+ on-screen key
+
+
+chan_mISDN:
+------------------
+ * Add options namedcallgroup and namedpickupgroup to support installations
+ where a higher number of groups (>64) is required.
+
+ * Added support to use private party ID information with calls.
+
+
+Core
+------------------
+ * The minimum DTMF duration can now be configured in asterisk.conf
+ as "mindtmfduration". The default value is (as before) set to 80 ms.
+ (previously it was only available in source code)
+
+ * Named ACLs can now be specified in acl.conf and used in configurations that
+ use ACLs. As a general rule, if some derivative of 'permit' or 'deny' is
+ used to specify an ACL, a similar form of 'acl' will add a named ACL to the
+ working ACL. In addition, some CLI commands have been added to provide
+ show information and allow for module reloading - see CLI Changes.
+
+ * Rules in ACLs (specified using 'permit' and 'deny') can now contain multiple
+ items (separated by commas), and items in the rule can be negated by prefixing
+ them with '!'. This simplifies Asterisk Realtime configurations, since it is no
+ longer necessray to control the order that the 'permit' and 'deny' columns are
+ returned from queries.
+
+ * DUNDi now allows the built in variables ${NUMBER}, ${IPADDR} and ${SECRET} to
+ be used within the dynamic weight attribute when specifying a mapping.
+
+ * CEL backends can now be configured to show "USER_DEFINED" in the EventName
+ header, instead of putting the user defined event name there. When enabled
+ the UserDefType header is added for user defined events. This feature is
+ enabled with the setting show_user_defined.
+
+ * Macro has been deprecated in favor of GoSub. For redirecting and connected
+ line purposes use the following variables instead of their macro equivalents:
+ REDIRECTING_SEND_SUB, REDIRECTING_SEND_SUB_ARGS, CONNECTED_LINE_SEND_SUB,
+ CONNECTED_LINE_SEND_SUB_ARGS. For CCSS, use cc_callback_sub instead of
+ cc_callback_macro in channel configurations.
+
+ * Asterisk can now use a system-provided NetBSD editline library (libedit) if it
+ is available.
+
+ * Call files now support the "early_media" option to connect with an outgoing
+ extension when early media is received.
+
+ * Added support to use private party ID information with calls.
+
+
+AGI
+------------------
+ * A new channel variable, AGIEXITONHANGUP, has been added which allows
+ Asterisk to behave like it did in Asterisk 1.4 and earlier where the
+ AGI application would exit immediately after a channel hangup is detected.
+
+ * IPv6 addresses are now supported when using FastAGI (agi://). Hostnames
+ are resolved and each address is attempted in turn until one succeeds or
+ all fail.
+
+
+AMI (Asterisk Manager Interface)
+------------------
+ * The originate action now has an option "EarlyMedia" that enables the
+ call to bridge when we get early media in the call. Previously,
+ early media was disregarded always when originating calls using AMI.
+
+ * Added setvar= option to manager accounts (much like sip.conf)
+
+ * Originate now generates an error response if the extension given is not found
+ in the dialplan
+
+ * MixMonitor will now show IDs associated with the mixmonitor upon creating
+ them if the i(variable) option is used. StopMixMonitor will accept
+ MixMonitorID as an option to close specific MixMonitors.
+
+ * The SIPshowpeer manager action response field "SIP-Forcerport" has been
+ updated to include information about peers configured with
+ nat=auto_force_rport by returning "A" if auto_force_rport is set and nat is
+ detected, and "a" if it is set and nat is not detected. "Y" and "N" are still
+ returned if auto_force_rport is not enabled.
+
+ * Added SIPpeerstatus manager command which will generate PeerStatus events
+ similar to the existing PeerStatus events found in chan_sip on demand.
+
+ * Hangup now can take a regular expression as the Channel option. If you want
+ to hangup multiple channels, use /regex/ as the Channel option. Existing
+ behavior to hanging up a single channel is unchanged, but if you pass a regex,
+ the manager will send you a list of channels back that were hung up.
+
+ * Support for IPv6 addresses has been added.
+
+ * AMI Events can now be documented in the Asterisk source. Note that AMI event
+ documentation is only generated when Asterisk is compiled using 'make full'.
+ See the CLI section for commands to display AMI event information.
+
+ * The AMI Hangup event now includes the AccountCode header so you can easily
+ correlate with AMI Newchannel events.
+
+ * The QueueMemberStatus, QueueMemberAdded, and QueueMember events now include
+ the StateInterface of the queue member.
+
+ * Added AMI event SessionTimeout in the Call category that is issued when a
+ call is terminated due to either RTP stream inactivity or SIP session timer
+ expiration.
+
+ * CEL events can now contain a user defined header UserDefType. See core
+ changes for more information.
+
+ * OOH323 ChannelUpdate events now contain a CallRef header.
+
+ * Added PresenceState command. This command will report the presence state for
+ the given presence provider.
+
+ * Added Parkinglots command. This will list all parking lots as a series of
+ AMI Parkinglot events.
+
+ * Added MessageSend command. This behaves in the same manner as the
+ MessageSend application, and is a technolgoy agnostic mechanism to send out
+ of call text messages.
+
+ * Added "message" class authorization. This grants an account permission to
+ send out of call messages. Write-only.
+
+
+CLI
+-------------------
+ * The "dialplan add include" command has been modified to create context a context
+ if one does not already exist. For instance, "dialplan add include foo into bar"
+ will create context "bar" if it does not already exist.
+
+ * A "dialplan remove context" command has been added to remove a context from
+ the dialplan
+
+ * The "mixmonitor list <channel>" command will now show MixMonitor ID, and the
+ filenames of all running mixmonitors on a channel.
+
+ * The debug level of "pri set debug" is now a bitmask ranging from 0 to 15 if
+ numeric instead of 0, 1, or 2.
+
+ * "stun show status" will show a table describing how the STUN client is
+ behaving.
+
+ * "acl show [named acl]" will show information regarding a Named ACL. The
+ acl module can be reloaded with "reload acl".
+
+ * Added CLI command to display AMI event information - "manager show events",
+ which shows a list of all known and documented AMI events, and "manager show
+ event [event name]", which shows detail information about a specific AMI
+ event.
+
+ * The result of the CLI command "queue show" now includes the state interface
+ information of the queue member.
+
+ * The command "core set verbose" will now set a separate level of logging for
+ each remote console without affecting any other console.
+
+ * Added command "cdr show pgsql status" to check connection status
+
+ * "sip show channel" will now display the complete route set.
+
+ * Added "presencestate list" command. This command will list all custom
+ presence states that have been set by using the PRESENCE_STATE dialplan
+ function.
+
+ * Added "presencestate change <entity> <state>[,<subtype>[,message[,options]]]"
+ command. This changes a custom presence to a new state.
+
+
+Codecs
+-------------------
+ * Codec lists may now be modified by the '!' character, to allow succinct
+ specification of a list of codecs allowed and disallowed, without the
+ requirement to use two different keywords. For example, to specify all
+ codecs except g729 and g723, one need only specify allow=all,!g729,!g723.
+
+ * Add support for parsing SDP attributes, generating SDP attributes, and
+ passing it through. This support includes codecs such as H.263, H.264, SILK,
+ and CELT. You are able to set up a call and have attribute information pass.
+ This should help considerably with video calls.
+
+ * The iLBC codec can now use a system-provided iLBC library if one is installed,
+ just like the GSM codec.
+
+DUNDi changes
+-------------
+ * Added CLI commands dundi show hints and dundi show cache which will list DUNDi
+ 'DONTASK' hints in the cache and list all DUNDi cache entires respectively.
+
+Logging
+-------------------
+ * Asterisk version and build information is now logged at the beginning of a
+ log file.
+
+ * Threads belonging to a particular call are now linked with callids which get
+ added to any log messages produced by those threads. Log messages can now be
+ easily identified as involved with a certain call by looking at their call id.
+ Call ids may also be attached to log messages for just about any case where
+ it can be determined to be related to a particular call.
+
+ * Each logging destination and console now have an independent notion of the
+ current verbosity level. Logger.conf now allows an optional argument to
+ the 'verbose' specifier, indicating the level of verbosity sent to that
+ particular logging destination. Additionally, remote consoles now each
+ have their own verbosity level. The command 'core set verbose' will now set
+ a separate level for each remote console without affecting any other
+ console.
+
+
+Music On Hold
+-------------------
+ * Added 'announcement' option which will play at the start of MOH and between
+ songs in modes of MOH that can detect transitions between songs (eg.
+ files, mp3, etc).
+
+
+Parking
+-------------------
+ * New per parking lot options: comebackcontext and comebackdialtime. See
+ configs/features.conf.sample for more details.
+
+ * Channel variable PARKER is now set when comebacktoorigin is disabled in
+ a parking lot.
+
+ * Channel variable PARKEDCALL is now set with the name of the parking lot
+ when a timeout occurs.
+
+
+CDRs
+-------------------
+
+CDR Postgresql Driver
+-------------------
+ * Added command "cdr show pgsql status" to check connection status
+
+
+CDR Adaptive ODBC Driver
+-------------------
+ * Added schema option for databases that support specifying a schema.
+
+
+Resource Modules
+-------------------
+
+Calendars
+-------------------
+ * A CALENDAR_SUCCESS=1/0 channel variable is now set to show whether or not
+ CALENDAR_WRITE has completed successfully.
+
+
+res_rtp_asterisk
+-------------------
+ * A new option, 'probation' has been added to rtp.conf
+ RTP in strictrtp mode can now require more than 1 packet to exit learning
+ mode with a new source (and by default requires 4). The probation option
+ allows the user to change the required number of packets in sequence to any
+ desired value. Use a value of 1 to essentially restore the old behavior.
+ Also, with strictrtp on, Asterisk will now drop all packets until learning
+ mode has successfully exited. These changes are based on how pjmedia handles
+ media sources and source changes.
+
+ * Add support for ICE/STUN/TURN in res_rtp_asterisk. This option can be
+ enabled or disabled using the icesupport setting. A variety of other
+ settings have been introduced to configure STUN/TURN connections.
+
+
+res_corosync
+-------------------
+ * A new module, res_corosync, has been introduced. This module uses the
+ Corosync cluster engineer (http://www.corosync.org) to allow a local cluster
+ of Asterisk servers to both Message Waiting Indication (MWI) and/or
+ Device State (presence) information. This module is very similar to, and
+ is a replacement for the res_ais module that was in previous releases of
+ Asterisk.
+
+
+res_xmpp
+-------------------
+ * This module adds a cleaned up, drop-in replacement for res_jabber called
+ res_xmpp. This provides the same externally facing functionality but is
+ implemented differently internally. res_jabber has been deprecated in favor
+ of res_xmpp; please see the UPGRADE.txt file for more information.
+
+
+Scripts
+-------------------
+ * The safe_asterisk script has been updated to allow several of its parameters
+ to be set from environment variables. This also enables a custom run
+ directory of Asterisk to be specified, instead of defaulting to /tmp.
+
+ * The live_ast script will now look for the LIVE_AST_BASE_DIR variable and use
+ its value to determine the directory to assume is the top-level directory of
+ the source tree. If the variable is not set, it defaults to the current
+ behavior and uses the current working directory.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 1.8 to Asterisk 10 -------------------
+------------------------------------------------------------------------------
+
+Text Messaging
+--------------
+ * Asterisk now has protocol independent support for processing text messages
+ outside of a call. Messages are routed through the Asterisk dialplan.
+ SIP MESSAGE and XMPP are currently supported. There are options in
+ jabber.conf and sip.conf to allow enabling these features.
+ -> jabber.conf: see the "sendtodialplan" and "context" options.
+ -> sip.conf: see the "accept_outofcall_message", "auth_message_requests"
+ and "outofcall_message_context" options.
+ The MESSAGE() dialplan function and MessageSend() application have been
+ added to go along with this functionality. More detailed usage information
+ can be found on the Asterisk wiki (http://wiki.asterisk.org/).
+ * If real-time text support (T.140) is negotiated, it will be preferred for
+ sending text via the SendText application. For example, via SIP, messages
+ that were once sent via the SIP MESSAGE request would be sent via RTP if
+ T.140 text is negotiated for a call.
+
+Parking
+-------
+ * parkedmusicclass can now be set for non-default parking lots.
+
+Asterisk Manager Interface
+--------------------------
+ * PeerStatus now includes Address and Port.
+ * Added Hold events for when the remote party puts the call on and off hold
+ for chan_dahdi ISDN channels.
+ * Added new action MeetmeListRooms to list active conferences (shows same
+ data as "meetme list" at the CLI).
+ * DAHDIShowChannels, SIPshowpeer, SIPpeers, and IAXpeers now contains a
+ Description field that is set by 'description' in the channel configuration
+ file.
+ * Added Uniqueid header to UserEvent.
+ * Added new action FilterAdd to control event filters for the current session.
+ This requires the system permission and uses the same filter syntax as
+ filters that can be defined in manager.conf
+ * The Unlink event is now a Bridge event with Bridgestatus: Unlink. Previous
+ versions had some instances of the event converted, but others were left
+ as-is. All Unlink events should now be converted to Bridge events. The AMI
+ protocol version number was incremented to 1.2 as a result of this change.
+
+Asterisk HTTP Server
+--------------------------
+ * The HTTP Server can bind to IPv6 addresses.
+
+chan_dahdi
+--------------------------
+ * Busy tone patterns featuring 2 silence and 2 tone lengths can now be used
+ with busydetect. usage example: busypattern=200,200,200,600
+
+CLI Changes
+--------------------------
+ * New 'gtalk show settings' command showing the current settings loaded from
+ gtalk.conf.
+ * The 'logger reload' command now supports an optional argument, specifying an
+ alternate configuration file to use.
+ * 'dialplan add extension' command will now automatically create a context if
+ the specified context does not exist with a message indicated it did so.
+ * 'sip show peers', 'iax show peers', and 'dahdi show peers' now contains a
+ Description field which can be populated with 'description' in the channel
+ configuration files (sip.conf, iax2.conf, and chan_dahdi.conf).
+
+CDR
+--------------------------
+ * The filter option in cdr_adaptive_odbc now supports negating the argument,
+ thus allowing records which do NOT match the specified filter.
+ * Added ability to log CONGESTION calls to CDR
+
+CODECS
+--------------------------
+ * Ability to define custom SILK formats in codecs.conf.
+ * Addition of speex32 audio format with translation.
+ * CELT codec pass-through support and ability to define
+ custom CELT formats in codecs.conf.
+ * Ability to read raw signed linear files with sample rates
+ ranging from 8khz - 192khz. The new file extensions introduced
+ are .sln12, .sln24, .sln32, .sln44, .sln48, .sln96, .sln192.
+ * Due to protocol limitations, channel drivers other than SIP (eg. IAX2, MGCP,
+ Skinny, H.323, etc) can still only support the following codecs:
+ Audio: ulaw, alaw, slin, slin16, g719, g722, g723, g726, g726aal2, g729, gsm,
+ siren7, siren14, speex, speex16, ilbc, lpc10, adpcm
+ Video: h261, h263, h263p, h264, mpeg4
+ Image: jpeg, png
+ Text: red, t140
+
+ConfBridge
+--------------------------
+ * New highly optimized and customizable ConfBridge application capable of
+ mixing audio at sample rates ranging from 8khz-96khz.
+ * CONFBRIDGE dialplan function capable of creating dynamic ConfBridge user
+ and bridge profiles on a channel.
+ * CONFBRIDGE_INFO dialplan function capable of retrieving information
+ about a conference such as locked status and number of parties, admins,
+ and marked users.
+ * Addition of video_mode option in confbridge.conf for adding video support
+ into a bridge profile.
+ * Addition of the follow_talker video_mode in confbridge.conf. This video
+ mode dynamically switches the video feed to always display the loudest talker
+ supplying video in the conference.
+
+Dialplan Variables
+------------------
+ * Added ASTETCDIR, ASTMODDIR, ASTVARLIBDIR, ASTDBDIR, ASTKEYDIR, ASTDATADIR,
+ ASTAGIDIR, ASTSPOOLDIR, ASTRUNDIR, ASTLOGDIR which hold the equivalent
+ variables from asterisk.conf.
+
+Dialplan Functions
+------------------
+ * Addition of the JITTERBUFFER dialplan function. This function allows
+ for jitterbuffering to occur on the read side of a channel. By using
+ this function conference applications such as ConfBridge and MeetMe can
+ have the rx streams jitterbuffered before conference mixing occurs.
+ * Added DB_KEYS, which lists the next set of keys in the Asterisk database
+ hierarchy.
+ * Added STRREPLACE function. This function let's the user search a variable
+ for a given string to replace with another string as many times as the
+ user specifies or just throughout the whole string.
+ * Added option to CHANNEL(pickupgroup) allow reading and setting the pickupgroup of channel.
+ * Mark VALID_EXTEN() deprecated in favor of DIALPLAN_EXISTS()
+ * Added extensions to chan_ooh323 in function CHANNEL()
+
+libpri channel driver (chan_dahdi) DAHDI changes
+--------------------------
+ * Added moh_signaling option to specify what to do when the channel's bridged
+ peer puts the ISDN channel on hold.
+ * Added display_send and display_receive options to control how the display ie
+ is handled. To send display text from the dialplan use the SendText()
+ application when the option is enabled.
+ * Added mcid_send option to allow sending a MCID request on a span.
+
+Calendaring
+--------------------------
+ * Added setvar option to calendar.conf to allow setting channel variables on
+ notification channels.
+ * Added "calendar show types" CLI command to list registered calendar
+ connectors.
+
+MixMonitor
+--------------------------
+ * Added two new options, r and t with file name arguments to record
+ single direction (unmixed) audio recording separate from the bidirectional
+ (mixed) recording. The mixed file name argument is optional now as long
+ as at least one recording option is used.
+
+FollowMe
+--------------------------
+ * Added a new option, l, which will disable local call optimization for
+ channels involved with the FollowMe thread. Use this option to improve
+ compatability for a FollowMe call with certain dialplan apps, options, and
+ functions.
+
+Meetme
+--------------------------
+ * Added option "k" that will automatically close the conference when there's
+ only one person left when a user exits the conference.
+
+CEL
+--------------------------
+ * cel_pgsql now supports the 'extra' column for data added using the
+ CELGenUserEvent() application.
+
+pbx_lua
+--------------------------
+ * Support for defining hints has been added to pbx_lua. See the 'hints' table
+ in the sample extensions.lua file for syntax details.
+ * Applications that perform jumps in the dialplan such as Goto will now
+ execute properly. When pbx_lua detects that the context, extension, or
+ priority we are executing on has changed it will immediately return control
+ to the asterisk PBX engine. Currently the engine cannot detect a Goto to
+ the priority after the currently executing priority.
+ * An autoservice is now started by default for pbx_lua channels. It can be
+ stopped and restarted using the autoservice_stop() and autoservice_start()
+ functions.
+
+res_fax
+--------------------------
+ * The ReceiveFAXStatus and SendFAXStatus manager events have been consolidated
+ into a FAXStatus event with an 'Operation' header that will be either
+ 'send', 'receive', and 'gateway'.
+ * T.38 gateway functionality has been added to res_fax (and res_fax_spandsp).
+ Set FAXOPT(gateway)=yes to enable this functionality on a channel. This
+ feature will handle converting a fax call between an audio T.30 fax terminal
+ and an IFP T.38 fax terminal.
+
+SIP Changes
+-----------
+ * Add T38 support for REJECTED state where T.38 Negotiation is explicitly rejected.
+ * Add option encryption_taglen to set auth taglen only 32 and 80 are supported currently.
+ * SIP now generates security events using the Security Events Framework for REGISTER and INVITE.
+
+Queue changes
+-------------
+ * Added general option negative_penalty_invalid default off. when set
+ members are seen as invalid/logged out when there penalty is negative.
+ for realtime members when set remove from queue will set penalty to -1.
+ * Added queue option autopausedelay when autopause is enabled it will be
+ delayed for this number of seconds since last successful call if there
+ was no prior call the agent will be autopaused immediately.
+ * Added member option ignorebusy this when set and ringinuse is not
+ will allow per member control of multiple calls as ringinuse does for
+ the Queue.
+
+Applications
+------------
+ * Added 'v' option to MeetMe to play voicemail greetings when a user joins/leaves
+ a MeetMe conference
+ * Added 'k' option to MeetMe to automatically kill the conference when there's only
+ one participant left (much like a normal call bridge)
+ * Added extra argument to Originate to set timeout.
+
+Asterisk Database
+-----------------
+ * The internal Asterisk database has been switched from Berkeley DB 1.86 to
+ SQLite 3. An existing Berkeley astdb file can be converted with the astdb2sqlite3
+ utility in the UTILS section of menuselect. If an existing astdb is found and no
+ astdb.sqlite3 exists, astdb2sqlite3 will be compiled automatically. Asterisk will
+ convert an existing astdb to the SQLite3 version automatically at runtime.
+
+Asterisk Modules
+----------------
+ * Modules marked as deprecated are no longer marked as building by default. Enabling
+ these modules is still available via menuselect.
+
+IAX2 Changes
+------------
+ * authdebug is now disabled by default. To enable this functionality again
+ set authdebug = yes in iax.conf.
+
+RTP Changes
+-----------
+ * The rtp.conf setting "strictrtp" is now enabled by default. In previous
+ releases it was disabled.
+
+PBX Core
+--------
+ * The PBX core previously made a call with a non-existing extension test for
+ extension s@default and jump there if the extension existed.
+ This was a bad default behaviour and violated the principle of least surprise.
+ It has therefore been changed in this release. It may affect some
+ applications and configurations that rely on this behaviour. Most channel
+ drivers have avoided this for many releases by testing whether the extension
+ called exists before starting the PBX and generating a local error.
+ This behaviour still exists and works as before.
+
+ Extension "s" is used when no extension is given in a channel driver,
+ like immediate answer in DAHDI or calling to a domain with no user part
+ in a SIP uri.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
+------------------------------------------------------------------------------
+
+SIP Changes
+-----------
+ * Due to potential username discovery vulnerabilities, the 'nat' setting in sip.conf
+ now defaults to force_rport. It is very important that phones requiring nat=no be
+ specifically set as such instead of relying on the default setting. If at all
+ possible, all devices should have nat settings configured in the general section as
+ opposed to configuring nat per-device.
+ * Added preferred_codec_only option in sip.conf. This feature limits the joint
+ codecs sent in response to an INVITE to the single most preferred codec.
+ * Added SIP_CODEC_OUTBOUND dialplan variable which can be used to set the codec
+ to be used for the outgoing call. It must be one of the codecs configured
+ for the device.
+ * Added tlsprivatekey option to sip.conf. This allows a separate .pem file
+ to be used for holding a private key. If tlsprivatekey is not specified,
+ tlscertfile is searched for both public and private key.
+ * Added tlsclientmethod option to sip.conf. This allows the protocol for
+ outbound client connections to be specified.
+ * The sendrpid parameter has been expanded to include the options
+ 'rpid' and 'pai'. Setting sendrpid to 'rpid' will cause Remote-Party-ID
+ header to be sent (equivalent to setting sendrpid=yes) and setting
+ sendrpid to 'pai' will cause P-Asserted-Identity header to be sent.
+ * The 'ignoresdpversion' behavior has been made automatic when the SDP received
+ is in response to a T.38 re-INVITE that Asterisk initiated. In this situation,
+ since the call will fail if Asterisk does not process the incoming SDP, Asterisk
+ will accept the SDP even if the SDP version number is not properly incremented,
+ but will generate a warning in the log indicating that the SIP peer that sent
+ the SDP should have the 'ignoresdpversion' option set.
+ * The 'nat' option has now been been changed to have yes, no, force_rport, and
+ comedia as valid values. Setting it to yes forces RFC 3581 behavior and enables
+ symmetric RTP support. Setting it to no only enables RFC 3581 behavior if the
+ remote side requests it and disables symmetric RTP support. Setting it to
+ force_rport forces RFC 3581 behavior and disables symmetric RTP support.
+ Setting it to comedia enables RFC 3581 behavior if the remote side requests it
+ and enables symmetric RTP support.
+ * Slave SIP channels now set HASH(SIP_CAUSE,<slave-channel-name>) on each
+ response. This permits the master channel to know how each channel dialled
+ in a multi-channel setup resolved in an individual way. This carries a
+ performance penalty and can be disabled in sip.conf using the
+ 'storesipcause' option.
+ * Added 'externtcpport' and 'externtlsport' options to allow custom port
+ configuration for the externip and externhost options when tcp or tls is used.
+ * Added support for message body (stored in content variable) to SIP NOTIFY message
+ accessible via AMI and CLI.
+ * Added 'media_address' configuration option which can be used to explicitly specify
+ the IP address to use in the SDP for media (audio, video, and text) streams.
+ * Added 'unsolicited_mailbox' configuration option which specifies the virtual mailbox
+ that the new/old count should be stored on if an unsolicited MWI NOTIFY message is
+ received.
+ * Added 'use_q850_reason' configuration option for generating and parsing
+ if available Reason: Q.850;cause=<cause code> header. It is implemented
+ in some gateways for better passing PRI/SS7 cause codes via SIP.
+ * When dialing SIP peers, a new component may be added to the end of the dialstring
+ to indicate that a specific remote IP address or host should be used when dialing
+ the particular peer. The dialstring format is SIP/peer/exten/host_or_IP.
+ * SRTP SDES support for encrypting calls to/from Asterisk over SIP. The
+ ability to selectively force bridged channels to also be encrypted is also
+ implemented. Branching in the dialplan can be done based on whether or not
+ a channel has secure media and/or signaling.
+ * Added directmediapermit/directmediadeny to limit which peers can send direct media
+ to each other
+ * Added the 'snom_aoc_enabled' option to turn on support for sending Advice of
+ Charge messages to snom phones.
+ * Added support for G.719 media streams.
+ * Added support for 16khz signed linear media streams.
+ * SIP is now able to bind to and communicate with IPv6 addresses. In addition,
+ RTP has been outfitted with the same abilities.
+ * Added support for setting the Max-Forwards: header in SIP requests. Setting is
+ available in device configurations as well as in the dial plan.
+ * Addition of the 'subscribe_network_change' option for turning on and off
+ res_stun_monitor module support in chan_sip.
+ * Addition of the 'auth_options_requests' option for turning on and off
+ authentication for OPTIONS requests in chan_sip.
+
+Configuration files
+-------------------
+ * Add #tryinclude statement for config files. This provides the same
+ functionality as the #include statement however an asterisk module will
+ still load if the filename does not exist. Using the #include statement
+ Asterisk will not allow the module to load.
+
+IAX2 Changes
+-----------
+ * Added rtsavesysname option into iax.conf to allow the systname to be saved
+ on realtime updates.
+ * Added the ability for chan_iax2 to inform the dialplan whether or not
+ encryption is being used. This interoperates with the SIP SRTP implementation
+ so that a secure SIP call can be bridged to a secure IAX call when the
+ dialplan requires bridged channels to be "secure".
+ * Addition of the 'subscribe_network_change' option for turning on and off
+ res_stun_monitor module support in chan_iax.
+
+
+MGCP Changes
+------------
+ * Added ability to preset channel variables on indicated lines with the setvar
+ configuration option. Also, clearvars=all resets the list of variables back
+ to none.
+ * PacketCable NCS 1.0 support has been added for Docsis/Eurodocsis Networks.
+ See configs/res_pktccops.conf for more information.
+
+XMPP Google Talk/Jingle changes
+-------------------------------
+ * Added the externip option to gtalk.conf.
+ * Added the stunaddr option to gtalk.conf which allows for the automatic
+ retrieval of the external ip from a stun server.
+
+Applications
+------------
+ * Added 'p' option to PickupChan() to allow for picking up channel by the first
+ match to a partial channel name.
+ * Added .m3u support for Mp3Player application.
+ * Added progress option to the app_dial D() option. When progress DTMF is
+ present, those values are sent immediately upon receiving a PROGRESS message
+ regardless if the call has been answered or not.
+ * Added functionality to the app_dial F() option to continue with execution
+ at the current location when no parameters are provided.
+ * Added the 'a' option to app_dial to answer the calling channel before any
+ announcements or macros are executed.
+ * Modified app_dial to set answertime when the called channel answers even if
+ the called channel hangs up during playback of an announcement.
+ * Modified app_dial 'r' option to support an additional parameter to play an
+ indication tone from indications.conf
+ * Added c() option to app_chanspy. This option allows custom DTMF to be set
+ to cycle through the next available channel. By default this is still '*'.
+ * Added x() option to app_chanspy. This option allows DTMF to be set to
+ exit the application.
+ * The Voicemail application has been improved to automatically ignore messages
+ that only contain silence.
+ * If you set maxmsg to 0 in voicemail.conf, Voicemail will consider the
+ associated mailbox(es) to be greetings-only.
+ * The ChanSpy application now has the 'S' option, which makes the application
+ automatically exit once it hits a point where no more channels are available
+ to spy on.
+ * The ChanSpy application also now has the 'E' option, which spies on a single
+ channel and exits when that channel hangs up.
+ * The MeetMe application now turns on the DENOISE() function by default, for
+ each participant. In our tests, this has significantly decreased background
+ noise (especially noisy data centers).
+ * Voicemail now permits storage of secrets in a separate file, located in the
+ spool directory of each individual user. The control for this is located in
+ the "passwordlocation" option in voicemail.conf. Please see the sample
+ configuration for more information.
+ * The ChanIsAvail application now exposes the returned cause code using a separate
+ variable, AVAILCAUSECODE, instead of overwriting the device state in AVAILSTATUS.
+ * Added 'd' option to app_followme. This option disables the "Please hold"
+ announcement.
+ * Added 'y' option to app_record. This option enables a mode where any DTMF digit
+ received will terminate recording.
+ * Voicemail now supports per mailbox settings for folders when using IMAP storage.
+ Previously the folder could only be set per context, but has now been extended
+ using the imapfolder option.
+ * Voicemail now supports per mailbox settings for nextaftercmd and minsecs.
+ * Voicemail now allows the pager date format to be specified separately from the
+ email date format.
+ * New applications JabberJoin, JabberLeave, and JabberSendGroup have been added
+ to allow joining, leaving, and sending text to group chats.
+ * MeetMe has a new option 'G' to play an announcement before joining a conference.
+ * Page has a new option 'A(x)' which will playback an announcement simultaneously
+ to all paged phones (and optionally excluding the caller's one using the new
+ option 'n') before the call is bridged.
+ * The 'f' option to Dial has been augmented to take an optional argument. If no
+ argument is provided, the 'f' option works as it always has. If an argument is
+ provided, then the connected party information of all outgoing channels created
+ during the Dial will be set to the argument passed to the 'f' option.
+ * Dial now inherits the GOSUB_RETVAL from the peer, when the U() option runs a
+ Gosub on the peer.
+ * The OSP lookup application adds in/outbound network ID, optional security,
+ number portability, QoS reporting, destination IP port, custom info and service
+ type features.
+ * Added new application VMSayName that will play the recorded name of the voicemail
+ user if it exists, otherwise will play the mailbox number.
+ * Added custom device states to ConfBridge bridges. Use 'confbridge:<name>' to
+ retrieve state for a particular bridge, where <name> is the conference name
+ * app_directory now allows exiting at any time using the operator or pound key.
+ * Voicemail now supports setting a locale per-mailbox.
+ * Two new applications are provided for declining counting phrases in multiple
+ languages. See the application notes for SayCountedNoun and SayCountedAdj for
+ more information.
+ * Voicemail now runs the externnotify script when pollmailboxes is activated and
+ notices a change.
+ * Voicemail now includes rdnis within msgXXXX.txt file.
+ * ExternalIVR now supports IPv6 addresses.
+ * Added 'D' command to ExternalIVR. Details are available on the Asterisk wiki
+ at https://wiki.asterisk.org/wiki/x/oQBB
+ * ParkedCall and Park can now specify the parking lot to use.
+
+Dialplan Functions
+------------------
+ * SRVQUERY and SRVRESULT functions added. This can be used to query and iterate
+ over SRV records associated with a specific service. From the CLI, type
+ 'core show function SRVQUERY' and 'core show function SRVRESULT' for more
+ details on how these may be used.
+ * PITCH_SHIFT dialplan function added. This function can be used to modify the
+ pitch of a channel's tx and rx audio streams.
+ * Added new dialplan functions CONNECTEDLINE and REDIRECTING which permits
+ setting various connected line and redirecting party information.
+ * CALLERID and CONNECTEDLINE dialplan functions have been extended to
+ support ISDN subaddressing.
+ * The CHANNEL() function now supports the "name" and "checkhangup" options.
+ * For DAHDI channels, the CHANNEL() dialplan function now allows
+ the dialplan to request changes in the configuration of the active
+ echo canceller on the channel (if any), for the current call only.
+ The syntax is:
+
+ exten => s,n,Set(CHANNEL(echocan_mode)=off)
+
+ The possible values are:
+
+ on - normal mode (the echo canceller is actually reinitialized)
+ off - disabled
+ fax - FAX/data mode (NLP disabled if possible, otherwise completely
+ disabled)
+ voice - voice mode (returns from FAX mode, reverting the changes that
+ were made when FAX mode was requested)
+ * Added new dialplan function MASTER_CHANNEL(), which permits retrieving
+ and setting variables on the channel which created the current channel.
+ Administrators should take care to avoid naming conflicts, when multiple
+ channels are dialled at once, especially when used with the Local channel
+ construct (which all could set variables on the master channel). Usage
+ of the HASH() dialplan function, with the key set to the name of the slave
+ channel, is one approach that will avoid conflicts.
+ * Added new dialplan function MUTEAUDIO() for muting inbound and/or outbound
+ audio in a channel.
+ * func_odbc now allows multiple row results to be retrieved without using
+ mode=multirow. If rowlimit is set, then additional rows may be retrieved
+ from the same query by using the name of the function which retrieved the
+ first row as an argument to ODBC_FETCH().
+ * Added JABBER_RECEIVE, which permits receiving XMPP messages from the
+ dialplan. This function returns the content of the received message.
+ * Added REPLACE, which searches a given variable name for a set of characters,
+ then either replaces them with a single character or deletes them.
+ * Added PASSTHRU, which literally passes the same argument back as its return
+ value. The intent is to be able to use a literal string argument to
+ functions that currently require a variable name as an argument.
+ * HASH-associated variables now can be inherited across channel creation, by
+ prefixing the name of the hash at assignment with the appropriate number of
+ underscores, just like variables.
+ * GROUP_MATCH_COUNT has been improved to allow regex matching on category
+ * CHANNEL(secure_bridge_signaling) and CHANNEL(secure_bridge_media) to set/get
+ whether or not channels that are bridged to the current channel will be
+ required to have secure signaling and/or media.
+ * CHANNEL(secure_signaling) and CHANNEL(secure_media) to get whether or not
+ the current channel has secure signaling and/or media.
+ * For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
+ "no_media_path" option.
+ Returns "0" if there is a B channel associated with the call.
+ Returns "1" if no B channel is associated with the call. The call is either
+ on hold or is a call waiting call.
+ * Added option to dialplan function CDR(), the 'f' option
+ allows for high resolution times for billsec and duration fields.
+ * FILE() now supports line-mode and writing.
+ * Added FIELDNUM(), which returns the 1-based offset of a field in a list.
+ * FRAME_TRACE(), for tracking internal ast_frames on a channel.
+
+Dialplan Variables
+------------------
+ * Added DYNAMIC_FEATURENAME which holds the last triggered dynamic feature.
+ * Added DYNAMIC_PEERNAME which holds the unique channel name on the other side
+ and is set when a dynamic feature is triggered.
+ * Added PARKINGLOT which can be used with parkeddynamic feature.conf option
+ to dynamically create a new parking lot matching the value this varible is
+ set to.
+ * Added PARKINGDYNAMIC which represents the template parkinglot defined in
+ features.conf that should be the base for dynamic parkinglots.
+ * Added PARKINGDYNCONTEXT which tells what context a newly created dynamic
+ parkinglot should have.
+ * Added PARKINGDYNEXTEN which tells what parking exten a newly created dynamic
+ parkinglot should have.
+ * Added PARKINGDYNPOS which holds what parking positions a dynamic parkinglot
+ should have.
+
+Queue changes
+-------------
+ * Added "ready" option to QUEUE_MEMBER counting to count free agents whose wrap-up
+ timeout has expired.
+ * Added 'R' option to app_queue. This option stops moh and indicates ringing
+ to the caller when an Agent's phone is ringing. This can be used to indicate
+ to the caller that their call is about to be picked up, which is nice when
+ one has been on hold for an extened period of time.
+ * A new config option, penaltymemberslimit, has been added to queues.conf.
+ When set this option will disregard penalty settings when a queue has too
+ few members.
+ * A new option, 'I' has been added to both app_queue and app_dial.
+ By setting this option, Asterisk will not update the caller with
+ connected line changes or redirecting party changes when they occur.
+ * A 'relative-periodic-announce' option has been added to queues.conf. When
+ enabled, this option will cause periodic announce times to be calculated
+ from the end of announcements rather than from the beginning.
+ * The autopause option in queues.conf can be passed a new value, "all." The
+ result is that if a member becomes auto-paused, he will be paused in all
+ queues for which he is a member, not just the queue that failed to reach
+ the member.
+ * Added dialplan function QUEUE_EXISTS to check if a queue exists
+ * The queue logger now allows events to optionally propagate to a file,
+ even when realtime logging is turned on. Additionally, realtime logging
+ supports sending the event arguments to 5 individual fields, although it
+ will fallback to the previous data definition, if the new table layout is
+ not found.
+
+mISDN channel driver (chan_misdn) changes
+----------------------------------------
+ * Added display_connected parameter to misdn.conf to put a display string
+ in the CONNECT message containing the connected name and/or number if
+ the presentation setting permits it.
+ * Added display_setup parameter to misdn.conf to put a display string
+ in the SETUP message containing the caller name and/or number if the
+ presentation setting permits it.
+ * Made misdn.conf parameters localdialplan and cpndialplan take a -1 to
+ indicate the dialplan settings are to be obtained from the asterisk
+ channel.
+ * Made misdn.conf parameter callerid accept the "name" <number> format
+ used by the rest of the system.
+ * Made use the nationalprefix and internationalprefix misdn.conf
+ parameters to prefix any received number from the ISDN link if that
+ number has the corresponding Type-Of-Number. NOTE: This includes
+ comparing the incoming call's dialed number against the MSN list.
+ * Added the following new parameters: unknownprefix, netspecificprefix,
+ subscriberprefix, and abbreviatedprefix in misdn.conf to prefix any
+ received number from the ISDN link if that number has the corresponding
+ Type-Of-Number.
+ * Added new dialplan application misdn_command which permits controlling
+ the CCBS/CCNR functionality.
+ * Added new dialplan function mISDN_CC which permits retrieval of various
+ values from an active call completion record.
+ * For PTP, you should manually send the COLR of the redirected-to party
+ for an incomming redirected call if the incoming call could experience
+ further redirects. Just set the REDIRECTING(to-num,i) = ${EXTEN} and
+ set the REDIRECTING(to-pres) to the COLR. A call has been redirected
+ if the REDIRECTING(from-num) is not empty.
+ * For outgoing PTP redirected calls, you now need to use the inhibit(i)
+ option on all of the REDIRECTING statements before dialing the
+ redirected-to party. You still have to set the REDIRECTING(to-xxx,i)
+ and the REDIRECTING(from-xxx,i) values. The PTP call will update the
+ redirecting-to presentation (COLR) when it becomes available.
+ * Added outgoing_colp parameter to misdn.conf to filter outgoing COLP
+ information.
+
+thirdparty mISDN enhancements
+-----------------------------
+mISDN has been modified by Digium, Inc. to greatly expand facility message
+support to allow:
+ * Enhanced COLP support for call diversion and transfer.
+ * CCBS/CCNR support.
+
+The latest modified mISDN v1.1.x based version is available at:
+http://svn.digium.com/svn/thirdparty/mISDN/trunk
+http://svn.digium.com/svn/thirdparty/mISDNuser/trunk
+
+Tagged versions of the modified mISDN code are available under:
+http://svn.digium.com/svn/thirdparty/mISDN/tags
+http://svn.digium.com/svn/thirdparty/mISDNuser/tags
+
+libpri channel driver (chan_dahdi) DAHDI changes
+-------------------------------------------
+ * The channel variable PRIREDIRECTREASON is now just a status variable
+ and it is also deprecated. Use the REDIRECTING(reason) dialplan function
+ to read and alter the reason.
+ * For Q.SIG and ETSI PRI/BRI-PTP, you should manually send the COLR of the
+ redirected-to party for an incomming redirected call if the incoming call
+ could experience further redirects. Just set the
+ REDIRECTING(to-num,i) = CALLERID(dnid) and set the REDIRECTING(to-pres)
+ to the COLR. A call has been redirected if the REDIRECTING(count) is not
+ zero.
+ * For outgoing Q.SIG and ETSI PRI/BRI-PTP redirected calls, you need to
+ use the inhibit(i) option on all of the REDIRECTING statements before
+ dialing the redirected-to party. You still have to set the
+ REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The call
+ will update the redirecting-to presentation (COLR) when it becomes available.
+ * Added the ability to ignore calls that are not in a Multiple Subscriber
+ Number (MSN) list for PTMP CPE interfaces.
+ * Added dynamic range compression support for dahdi channels. It is
+ configured via the rxdrc and txdrc parameters in chan_dahdi.conf.
+ * Added support for ISDN calling and called subaddress with partial support
+ for connected line subaddress.
+ * Added support for BRI PTMP NT mode. (Requires latest LibPRI.)
+ * Added handling of received HOLD/RETRIEVE messages and the optional ability
+ to transfer a held call on disconnect similar to an analog phone.
+ * Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
+ Will reroute/deflect an outgoing call when receive the message.
+ Can use the DAHDISendCallreroutingFacility to send the message for the
+ supported switches.
+ * Added standard location to add options to chan_dahdi dialing:
+ Dial(DAHDI/g1[/extension[/options]])
+ Current options:
+ K(<keypad_digits>)
+ R Reverse charging indication
+ * Added Reverse Charging Indication (Collect calls) send/receive option.
+ Send reverse charging in SETUP message with the chan_dahdi R dialing option.
+ Dial(DAHDI/g1/extension/R)
+ Access received reverse charge in SETUP message by: ${CHANNEL(reversecharge)}
+ (requires latest LibPRI)
+ * Added ability to send/receive keypad digits in the SETUP message.
+ Send keypad digits in SETUP message with the chan_dahdi K(<keypad_digits>)
+ dialing option. Dial(DAHDI/g1/[extension]/K(<keypad_digits>))
+ Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
+ (requires latest LibPRI)
+ * Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages
+ to eliminate tromboned calls. A tromboned call goes out an interface and comes
+ back into the same interface. Tromboned calls happen because of call routing,
+ call deflection, call forwarding, and call transfer.
+ * Added the ability to send and receive ETSI Advice-Of-Charge messages.
+ * Added the ability to support call waiting calls. (The SETUP has no B channel
+ assigned.)
+ * Added Malicious Call ID (MCID) event to the AMI call event class.
+ * Added Message Waiting Indication (MWI) support for ISDN PTMP endpoints (phones).
+
+Asterisk Manager Interface
+--------------------------
+ * The Hangup action now accepts a Cause header which may be used to
+ set the channel's hangup cause.
+ * sslprivatekey option added to manager.conf and http.conf. Adds the ability
+ to specify a separate .pem file to hold a private key. By default sslcert
+ is used to hold both the public and private key.
+ * Options in manager.conf and http.conf with the 'ssl' prefix have been replaced
+ for options containing the 'tls' prefix. For example, 'sslenable' is now
+ 'tlsenable'. This has been done in effort to keep ssl and tls options consistent
+ across all .conf files. All affected sample.conf files have been modified to
+ reflect this change. Previous options such as 'sslenable' still work,
+ but options with the 'tls' prefix are preferred.
+ * Added a MuteAudio AMI action for muting inbound and/or outbound audio
+ in a channel. (res_mutestream.so)
+ * The configuration file manager.conf now supports a channelvars option, which
+ specifies a list of channel variables to include in each channel-oriented
+ event.
+ * The redirect command now has new parameters ExtraContext, ExtraExtension,
+ and ExtraPriority to allow redirecting the second channel to a different
+ location than the first.
+ * Added new event "JabberStatus" in the Jabber module to monitor buddies
+ status.
+ * Added a "MixMonitorMute" AMI action for muting inbound and/or outbound audio
+ in a MixMonitor recording.
+ * The 'iax2 show peers' output is now similar to the expected output of
+ 'sip show peers'.
+ * Added Advice-Of-Charge events (AOC-S, AOC-D, and AOC-E) in the new
+ aoc event class.
+ * Added Advice-Of-Charge manager action, AOCMessage, for generating AOC-D and
+ AOC-E messages on a channel.
+ * A DBGetComplete event now follows a DBGetResponse, to make the DBGet action
+ conform more closely to similar events.
+ * Added a new eventfilter option per user to allow whitelisting and blacklisting
+ of events.
+ * Added optional parkinglot variable for park command.
+ * Added ConnectedLineNum and ConnectedLineName headers to AMI events/responses
+ if CallerIDNum and CallerIDName headers are also present.
+
+Channel Event Logging
+---------------------
+ * A new interface, CEL, is introduced here. CEL logs single events, much like
+ the AMI, but it differs from the AMI in that it logs to db backends much
+ like CDR does; is based on the event subsystem introduced by Russell, and
+ can share in all its benefits; allows multiple backends to operate like CDR;
+ is specialized to event data that would be of concern to billing systems,
+ like CDR. Backends for logging and accounting calls have been produced,
+ but a new CDR backend is still in development.
+
+CDR
+---
+ * 'linkedid' and 'peeraccount' are new CDR fields available to CDR aficionados.
+ linkedid is based on uniqueID, but spreads to other channels as transfers, dials,
+ etc are performed. Thus the pieces of CDR can be grouped into multilegged sets.
+ * Multiple files and formats can now be specified in cdr_custom.conf.
+ * cdr_syslog has been added which allows CDRs to be written directly to syslog.
+ See configs/cdr_syslog.conf.sample for more information.
+ * A 'sequence' field has been added to CDRs which can be combined with
+ linkedid or uniqueid to uniquely identify a CDR.
+ * Handling of billsec and duration field has changed. If your table definition
+ specifies those fields as float,double or similar they will now be logged with
+ microsecond accuracy instead of a whole integer.
+
+Calendaring for Asterisk
+------------------------
+ * A new set of modules were added supporting calendar integration with Asterisk.
+ Dialplan functions for reading from and writing to calendars are included,
+ as well as the ability to execute dialplan logic upon calendar event notifications.
+ iCalendar, CalDAV, and Exchange Server calendars (via res_calendar_exchange for
+ Exchange Server 2003 with no write or attendee support, and res_calendar_ews for
+ Exchange Server 2007+ with full write and attendee support) are supported (Exchange
+ 2003 support does not support forms-based authentication).
+
+Call Completion Supplementary Services for Asterisk
+---------------------------------------------------
+ * Call completion support has been added for SIP, DAHDI/ISDN, and DAHDI/analog.
+ DAHDI/ISDN supports call completion for the following switch types:
+ EuroIsdn(ETSI) for PTP and PTMP modes, and Qsig.
+ See https://wiki.asterisk.org/wiki/x/2ABQ for details.
+
+Multicast RTP Support
+---------------------
+ * A new RTP engine and channel driver have been added which supports Multicast RTP.
+ The channel driver can be used with the Page application to perform multicast RTP
+ paging. The dial string format is: MulticastRTP/<type>/<destination>/<control address>
+ Type can be either basic or linksys.
+ Destination is the IP address and port for the RTP packets.
+ Control address is specific to the linksys type and is used for sending the control
+ packets unique to them.
+
+Security Events Framework
+-------------------------
+ * Asterisk has a new C API for reporting security events. The module res_security_log
+ sends these events to the "security" logger level. Currently, AMI is the only
+ Asterisk component that reports security events. However, SIP support will be
+ coming soon. For more information on the security events framework, see the
+ "Asterisk Security Framework" section of the Asterisk wiki at
+ https://wiki.asterisk.org/wiki/x/wgBQ
+ * SIP support was added in Asterisk 10
+ * This API now supports IPv6 addresses
+
+Fax
+---
+ * A technology independent fax frontend (res_fax) has been added to Asterisk.
+ * A spandsp based fax backend (res_fax_spandsp) has been added.
+ * The app_fax module has been deprecated in favor of the res_fax module and
+ the new res_fax_spandsp backend.
+ * The SendFAX and ReceiveFAX applications now send their log messages to a
+ 'fax' logger level, instead of to the generic logger levels. To see these
+ messages, the system's logger.conf file will need to direct the 'fax' logger
+ level to one or more destinations; the logger.conf.sample file includes an
+ example of how to do this. Note that if the 'fax' logger level is *not*
+ directed to at least one destination, log messages generated by these
+ applications will be lost, and that if the 'fax' logger level is directed to
+ the console, the 'core set verbose' and 'core set debug' CLI commands will
+ have no effect on whether the messages appear on the console or not.
+
+Miscellaneous
+-------------
+ * The transmit_silence_during_record option in asterisk.conf.sample has been removed.
+ Now, in order to enable transmitting silence during record the transmit_silence
+ option should be used. transmit_silence_during_record remains a valid option, but
+ defaults to the behavior of the transmit_silence option.
+ * Addition of the Unit Test Framework API for managing registration and execution
+ of unit tests with the purpose of verifying the operation of C functions.
+ * SendText is now implemented in chan_gtalk and chan_jingle. It will simply send
+ XMPP text messages to the remote JID.
+ * Modules.conf has a new option - "require" - that marks a module as critical for
+ the execution of Asterisk.
+ If one of the required modules fail to load, Asterisk will exit with a return
+ code set to 2.
+ * An 'X' option has been added to the asterisk application which enables #exec support.
+ This allows #exec to be used in asterisk.conf.
+ * jabber.conf supports a new option auth_policy that toggles auto user registration.
+ * A new lockconfdir option has been added to asterisk.conf to protect the
+ configuration directory (/etc/asterisk by default) during reloads.
+ * The parkeddynamic option has been added to features.conf to enable the creation
+ of dynamic parkinglots.
+ * chan_dahdi now supports reporting alarms over AMI either by channel or span via
+ the reportalarms config option.
+ * chan_dahdi supports dialing configuring and dialing by device file name.
+ DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
+ it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
+ * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
+ False by default. If set, chan_dahdi will ignore failed 'channel' entries.
+ Handy for the above name-based syntax as it does not depend on
+ initialization order.
+ * The Realtime dialplan switch now caches entries for 1 second. This provides a
+ significant increase in performance (about 3X) for installations using this switchtype.
+ * Distributed devicestate now supports the use of the XMPP protocol, in addition to
+ AIS. For more information, please see the Distributed Device State section of the
+ Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
+ * The addition of G.719 pass-through support.
+ * Added support for 16khz Speex audio. This can be enabled by using 'allow=speex16'
+ during device configuration.
+ * The UNISTIM channel driver (chan_unistim) has been updated to support devices that
+ have less than 3 lines on the LCD.
+ * Realtime now supports database failover. See the sample extconfig.conf for details.
+ * The addition of improved translation path building for wideband codecs. Sample
+ rate changes during translation are now avoided unless absolutely necessary.
+ * The addition of the res_stun_monitor module for monitoring and reacting to network
+ changes while behind a NAT.
+ * DTMF: Normal and Reverse Twist acceptance values can be set in dsp.conf.
+ DTMF Valid/Invalid number of hits/misses can be set in dsp.conf.
+ These allow support for any Administration. Default is AT&T values.
+
+CLI Changes
+-----------
+ * The 'core set debug' and 'core set verbose' commands, in previous versions, could
+ optionally accept a filename, to apply the setting only to the code generated from
+ that source file when Asterisk was built. However, there are some modules in Asterisk
+ that are composed of multiple source files, so this did not result in the behavior
+ that users expected. In this version, 'core set debug' and 'core set verbose'
+ can optionally accept *module* names instead (with or without the .so extension),
+ which applies the setting to the entire module specified, regardless of which source
+ files it was built from.
+ * New 'manager show settings' command showing the current settings loaded from
+ manager.conf.
+ * Added 'all' keyword to the CLI command "channel request hangup" so that you can send
+ the channel hangup request to all channels.
+ * Added a "core reload" CLI command that executes a global reload of Asterisk.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 1.6.1 to Asterisk 1.6.2 -------------
+------------------------------------------------------------------------------
+
+SIP Changes
+-----------
+ * Added support for SUBSCRIBE/NOTIFY with dialog-info based call pickups.
+ Snom phones use this for call pickup of extensions that the phone is
+ subscribed to.
+ * Added support for setting the domain in the URI for caller of an
+ outbound call by using the SIPFROMDOMAIN channel variable.
+ * Added a new configuration option "remotesecret" for authentication to
+ remote services. For backwards compatibility, "secret" still has the
+ same function as before, but now you can configure both a remote secret and a
+ local secret for mutual authentication.
+ * If the channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is set,
+ the sound will be played to the target of an attended transfer
+ * Added two new configuration options, "qualifygap" and "qualifypeers", which allow
+ finer control over how many peers Asterisk will qualify and the gap between them
+ when all peers need to be qualified at the same time.
+ * Added a new 'ignoresdpversion' option to sip.conf. When this is enabled
+ (either globally or for a specific peer), chan_sip will treat any SDP data
+ it receives as new data and update the media stream accordingly. By
+ default, Asterisk will only modify the media stream if the SDP session
+ version received is different from the current SDP session version. This
+ option is required to interoperate with devices that have non-standard SDP
+ session version implementations (observed with Microsoft OCS). This option
+ is disabled by default.
+ * The parsing of register => lines in sip.conf has been modified to allow a port
+ to be present in the "user" portion. Please see the sip.conf.sample file for more
+ information
+ * Added support for subscribing to MWI on a remote server and making the status available
+ as a mailbox. Please see the sip.conf.sample file for more information.
+ * Added a function to remove SIP headers added in the dialplan before the
+ first INVITE is generated - SIPRemoveHeader()
+ * Channel variables set with setvar= in a device configuration is now
+ set both for inbound and outbound calls.
+ * Added support for ITU G.722.1 and G.722.1C (Siren7 and Siren14) media streams.
+
+IAX2 changes
+------------
+ * Added immediate option to iax.conf
+ * Added forceencryption option to iax.conf
+ * Added Encryption and Trunk status to manager command "iaxpeers"
+
+Skinny Changes
+--------------
+ * The configuration file now holds separate sections for devices and lines.
+ Please have a look at configs/skinny.conf.sample and change your skinny.conf
+ accordingly.
+
+DAHDI Changes
+-------------
+ * chan_dahdi now supports MFC/R2 signaling when Asterisk is compiled with
+ support for LibOpenR2. http://www.libopenr2.org/
+ * The UK option waitfordialtone has been added for use with BT analog
+ lines.
+ * Added a 'faxbuffers' configuration option to chan_dahdi.conf. This option
+ is used in conjunction with the 'faxdetect' configuration option. When
+ 'faxbuffers' is used and fax tones are detected, the channel will dynamically
+ switch to the configured faxbuffers policy. For example, to use 6 buffers
+ and a 'full' buffer policy for a fax transmission, add:
+ faxbuffers=>6,full
+ The faxbuffers configuration will be in affect until the call is torn down.
+ * Added service message support for 4ESS/5ESS switches.
+
+Dialplan Functions
+------------------
+ * For DAHDI channels, the CHANNEL() dialplan function now
+ supports changing the channel's buffer policy (for the current
+ call only), using this syntax:
+
+ exten => s,n,Set(CHANNEL(buffers)=6,full)
+
+ This would change the channel to the 'full' buffer policy and
+ 6 (six) buffers. Possible options for this setting are the same
+ as those in chan_dahdi.conf.
+ * Added a new dialplan function, CURLOPT, which permits setting various
+ options that may be useful with the CURL dialplan function, such as
+ cookies, proxies, connection timeouts, passwords, etc.
+ * Permit the syntax and synopsis fields of the corresponding dialplan
+ functions to be individually set from func_odbc.conf.
+ * Added debugging CLI functions to func_odbc, 'odbc read' and 'odbc write'.
+ * func_odbc now may specify an insert query to execute, when the write query
+ affects 0 rows (usually indicating that no such row exists).
+ * Added a new dialplan function, LISTFILTER, which permits removing elements
+ from a set list, by name. Uses the same general syntax as the existing CUT
+ and FIELDQTY dialplan functions, which also manage lists.
+ * Added REALTIME_FIELD and REALTIME_HASH, which should aid users in better
+ obtaining realtime data from the dialplan.
+ * Added LOCAL_PEEK, which allows access to variables in any stack frame within
+ a subroutine when using the GoSub() and Return() applications.
+ * Added AUDIOHOOK_INHERIT. For information on its use, please see the output
+ of "core show function AUDIOHOOK_INHERIT" from the CLI
+ * Added AES_ENCRYPT. For information on its use, please see the output
+ of "core show function AES_ENCRYPT" from the CLI
+ * Added AES_DECRYPT. For information on its use, please see the output
+ of "core show function AES_DECRYPT" from the CLI
+ * func_odbc now supports database transactions across multiple queries.
+
+Applications
+------------
+ * Scheduled meetme conferences may now have their end times extended by
+ using MeetMeAdmin.
+ * app_authenticate now gives the ability to select a prompt other than
+ the default.
+ * app_directory now pays attention to the searchcontexts setting in
+ voicemail.conf and will look through all contexts, if no context is
+ specified in the initial argument.
+ * A new application, Originate, has been introduced, that allows asynchronous
+ call origination from the dialplan.
+ * Voicemail now permits setting the emailsubject and emailbody per mailbox,
+ in addition to the setting in the "general" context.
+ * Added ConfBridge dialplan application which does conference bridges without
+ DAHDI. For information on its use, please see the output of
+ "core show application ConfBridge" from the CLI.
+
+Miscellaneous
+-------------
+ * The Asterisk CLI has a new command, "channel redirect", which is similar in
+ operation to the AMI Redirect action.
+ * extensions.conf now allows you to use keyword "same" to define an extension
+ without actually specifying an extension. It uses exactly the same pattern
+ as previously used on the last "exten" line. For example:
+ exten => 123,1,NoOp(something)
+ same => n,SomethingElse()
+ * musiconhold.conf classes of type 'files' can now use relative directory paths,
+ which are interpreted as relative to the astvarlibdir setting in asterisk.conf.
+ * All deprecated CLI commands are removed from the sourcecode. They are now handled
+ by the new clialiases module. See cli_aliases.conf.sample file.
+ * Times within timespecs are now accurate down to the minute. This is a change
+ from historical Asterisk, which only provided timespecs rounded to the nearest
+ even (read: evenly divisible by 2) minute mark.
+ * The realtime switch now supports an option flag, 'p', which disables searches for
+ pattern matches.
+ * In addition to a time range and date range, timespecs now accept a 5th optional
+ argument, timezone. This allows you to perform time checks on alternate
+ timezones, especially if those daylight savings time ranges vary from your
+ machine's native timezone. See GotoIfTime, ExecIfTime, IFTIME(), and timed
+ includes.
+ * The contrib/scripts/ directory now has a script called sip_nat_settings that will
+ give you the correct output for an asterisk box behind nat. It will give you the
+ externhost and localnet settings.
+ * The Asterisk core now supports ITU G.722.1 and G.722.1C media streams, and
+ can connect calls in passthrough mode, as well as record and play back files.
+ * Successful and unsuccessful call pickup can now be alerted through sounds, by
+ using pickupsound and pickupfailsound in features.conf.
+ * ASTVARRUNDIR is now set to $(localstatedir)/run/asterisk by default.
+ This means the asterisk pid file will now be in /var/run/asterisk/asterisk.pid on LINUX
+ instead of the /var/run/asterisk.pid where it used to be. This will make
+ installs as non-root easier to manage.
+
+CDR
+---
+
+* The cdr.conf file must exist and be correctly programmed in order for CDR records to
+ be written; they will no longer be explicitly written.
+
+Asterisk Manager Interface
+--------------------------
+ * When using the AMI over HTTP, you can now include a 'SuppressEvents' header (with
+ a non-empty value) in your request. If you do this, any pending AMI events will
+ *not* be included in the response to your request as they would normally, but
+ will be left in the event queue for the next request you make to retrieve. For
+ some applications, this will allow you to guarantee that you will only see
+ events in responses to 'WaitEvent' actions, and can better know when to expect them.
+ To know whether the Asterisk server supports this header or not, your client can
+ inspect the first response back from the server to see if it includes this header:
+
+ Pragma: SuppressEvents
+
+ If this is included, the server supports event suppression.
+
+ * Added 4 new Actions to list skinny device(s) and line(s)
+ SKINNYdevices
+ SKINNYshowdevice
+ SKINNYlines
+ SKINNYshowline
+
+LDAP Schema File Additions
+--------------------------
+ * Added AsteriskDialplan, AsteriskAccount and AsteriskMailbox objectClasses
+ to allow standalone dialplan, account and mailbox entries (STRUCTURAL)
+ * Added new Fields:
+ - AstAccountLanguage, AstAccountTransport, AstAccountPromiscRedir,
+ - AstAccountAccountCode, AstAccountSetVar, AstAccountAllowOverlap,
+ - AstAccountVideoSupport, AstAccountIgnoreSDPVersion
+ * Removed redundant IPaddr (there's already IPAddress)
+ - Gives more configuration Flags for SIP-Users available (tested)
+ - Allows to create Asterisk Attributes in defined Asterisk ObjectClasses
+ without extensibleObject (which really should be the last resort); gives
+ also additional possibilities for LDAP-filter
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 1.6.0 to Asterisk 1.6.1 -------------
+------------------------------------------------------------------------------
+
+Device State Handling
+---------------------
+ * The event infrastructure in Asterisk got another big update to help support
+ distributed events. It currently supports distributed device state and
+ distributed Voicemail MWI (Message Waiting Indication). A new module has
+ been merged, res_ais, which facilitates communicating events between servers.
+ It uses the SAForum AIS (Service Availability Forum Application Interface
+ Specification) CLM (Cluster Management) and EVT (Event) services to maintain
+ a cluster of Asterisk servers, and to share events between them. For more
+ information on setting this up, refer to the Distributed Device State section
+ of the Asterisk wiki at https://wiki.asterisk.org/wiki/x/jw4iAQ
+
+Dialplan Functions
+------------------
+ * Added a new dialplan function, AST_CONFIG(), which allows you to access
+ variables from an Asterisk configuration file.
+ * The JACK_HOOK function now has a c() option to supply a custom client name.
+ * Added two new dialplan functions from libspeex for audio gain control and
+ denoise, AGC() and DENOISE(). Both functions can be applied to the tx and
+ rx directions of a channel from the dialplan.
+ * The SMDI_MSG_RETRIEVE function now has the ability to search for SMDI messages
+ based on other parameters. The default is still to search based on the
+ forwarding station ID. However, there are new options that allow you to search
+ based on the message desk terminal ID, or the message desk number.
+ * TIMEOUT() has been modified to be accurate down to the millisecond.
+ * ENUM*() functions now include the following new options:
+ - 'u' returns the full URI and does not strip off the URI-scheme.
+ - 's' triggers ISN specific rewriting
+ - 'i' looks for branches into an Infrastructure ENUM tree
+ - 'd' for a direct DNS lookup without any flipping of digits.
+ * TXCIDNAME() has a new zone-suffix parameter (which defaults to 'e164.arpa')
+ * CHANNEL() now has options for the maximum, minimum, and standard or normal
+ deviation of jitter, rtt, and loss for a call using chan_sip.
+
+DAHDI channel driver (chan_dahdi) Changes
+----------------------------------------
+ * Channels can now be configured using named sections in chan_dahdi.conf, just
+ like other channel drivers, including the use of templates.
+ * The default for pridialplan has changed from 'national' to 'unknown'.
+
+PBX Changes
+-----------
+ * It is now possible to specify a pattern match as a hint. Once a phone subscribes
+ to something that matches the pattern a hint will be created using the contents
+ and variables evaluated.
+ * Dialplan matching has been extended to allow an extension to return to the
+ PBX core to wait for more digits. This is done by using the new dialplan
+ application called "Incomplete". This will permit a whole new level of
+ extension control, by giving the administrator more control over early
+ matches employing one of the short-circuit pattern match operators. Note
+ that custom applications can trigger this same behavior by returning the
+ special value AST_PBX_INCOMPLETE.
+
+Application Changes
+-------------------
+ * Directory now permits both first and last names to be matched at the same
+ time. In addition, the number of digits to enter of the name can be set in
+ the arguments to Directory; previously, you could enter only 3, regardless
+ of how many names are in your company. For large companies, this should be
+ quite helpful.
+ * Voicemail now permits a mailbox setting to wrap around from first to last
+ messages, if the "messagewrap" option is set to a true value.
+ * Voicemail now permits an external script to be run, for password validation.
+ The script should output "VALID" or "INVALID" on stdout, depending upon the
+ wish to validate or invalidate the password given. Arguments are:
+ "mailbox" "context" "oldpass" "newpass". See the sample voicemail.conf for
+ more details
+ * Dial has a new option: F(context^extension^pri), which permits a callee to
+ continue in the dialplan, at the specified label, if the caller hangs up.
+ * ChanSpy and ExtenSpy have a new option, 's' which suppresses speaking the
+ technology name (e.g. SIP, IAX, etc) of the channel being spied on.
+ * The Jack application now has a c() option to supply a custom client name.
+ * Chanspy has a new option, 'B', which can be used to "barge" on a call. This is
+ like the pre-existing whisper mode, except that the spy can also talk to the
+ participant on the bridged channel as well.
+ * Chanspy has a new option, 'n', which will allow for the spied-on party's name
+ to be spoken instead of the channel name or number. For more information on the
+ use of this option, issue the command "core show application ChanSpy" from the
+ Asterisk CLI.
+ * Chanspy has a new option, 'd', which allows the spy to use DTMF to swap between
+ spy modes. Use of this feature overrides the typical use of numeric DTMF. In other
+ words, if using the 'd' option, it is not possible to enter a number to append to
+ the first argument to Chanspy(). Pressing 4 will change to spy mode, pressing 5 will
+ change to whisper mode, and pressing 6 will change to barge mode.
+ * ExternalIVR now takes several options that affect the way it performs, as
+ well as having several new commands. Please see the External IVR page on the Asterisk
+ wiki for complete documentation: https://wiki.asterisk.org/wiki/x/oQBB
+ * Added ability to communicate over a TCP socket instead of forking a child process for the
+ ExternalIVR application.
+ * ChanIsAvail has a new option, 'a', which will return all available channels instead
+ of just the first one if you give the function more then one channel to check.
+ * PrivacyManager now takes an option where you can specify a context where the
+ given number will be matched. This way you have more control over who is allowed
+ and it stops the people who blindly enter 10 digits.
+ * ForkCDR has new options: 'a' updates the answer time on the new CDR; 'A' locks
+ answer times, disposition, on orig CDR against updates; 'D' Copies the disposition
+ from the orig CDR to the new CDR after reset; 'e' sets the 'end' time on the
+ original CDR; 'R' prevents the new CDR from being reset; 's(var=val)' adds/changes
+ the 'var' variable on the original CDR; 'T' forces ast_cdr_end(), ast_cdr_answer(),
+ obey the LOCKED flag on cdr's in the chain, and also the ast_cdr_setvar() func.
+ * The Dial() application no longer copies the language used by the caller to the callee's
+ channel. If you desire for the caller's channel's language to be used for file playback
+ to the callee, then the file specified may be prepended with "${CHANNEL(language)}/" .
+ * SendImage() no longer hangs up the channel on error; instead, it sets the
+ status variable SENDIMAGESTATUS to one of 'SUCCESS', 'FAILURE', or
+ 'UNSUPPORTED'. This change makes SendImage() more consistent with other
+ applications.
+ * Park has a new option, 's', which silences the announcement of the parking space number.
+ * A non-numeric, zero, or negative timeout specified to Dial() will now be interpreted as
+ invalid input and will be assumed to mean that no timeout is desired.
+
+SIP Changes
+-----------
+ * Added DNS manager support to registrations for peers referencing peer entries.
+ DNS manager runs in the background which allows DNS lookups to be run asynchronously
+ as well as periodically updating the IP address. These properties allow for
+ better performance as well as recovery in the event of an IP change.
+ * Performance improvements via using hash tables (astobj2) and doubly-linked lists to improve
+ load/reload of large numbers of peers/users by ~40x (for large lists of peers).
+ These changes also provide performance improvements for call setup and tear down.
+ * Added ability to specify registration expiry time on a per registration basis in
+ the register line.
+ * Added support for T140 RED - redundancy in T.140 to prevent text loss due to
+ lost packets.
+ * Added t38pt_usertpsource option. See sip.conf.sample for details.
+ * Added SIPnotify AMI command, for sending arbitrary SIP notify commands.
+ * 'sip show peers' and 'sip show users' display their entries sorted in
+ alphabetical order, as opposed to the order they were in, in the config
+ file or database.
+ * Videosupport now supports an additional option, "always", which always sets
+ up video RTP ports, even on clients that don't support it. This helps with
+ callfiles and certain transfers to ensure that if two video phones are
+ connected, they will always share video feeds.
+
+IAX Changes
+-----------
+ * Existing DNS manager lookups extended to check for SRV records.
+ * IAX2 encryption support has been improved to support periodic key rotation
+ within a call for enhanced security. The option "keyrotate" has been
+ provided to disable this functionality to preserve backwards compatibility
+ with older versions of IAX2 that do not support key rotation.
+
+CLI Changes
+-----------
+ * New CLI command, "data get <path> [<search> [<filter>]]" which retrieves the
+ data tree based on the given <path>.
+ * New CLI command "data show providers" that will display all the registered
+ callbacks.
+ * New CLI command, "config reload <file.conf>" which reloads any module that
+ references that particular configuration file. Also added "config list"
+ which shows which configuration files are in use.
+ * New CLI commands, "pri show version" and "ss7 show version" that will
+ display which version of libpri and libss7 are being used, respectively.
+ A new API call was added so trunk will now have to be compiled against
+ a versions of libpri and libss7 that have them or it will not know that
+ these libraries exist.
+ * The commands "core show globals", "core set global" and "core set chanvar" has
+ been deprecated in favor of the more semantically correct "dialplan show globals",
+ "dialplan set chanvar" and "dialplan set global".
+ * New CLI command "dialplan show chanvar" to list all variables associated
+ with a given channel.
+
+DNS manager changes
+-------------------
+ * Addresses managed by DNS manager now can check to see if there is a DNS
+ SRV record for a given domain and will use that hostname/port if present.
+
+AMI - The manager (TCP/TLS/HTTP)
+--------------------------------
+ * The Status command now takes an optional list of variables to display
+ along with channel status.
+ * The QueueEntry event now also includes the channel's uniqueid
+
+ODBC Changes
+------------
+ * res_odbc no longer has a limit of 1023 total possible unshared connections,
+ as some people were running into this limit. This limit has been increased
+ to 4.2 billion.
+
+Queue changes
+-------------
+ * The TRANSFER queue log entry now includes the the caller's original
+ position in the transferred-from queue.
+ * A new configuration option, "timeoutpriority" has been added. Please see the section labeled
+ "QUEUE TIMING OPTIONS" in configs/queues.conf.sample for a detailed explanation of the option
+ as well as an explanation about timeout options in general
+ * Added a new option - C - for forcing the "answered elsewhere" flag on
+ cancellation of calls in to members of the queue. This is to avoid the
+ call to a member of a queue having the call listed as a "missed call".
+
+Realtime changes
+----------------
+ * Several (ODBC, Postgres, MySQL, SQLite) realtime drivers have been given
+ adaptive capabilities. What this means in practical terms is that if your
+ realtime table lacks critical fields, Asterisk will now emit warnings to
+ that effect. Also, some of the realtime drivers have the ability (if
+ configured) to automatically add those columns to the table with the
+ correct type and length.
+
+Miscellaneous
+-------------
+ * The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND can now be set using
+ the 'setvar' option to cause a given audio file to be played upon completion
+ of an attended transfer. Currently it works for DAHDI, IAX2, SIP, and
+ Skinny channels only.
+ * You can now compile Asterisk against the Hoard Memory Allocator, see the
+ Hoard page on the Asterisk wiki for more information:
+ https://wiki.asterisk.org/wiki/x/pQBB
+ * Config file variables may now be appended to, by using the '+=' append
+ operator. This is most helpful when working with long SQL queries in
+ func_odbc.conf, as the queries no longer need to be specified on a single
+ line.
+ * CDR config file, cdr.conf, has an added option, "initiatedseconds",
+ which will add a second to the billsec when the ending
+ time is set, if the number in the microseconds field of the end time is
+ greater than the number of microseconds in the answer time. This allows
+ users to count the 'initiated' seconds in their billing records.
+
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 1.4.X to Asterisk 1.6.0 -------------
+------------------------------------------------------------------------------
+
+AMI - The manager (TCP/TLS/HTTP)
+--------------------------------
+ * Manager has undergone a lot of changes, all of them documented
+ on the Asterisk wiki at https://wiki.asterisk.org/wiki/x/tQBB
+ * Manager version has changed to 1.1
+ * Added a new action 'CoreShowChannels' to list currently defined channels
+ and some information about them.
+ * Added a new action 'SIPshowregistry' to list SIP registrations.
+ * Added TLS support for the manager interface and HTTP server
+ * Added the URI redirect option for the built-in HTTP server
+ * The output of CallerID in Manager events is now more consistent.
+ CallerIDNum is used for number and CallerIDName for name.
+ * Enable https support for builtin web server.
+ See configs/http.conf.sample for details.
+ * Added a new action, GetConfigJSON, which can return the contents of an
+ Asterisk configuration file in JSON format. This is intended to help
+ improve the performance of AJAX applications using the manager interface
+ over HTTP.
+ * SIP and IAX manager events now use "ChannelType" in all cases where we
+ indicate channel driver. Previously, we used a mixture of "Channel"
+ and "ChannelDriver" headers.
+ * Added a "Bridge" action which allows you to bridge any two channels that
+ are currently active on the system.
+ * Added a "ListAllVoicemailUsers" action that allows you to get a list of all
+ the voicemail users setup.
+ * Added 'DBDel' and 'DBDelTree' manager commands.
+ * cdr_manager now reports events via the "cdr" level, separating it from
+ the very verbose "call" level.
+ * Manager users are now stored in memory. If you change the manager account
+ list (delete or add accounts) you need to reload manager.
+ * Added Masquerade manager event for when a masquerade happens between
+ two channels.
+ * Added "manager reload" command for the CLI
+ * Lots of commands that only provided information are now allowed under the
+ Reporting privilege, instead of only under Call or System.
+ * The IAX* commands now require either System or Reporting privilege, to
+ mirror the privileges of the SIP* commands.
+ * Added ability to retrieve list of categories in a config file.
+ * Added ability to retrieve the content of a particular category.
+ * Added ability to empty a context.
+ * Created new action to create a new file.
+ * Updated delete action to allow deletion by line number with respect to category.
+ * Added new action insert to add new variable to category at specified line.
+ * Updated action newcat to allow new category to be inserted in file above another
+ existing category.
+ * Added new event "JitterBufStats" in the IAX2 channel
+ * Originate now requires the Originate privilege and, if you want to call out
+ to a subshell, it requires the System privilege, as well. This was done to
+ enhance manager security.
+ * Originate now accepts codec settings with "Codecs: alaw, ulaw, h264"
+ * New command: Atxfer. See https://wiki.asterisk.org/wiki/x/uABB for more details
+ or manager show command Atxfer from the CLI
+ * New command: IAXregistry. See https://wiki.asterisk.org/wiki/x/uABB for more
+ details or manager show command IAXregistry from the CLI
+
+Dialplan functions
+------------------
+ * Added the DEVICE_STATE() dialplan function which allows retrieving any device
+ state in the dialplan, as well as creating custom device states that are
+ controllable from the dialplan.
+ * Extend CALLERID() function with "pres" and "ton" parameters to
+ fetch string representation of calling number presentation indicator
+ and numeric representation of type of calling number value.
+ * MailboxExists converted to dialplan function
+ * A new option to Dial() for telling IP phones not to count the call
+ as "missed" when dial times out and cancels.
+ * Added LOCK(), TRYLOCK(), and UNLOCK(), which provide a single level dialplan
+ mutex. No deadlocks are possible, as LOCK() only allows a single lock to be
+ held for any given channel. Also, locks are automatically freed when a
+ channel is hung up.
+ * Added HINT() dialplan function that allows retrieving hint information.
+ Hints are mappings between extensions and devices for the sake of
+ determining the state of an extension. This function can retrieve the list
+ of devices or the name associated with a hint.
+ * Added EXTENSION_STATE() dialplan function which allows retrieving the state
+ of any extension.
+ * Added SYSINFO() dialplan function which allows retrieval of system information
+ * Added a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for
+ the existence of a dialplan target.
+ * Added two new dialplan functions, TOUPPER and TOLOWER, which convert a string to
+ upper and lower case, respectively.
+ * When bridging, Asterisk sets the BRIDGEPVTCALLID to the channel drivers unique
+ ID for the call (not the Asterisk call ID or unique ID), provided that the
+ channel driver supports this. For SIP, you get the SIP call-ID for the
+ bridged channel which you can store in the CDR with a custom field.
+
+CLI Changes
+-----------
+ * Added CLI permissions, config file: cli_permissions.conf
+ default is to allow all commands for every local user/group.
+ Also this new feature added three new CLI commands:
+ - cli check permissions {<username>|@<groupname>|<username>@<groupname>} [<command>]
+ - cli reload permissions
+ - cli show permissions
+ * New CLI command "core show hint" (usage: core show hint <exten>)
+ * New CLI command "core show settings"
+ * Added 'core show channels count' CLI command.
+ * Added the ability to set the core debug and verbose values on a per-file basis.
+ * Added 'queue pause member' and 'queue unpause member' CLI commands
+ * Ability to set process limits ("ulimit") without restarting Asterisk
+ * Enhanced "agi debug" to print the channel name as a prefix to the debug
+ output to make debugging on busy systems much easier.
+ * New CLI commands "dialplan set extenpatternmatching true/false"
+ * New CLI command: "core set chanvar" to set a channel variable from the CLI.
+ * Added an easy way to execute Asterisk CLI commands at startup. Any commands
+ listed in the startup_commands section of cli.conf will get executed.
+ * Added a CLI command, "devstate change", which allows you to set custom device
+ states from the func_devstate module that provides the DEVICE_STATE() function
+ and handling of the "Custom:" devices.
+ * New CLI command: "sip show sched" which shows all ast_sched entries for sip,
+ sorted into the different possible callbacks, with the number of entries
+ currently scheduled for each. Gives you a feel for how busy the sip channel
+ driver is.
+ * Added 'skinny show lines verbose' CLI command. This will show the subs for every channel.
+ * Cleanup another bunch of CLI commands. Now all modules follow the same schema.
+ (Done by lmadsen, junky and mvanbaak during the devcon 2008)
+
+SIP changes
+-----------
+ * Added a new 'faxdetect=yes|no' configuration option to sip.conf. When this
+ option is enabled, Asterisk will watch for a CNG tone in the incoming audio
+ for a received call. If it is detected, the channel will jump to the
+ 'fax' extension in the dialplan.
+ * The default SIP useragent= identifier now includes the Asterisk version
+ * A new option, match_auth_username in sip.conf changes the matching of incoming requests.
+ If set, and the incoming request carries authentication info,
+ the username to match in the users list is taken from the Digest header
+ rather than from the From: field. This feature is considered experimental.
+ * The "musiconhold" and "musicclass" settings in sip.conf are now removed,
+ since they where replaced by "mohsuggest" and "mohinterpret" in version 1.4
+ * The "localmask" setting was removed in version 1.2 and the reminder about it
+ being removed is now also removed.
+ * A new option "busylevel" for setting a level of calls where asterisk reports
+ a device as busy, to separate it from call-limit. This value is also added
+ to the SIP_PEER dialplan function.
+ * A new realtime family called "sipregs" is now supported to store SIP registration
+ data. If this family is defined, "sippeers" will be used for configuration and
+ "sipregs" for registrations. If it's not defined, "sippeers" will be used for
+ registration data, as before.
+ * The SIPPEER function have new options for port address, call and pickup groups
+ * Added support for T.140 realtime text in SIP/RTP
+ * The "checkmwi" option has been removed from sip.conf, as it is no longer
+ required due to the restructuring of how MWI is handled. See the descriptions
+ in this file of the "pollmailboxes" and "pollfreq" options to voicemail.conf
+ for more information.
+ * Added rtpdest option to CHANNEL() dialplan function.
+ * Added SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables which are set when a transfer takes place.
+ * SIP now adds a header to the CANCEL if the call was answered by another phone
+ in the same dial command, or if the new c option in dial() is used.
+ * The new default is that 100 Trying is not sent on REGISTER attempts as the RFC specifically
+ states it is not needed. For phones, however, that do require it the "registertrying" option
+ has been added so it can be enabled.
+ * A new option called "callcounter" (global/peer/user level) enables call counters needed
+ for better status reports needed for queues and SIP subscriptions. (Call-Limit was previously
+ used to enable this functionality).
+ * New settings for timer T1 and timer B on a global level or per device. This makes it
+ possible to force timeout faster on non-responsive SIP servers. These settings are
+ considered advanced, so don't use them unless you have a problem.
+ * Added a dial string option to be able to set the To: header in an INVITE to any
+ SIP uri.
+ * Added a new global and per-peer option, qualifyfreq, which allows you to configure
+ the qualify frequency.
+ * Added SIP Session Timers support (RFC 4028). This prevents stuck SIP sessions that
+ were not properly torn down due to network or endpoint failures during an established
+ SIP session.
+ * Added experimental TCP and TLS support for SIP. See https://wiki.asterisk.org/wiki/x/ygBB
+ and configs/sip.conf.sample for more information on how it is used.
+ * Added a new configuration option "authfailureevents" that enables manager events when
+ a peer can't authenticate properly.
+ * Added DNS manager support to registrations for peers not referencing a peer entry.
+
+IAX2 changes
+------------
+ * Added the trunkmaxsize configuration option to chan_iax2.
+ * Added the srvlookup option to iax.conf
+ * Added support for OSP. The token is set and retrieved through the CHANNEL()
+ dialplan function.
+
+XMPP Google Talk/Jingle changes
+-------------------------------
+ * Added the bindaddr option to gtalk.conf.
+
+Skinny changes
+-------------
+ * Added skinny show device, skinny show line, and skinny show settings CLI commands.
+ * Proper codec support in chan_skinny.
+ * Added settings for IP and Ethernet QoS requests
+
+MGCP changes
+------------
+ * Added separate settings for media QoS in mgcp.conf
+
+Console Channel Driver changes
+------------------------------
+ * Added experimental support for video send & receive to chan_oss.
+ This requires SDL and ffmpeg/avcodec, plus Video4Linux or X11 to act as
+ a video source.
+
+Phone channel changes (chan_phone)
+----------------------------------
+ * Added G729 passthrough support to chan_phone for Sigma Designs boards.
+
+H.323 channel Changes
+---------------------
+ * H323 remote hold notification support added (by NOTIFY message
+ and/or H.450 supplementary service)
+
+Local channel changes
+---------------------
+ * The device state functionality in the Local channel driver has been updated
+ to indicate INUSE or NOT_INUSE when a Local channel is being used as opposed
+ to just UNKNOWN if the extension exists.
+ * Added jitterbuffer support for chan_local. This allows you to use the
+ generic jitterbuffer on incoming calls going to Asterisk applications.
+ For example, this would allow you to use a jitterbuffer for an incoming
+ SIP call to Voicemail by putting a Local channel in the middle. This
+ feature is enabled by using the 'j' option in the Dial string to the Local
+ channel in conjunction with the existing 'n' option for local channels.
+ * A 'b' option has been added which causes chan_local to return the actual channel
+ that is behind it when queried. This is useful for transfer scenarios as the
+ actual channel will be transferred, not the Local channel.
+
+Agent channel changes
+----------------------
+ * The ackcall and endcall options are now supplemented with options acceptdtmf
+ and enddtmf. These allow for the DTMF keypress to be configurable. The options
+ default to their old hard-coded values ('#' and '*' respectively) so this should
+ not break any existing agent installations.
+
+DAHDI channel driver (chan_dahdi) Changes
+----------------------------------------
+ * SS7 support (via libss7 library)
+ * In India, some carriers transmit CID via dtmf. Some code has been added
+ that will handle some situations. The cidstart=polarity_IN choice has been added for
+ those carriers that transmit CID via dtmf after a polarity change.
+ * CID matching information is now shown when doing 'dialplan show'.
+ * Added dahdi show version CLI command.
+ * Added setvar support to chan_dahdi.conf channel entries.
+ * Added two new options: mwimonitor and mwimonitornotify. These options allow
+ you to enable MWI monitoring on FXO lines. When the MWI state changes,
+ the script specified in the mwimonitornotify option is executed. An internal
+ event indicating the new state of the mailbox is also generated, so that
+ the normal MWI facilities in Asterisk work as usual.
+ * Added signalling type 'auto', which attempts to use the same signalling type
+ for a channel as configured in DAHDI. This is primarily designed for analog
+ ports, but will also work for digital ports that are configured for FXS or FXO
+ signalling types. This mode is also the default now, so if your chan_dahdi.conf
+ does not specify signalling for a channel (which is unlikely as the sample
+ configuration file has always recommended specifying it for every channel) then
+ the 'auto' mode will be used for that channel if possible.
+ * Added a 'dahdi set dnd' command to allow CLI control of the Do-Not-Disturb
+ state for a channel; also ensured that the DNDState Manager event is
+ emitted no matter how the DND state is set or cleared.
+
+New Channel Drivers
+-------------------
+ * Added a new channel driver, chan_unistim. See the Asterisk wiki at
+ https://wiki.asterisk.org/wiki/x/vgsiAQ and configs/unistim.conf.sample
+ for details. This new channel driver allows you to use Nortel i2002,
+ i2004, and i2050 phones with Asterisk.
+ * Added a new channel driver, chan_console, which uses portaudio as a cross
+ platform audio interface. It was written as a channel driver that would
+ work with Mac CoreAudio, but portaudio supports a number of other audio
+ interfaces, as well. Note that this channel driver requires v19 or higher
+ of portaudio; older versions have a different API.
+
+DUNDi changes
+-------------
+ * Added the ability to specify arguments to the Dial application when using
+ the DUNDi switch in the dialplan.
+ * Added the ability to set weights for responses dynamically. This can be
+ done using a global variable or a dialplan function. Using the SHELL()
+ function would allow you to have an external script set the weight for
+ each response.
+ * Added two new dialplan functions, DUNDIQUERY and DUNDIRESULT. These
+ functions will allow you to initiate a DUNDi query from the dialplan,
+ find out how many results there are, and access each one.
+ * Added the ability to specify a port for a dundi peer.
+
+ENUM changes
+------------
+ * Added two new dialplan functions, ENUMQUERY and ENUMRESULT. These
+ functions will allow you to initiate an ENUM lookup from the dialplan,
+ and Asterisk will cache the results. ENUMRESULT can be used to access
+ the results without doing multiple DNS queries.
+
+Voicemail Changes
+-----------------
+ * Added the ability to customize which sound files are used for some of the
+ prompts within the Voicemail application by changing them in voicemail.conf
+ * Added the ability for the "voicemail show users" CLI command to show users
+ configured by the dynamic realtime configuration method.
+ * MWI (Message Waiting Indication) handling has been significantly
+ restructured internally to Asterisk. It is now totally event based
+ instead of polling based. The voicemail application will notify other
+ modules that have subscribed to MWI events when something in the mailbox
+ changes.
+ This also means that if any other entity outside of Asterisk is changing
+ the contents of mailboxes, then the voicemail application still needs to
+ poll for changes. Examples of situations that would require this option
+ are web interfaces to voicemail or an email client in the case of using
+ IMAP storage. So, two new options have been added to voicemail.conf
+ to account for this: "pollmailboxes" and "pollfreq". See the sample
+ configuration file for details.
+ * Added "tw" language support
+ * Added support for storage of greetings using an IMAP server
+ * Added ability to customize forward, reverse, stop, and pause keys for message playback
+ * SMDI is now enabled in voicemail using the smdienable option.
+ * A "lockmode" option has been added to asterisk.conf to configure the file
+ locking method used for voicemail, and potentially other things in the
+ future. The default is the old behavior, lockfile. However, there is a
+ new method, "flock", that uses a different method for situations where the
+ lockfile will not work, such as on SMB/CIFS mounts.
+ * Added the ability to backup deleted messages, to ease recovery in the case
+ that a user accidentally deletes a message, and discovers that they need it.
+ * Reworked the SMDI interface in Asterisk. The new way to access SMDI information
+ is through the new functions, SMDI_MSG_RETRIEVE() and SMDI_MSG(). The file
+ smdi.conf can now be configured with options to map SMDI station IDs to Asterisk
+ voicemail boxes. The SMDI interface can also poll for MWI changes when some
+ outside entity is modifying the state of the mailbox (such as IMAP storage or
+ a web interface of some kind).
+ * Added the support for marking messages as "urgent." There are two methods to accomplish
+ this. One is to pass the 'U' option to VoiceMail(). Another way to mark a message as urgent
+ is to specify "review=yes" in voicemail.conf. Doing this will cause allow the user to mark
+ the message as urgent after he has recorded a voicemail by following the voice instructions.
+ When listening to voicemails using VoiceMailMain urgent messages will be presented before other
+ messages
+ * Added "is" language support
+
+Queue changes
+-------------
+ * Added the general option 'shared_lastcall' so that member's wrapuptime may be
+ used across multiple queues.
+ * Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
+ setqueueentryvar options for each queue, see queues.conf.sample for details.
+ * Added keepstats option to queues.conf which will keep queue
+ statistics during a reload.
+ * setinterfacevar option in queues.conf also now sets a variable
+ called MEMBERNAME which contains the member's name.
+ * Added 'Strategy' field to manager event QueueParams which represents
+ the queue strategy in use.
+ * Added option to run macro when a queue member is connected to a caller,
+ see queues.conf.sample for details.
+ * app_queue now has a 'loose' option which is almost exactly like 'strict' except it
+ does not count paused queue members as unavailable.
+ * Added min-announce-frequency option to queues.conf which allows you to control the
+ minimum amount of time between queue announcements for use when the caller's queue
+ position changes frequently.
+ * Added additional information to EXITWITHTIMEOUT and EXITWITHKEY events in the
+ queue log.
+ * Added ability for non-realtime queues to have realtime members
+ * Added the "linear" strategy to queues.
+ * Added the "wrandom" strategy to queues.
+ * Added new channel variable QUEUE_MIN_PENALTY
+ * QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY may be adjusted in mid-call by defining
+ rules in queuerules.conf. See configs/queuerules.conf.sample for details
+ * Added a new parameter for member definition, called state_interface. This may be
+ used so that a member may be called via one interface but have a different interface's
+ device state reported.
+ * Added new CLI and Manager commands relating to reloading queues. From the CLI, see
+ "queue reload", "queue reset stats". Also see "manager show command QueueReload" and
+ "manager show command QueueReset."
+ * New configuration option: randomperiodicannounce. If a list of periodic announcements is
+ specified by the periodic-announce option, then one will be chosen randomly when it is time
+ to play a periodic announcment
+ * New configuration options: announce-position now takes two more values in addition to "yes" and
+ "no." Two new options, "limit" and "more," are allowed. These are tied to another option,
+ announce-position-limit. By setting announce-position to "limit" callers will only have their
+ position announced if their position is less than what is specified by announce-position-limit.
+ If announce-position is set to "more" then callers beyond the position specified by announce-position-limit
+ will be told that their are more than announce-position-limit callers waiting.
+ * Two new queue log events have been added. An ADDMEMBER event will be logged
+ when a realtime queue member is added and a REMOVEMEMBER event will be logged
+ when a realtime queue member is removed. Since there is no calling channel associated
+ with these events, the string "REALTIME" is placed where the channel's unique id
+ is typically placed.
+ * The configuration method for the "joinempty" and "leavewhenempty" options has
+ changed to a comma-separated list of methods of determining member availability
+ instead of vague terms such as "yes," "loose," "no," and "strict." These old four
+ values are still accepted for backwards-compatibility, though.
+ * The average talktime is now calculated on queues. This information is reported via the
+ CLI commands "queue show" and "queues show"; through the AMI events AgentComplete, QueueSummary,
+ and QueueParams; and through the channelvariable QUEUETALKTIME if setinterfacevar=yes is set for
+ the queue.
+
+MeetMe Changes
+--------------
+ * The 'o' option to provide an optimization has been removed and its functionality
+ has been enabled by default.
+ * When a conference is created, the UNIQUEID of the channel that caused it to be
+ created is stored. Then, every channel that joins the conference will have the
+ MEETMEUNIQUEID channel variable set with this ID. This can be used to relate
+ callers that come and go from long standing conferences.
+ * Added a new application, MeetMeChannelAdmin, which is similar to MeetMeAdmin,
+ except it does operations on a channel by name, instead of number in a conference.
+ This is a very useful feature in combination with the 'X' option to ChanSpy.
+ * Added 'C' option to Meetme which causes a caller to continue in the dialplan
+ when kicked out.
+ * Added new RealTime functionality to provide support for scheduled conferencing.
+ This includes optional messages to the caller if they attempt to join before
+ the schedule start time, or to allow the caller to join the conference early.
+ Also included is optional support for limiting the number of callers per
+ RealTime conference.
+ * Added the S() and L() options to the MeetMe application. These are pretty
+ much identical to the S() and L() options to Dial(). They let you set
+ timeouts for the conference, as well as have warning sounds played to
+ let the caller know how much time is left, and when it is running out.
+ * Added the ability to do "meetme concise" with the "meetme" CLI command.
+ This extends the concise capabilities of this CLI command to include
+ listing all conferences, instead of an addition to the other sub commands
+ for the "meetme" command.
+ * Added the ability to specify the music on hold class used to play into the
+ conference when there is only one member and the M option is used.
+ * Added MEETME_INFO dialplan function which provides a way to query
+ various properties of a Meetme conference.
+ * Added new admin features: *81: Roll call, *82: eject all, *83: mute all,
+ and *84: record in-conf
+
+Other Dialplan Application Changes
+----------------------------------
+ * Argument support for Gosub application
+ * From the to-do lists: straighten out the app timeout args:
+ Wait() app now really does 0.3 seconds- was truncating arg to an int.
+ WaitExten() same as Wait().
+ Congestion() - Now takes floating pt. argument.
+ Busy() - now takes floating pt. argument.
+ Read() - timeout now can be floating pt.
+ WaitForRing() now takes floating pt timeout arg.
+ SpeechBackground() -- clarified in the docstrings that the timeout is an integer seconds.
+ * Added 's' option to Page application.
+ * Added an optional timeout argument to the Page application.
+ * Added 'E', 'V', and 'P' commands to ExternalIVR.
+ * Added 'o' and 'X' options to Chanspy.
+ * Added a new dialplan application, Bridge, which allows you to bridge the
+ calling channel to any other active channel on the system.
+ * Added the ability to specify a music on hold class to play instead of ringing
+ for the SLATrunk application.
+ * The Read application no longer exits the dialplan on error. Instead, it sets
+ READSTATUS to ERROR, which you can catch and handle separately.
+ * Added 'm' option to Directory, which lists out names, 8 at a time, instead
+ of asking for verification of each name, one at a time.
+ * Privacy() no longer uses privacy.conf, as all options are specifiable as
+ direct options to the app.
+ * AMD() has a new "maximum word length" option. "show application AMD" from the CLI
+ for more details
+ * GotoIfTime() now may branch based on a "false" condition, like other Goto-related applications
+ * The ChannelRedirect application no longer exits the dialplan if the given channel
+ does not exist. It will now set the CHANNELREDIRECT_STATUS variable to SUCCESS upon success
+ or NOCHANNEL if the given channel was not found.
+ * The silencethreshold setting that was previously configurable in multiple
+ applications is now settable globally via dsp.conf.
+
+Music On Hold Changes
+---------------------
+ * A new option, "digit", has been added for music on hold classes in
+ musiconhold.conf. If this is set for a music on hold class, a caller
+ listening to music on hold can press this digit to switch to listening
+ to this music on hold class.
+ * Support for realtime music on hold has been added.
+ * In conjunction with the realtime music on hold, a general section has
+ been added to musiconhold.conf, its sole variable is cachertclasses. If this
+ is set, then music on hold classes found in realtime will be cached in memory.
+
+AEL Changes
+-----------
+ * AEL upgraded to use the Gosub with Arguments instead
+ of Macro application, to hopefully reduce the problems
+ seen with the artificially low stack ceiling that
+ Macro bumps into. Macros can only call other Macros
+ to a depth of 7. Tests run using gosub, show depths
+ limited only by virtual memory. A small test demonstrated
+ recursive call depths of 100,000 without problems.
+ -- in addition to this, all apps that allowed a macro
+ to be called, as in Dial, queues, etc, are now allowing
+ a gosub call in similar fashion.
+ * AEL now generates LOCAL(argname) declarations when it
+ Set()'s the each arg name to the value of ${ARG1}, ${ARG2),
+ etc. That makes the arguments local in scope. The user
+ can define their own local variables in macros, now,
+ by saying "local myvar=someval;" or using Set() in this
+ fashion: Set(LOCAL(myvar)=someval); ("local" is now
+ an AEL keyword).
+ * utils/conf2ael introduced. Will convert an extensions.conf
+ file into extensions.ael. Very crude and unfinished, but
+ will be improved as time goes by. Should be useful for a
+ first pass at conversion.
+ * aelparse will now read extensions.conf to see if a referenced
+ macro or context is there before issuing a warning.
+ * AEL parser sets a local channel variable ~~EXTEN~~, to
+ preserve the value of ${EXTEN} thru switch statements.
+ * New operator in $[...] expressions: the ~~ operator serves
+ as a concatenation operator. AT THE MOMENT, it is really only
+ necessary and useful in AEL, especially in if() expressions.
+ Operation: ${a} ~~ ${b| with force both a and b to strings, strip
+ any enclosing double-quotes, and evaluate to the value of a
+ concatenated with the value of b. For example if a is set to
+ "xyz" and b has the value "abc", then ${a} ~~ ${b| would
+ evaluate to xyzabc .
+
+
+Call Features (res_features) Changes
+------------------------------------
+ * Added the parkedcalltransfers option to features.conf
+ * Added parkedcallparking option to control one touch parking w/ parking
+ pickup
+ * Added parkedcallhangup option to control disconnect feature w/ parking
+ pickup
+ * Added parkedcallrecording option to control one-touch record w/ parking
+ pickup
+ * Added parkedcallparking, parkedcallhangup, parkedcallrecording, and
+ parkedcalltransfers option support for multiple parking lots.
+ * Added BRIDGE_FEATURES variable to set available features for a channel
+ * The built-in method for doing attended transfers has been updated to
+ include some new options that allow you to have the transferee sent
+ back to the person that did the transfer if the transfer is not successful.
+ See the options "atxferdropcall", "atxferloopdelay", and "atxfercallbackretries"
+ in features.conf.sample.
+ * Added support for configuring named groups of custom call features in
+ features.conf. This means that features can be written a single time, and
+ then mapped into groups of features for different key mappings or easier
+ access control.
+ * Updated the ParkedCall application to allow you to not specify a parking
+ extension. If you don't specify a parking space to pick up, it will grab
+ the first one available.
+ * Added cli command 'features reload' to reload call features from features.conf
+ * Moved into core asterisk binary.
+ * Changed the default setting for featuredigittimeout to 2000 ms from 500 ms.
+ * Added the ability for custom parking lots to be configured with their own
+ parking extension with the parkext option.
+
+Language Support Changes
+------------------------
+ * Brazilian Portuguese (pt-BR) in VM, and say.c was added
+ * Added support for the Hungarian language for saying numbers, dates, and times.
+
+AGI Changes
+-----------
+ * Added SPEECH commands for speech recognition. A complete listing can be found
+ using agi show.
+ * If app_stack is loaded, GOSUB is a native AGI command that may be used to
+ invoke subroutines in the dialplan. Note that calling EXEC with Gosub
+ does not behave as expected; the native command needs to be used, instead.
+ * Added the ability to perform SRV lookups on fast AGI calls. To use this
+ feature, simply use hagi: instead of agi: as the protocol portion
+ of the URI parameter to the AGI function call in your dial plan. Also note
+ that specifying a port number in the AGI URI will disable SRV lookups,
+ even if you use the hagi: protocol.
+ * No longer support MSG_OOB flag on HANGUP.
+
+Logger changes
+--------------
+ * Added rotatestrategy option to logger.conf, along with two new options:
+ "timestamp" which will use the time to name the logger files instead of
+ sequence number; and "rotate", which rotates the names of the log files,
+ similar to the way syslog rotates files.
+ * Added exec_after_rotate option to logger.conf, which allows a system
+ command to be run after rotation. This is primarily useful with
+ rotatestrategy=rotate, to allow a limit on the number of log files kept
+ and to ensure that the oldest log file gets deleted.
+ * Added realtime support for the queue log
+
+Call Detail Records
+-------------------
+ * The cdr_manager module has a [mappings] feature, like cdr_custom,
+ to add fields to the manager event from the CDR variables.
+ * Added cdr_adaptive_odbc, a new module that adapts to the structure of your
+ backend database CDR table. Specifically, additional, non-standard
+ columns are supported, merely by setting the corresponding CDR variable in
+ your dialplan. In addition, you may alias any column to another name (for
+ example, if you want the 'src' CDR variable to be column 'ANI' in the DB,
+ simply "alias src => ANI" in the configuration file). Records may be
+ posted to more than one backend, simply by specifying multiple categories
+ in the configuration file. And finally, you may filter which CDRs get
+ posted to each backend, by specifying a filter (which the record must
+ match) for the particular category. Filters are additive (meaning all
+ rules must match to post that CDR).
+ * The Postgres CDR module now supports some features of the cdr_adaptive_odbc
+ module. Specifically, you may add additional columns into the table and
+ they will be set, if you set the corresponding CDR variable name. Also,
+ if you omit columns in your database table, they will be silently skipped
+ (but a record will still be inserted, based on what columns remain). Note
+ that the other two features from cdr_adaptive_odbc (alias and filter) are
+ not currently supported.
+ * The ResetCDR application now has an 'e' option that re-enables a CDR if it
+ has been disabled using the NoCDR application.
+
+Miscellaneous New Modules
+-------------------------
+ * Added a new CDR module, cdr_sqlite3_custom.
+ * Added a new realtime configuration module, res_config_sqlite
+ * Added a new codec translation module, codec_resample, which re-samples
+ signed linear audio between 8 kHz and 16 kHz to help support wideband
+ codecs.
+ * Added a new module, res_phoneprov, which allows auto-provisioning of phones
+ based on configuration templates that use Asterisk dialplan function and
+ variable substitution. It should be possible to create phone profiles and
+ templates that work for the majority of phones provisioned over http. It
+ is currently only intended to provision a single user account per phone.
+ An example profile and set of templates for Polycom phones is provided.
+ NOTE: Polycom firmware is not included, but should be placed in
+ AST_DATA_DIR/phoneprov/configs to match up with the included templates.
+ * Added a new module, app_jack, which provides interfaces to JACK, the Jack
+ Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
+ provided; there is a JACK() application, and a JACK_HOOK() function. Both
+ interfaces create an input and output JACK port. The application makes
+ these ports the endpoint of the call. The audio coming from the channel
+ goes out the output port and whatever comes back in on the input port is
+ what gets sent to the channel. The JACK_HOOK() function turns on a JACK
+ audiohook on the channel. This lets you run the audio coming from a
+ channel through JACK, and whatever comes back in is what gets forwarded
+ on as the channel's audio. This is very useful for building custom
+ vocoders or doing recording or analysis of the channel's audio in another
+ application.
+ * Added a new module, res_config_curl, which permits using a HTTP POST url
+ to retrieve, create, update, and delete realtime information from a remote
+ web server. Note that this module requires func_curl.so to be loaded for
+ backend functionality.
+ * Added a new module, res_config_ldap, which permits the use of an LDAP
+ server for realtime data access.
+ * Added support for writing and running your dialplan in lua using the pbx_lua
+ module. See configs/extensions.lua.sample for examples of how to do this.
+
+Miscellaneous
+-------------
+ * Ability to use libcap to set high ToS bits when non-root
+ on Linux. If configure is unable to find libcap then you
+ can use --with-cap to specify the path.
+ * Added maxfiles option to options section of asterisk.conf which allows you to specify
+ what Asterisk should set as the maximum number of open files when it loads.
+ * Added the jittertargetextra configuration option.
+ * Added support for setting the CoS for VLAN traffic (802.1p). See the sample
+ configuration files for the IP channel drivers. The new option is "cos".
+ This information is also documented on the Asterisk wiki at
+ https://wiki.asterisk.org/wiki/x/EYBG
+ * When originating a call using AMI or pbx_spool that fails the reason for failure
+ will now be available in the failed extension using the REASON dialplan variable.
+ * Added support for reading the TOUCH_MONITOR_PREFIX channel variable.
+ It allows you to configure a prefix for auto-monitor recordings.
+ * A new extension pattern matching algorithm, based on a trie, is introduced
+ here, that could noticeably speed up mid-sized to large dialplans.
+ It is NOT used by default, as duplicating the behaviour of the old pattern
+ matcher is still under development. A config file option, in extensions.conf,
+ in the [general] section, called "extenpatternmatchingnew", is by default
+ set to false; setting that to true will force the use of the new algorithm.
+ Also, the cli commands "dialplan set extenpatternmatchingnew true/false" can
+ be used to switch the algorithms at run time.
+ * A new option when starting a remote asterisk (rasterisk, asterisk -r) for
+ specifying which socket to use to connect to the running Asterisk daemon
+ (-s)
+ * Performance enhancements to the sched facility, which is used in
+ the channel drivers, etc. Added hashtabs and doubly-linked lists
+ to speed up deletion; start at the beginning or end of list to
+ speed up insertion.
+ * Added Doubly-linked lists after the fashion of linkedlists.h. They are in
+ dlinkedlists.h. Doubly-linked lists feature fast deletion times.
+ Added regression tests to the tests/ dir, also.
+ * Added a refcount trace feature to astobj2 for those trying to balance
+ object creation, deletion; work, play; space and time. See the
+ notes in astobj2.h. Also, see utils/refcounter as well, as a
+ quick way to find unbalanced refcounts in what could be a sea
+ of objects that were balanced.
+ * Added logging to 'make update' command. See update.log
+ * Added strictrtp option to rtp.conf. If enabled this will drop RTP packets that
+ do not come from the remote party.
+ * Added the 'n' option to the SpeechBackground application to tell it to not
+ answer the channel if it has not already been answered.
+ * Added a compiler flag, CHANNEL_TRACE, which permits channel tracing to be
+ turned on, via the CHANNEL(trace) dialplan function. Could be useful for
+ dialplan debugging.
+ * iLBC source code no longer included (see UPGRADE.txt for details)
+ * If compiled with DETECT_DEADLOCKS enabled and if you have glibc, then if
+ deadlock is detected, a backtrace of the stack which led to the lock calls
+ will be output to the CLI.
+ * If compiled with DEBUG_THREADS enabled and if you have glibc, then issuing
+ the "core show locks" CLI command will give lock information output as well
+ as a backtrace of the stack which led to the lock calls.
+ * users.conf now sports an optional alternateexts property, which permits
+ allocation of additional extensions which will reach the specified user.
+ * A new option for the configure script, --enable-internal-poll, has been added
+ for use with systems which may have a buggy implementation of the poll system
+ call. If you notice odd behavior such as the CLI being unresponsive on remote
+ consoles, you may want to try using this option. This option is enabled by default
+ on Darwin systems since it is known that the Darwin poll() implementation has
+ odd issues.
+
+Timer Changes
+--------------------
+* In addition to timing from DAHDI, there is a new timing module called
+ res_timing_timerfd. In order to use this, you must be running Linux with
+ a kernel version 2.6.25 or newer as well as glibc 2.8 or newer. The configure
+ script will be able to tell if you have the requirements. From menuselect, select
+ res_timing_timerfd from the Resource Modules menu.
projects / thirdparty / asterisk.git / commitdiff
