From a818b05ca18a97e6f61a8257d24a61d0fe2d65ee Mon Sep 17 00:00:00 2001 From: Asterisk Development Team Date: Wed, 20 Jul 2022 05:44:50 -0500 Subject: [PATCH] Update CHANGES and UPGRADE.txt for 20.0.0 --- CHANGES | 494 ++++++++++++++++++ UPGRADE.txt | 67 +++ .../add_mix_option_to_playback.txt | 7 - doc/CHANGES-staging/allow_wildcard_certs.txt | 9 - doc/CHANGES-staging/ami_wink.txt | 3 - .../app_confbridge_channels.txt | 7 - .../app_confbridge_hear_join.txt | 8 - doc/CHANGES-staging/app_dtmfstore.txt | 6 - doc/CHANGES-staging/app_mf_maxdigits.txt | 5 - doc/CHANGES-staging/app_mf_mf.txt | 5 - doc/CHANGES-staging/app_milliwatt.txt | 11 - doc/CHANGES-staging/app_morsecode.txt | 6 - doc/CHANGES-staging/app_originate_codecs.txt | 6 - doc/CHANGES-staging/app_queue.txt | 4 - .../app_queue_DIALEDPEERNUMBER.txt | 6 - doc/CHANGES-staging/app_queue_logintime.txt | 9 - doc/CHANGES-staging/app_queue_music.txt | 5 - .../app_queue_nordic_language.txt | 3 - doc/CHANGES-staging/app_queue_say_thanks.txt | 4 - doc/CHANGES-staging/app_queue_stats.txt | 7 - doc/CHANGES-staging/app_read.txt | 5 - doc/CHANGES-staging/app_sendtext.txt | 4 - doc/CHANGES-staging/app_voicemail.txt | 7 - .../app_voicemail_danish_syntax.txt | 6 - .../app_voicemail_nodelete.txt | 5 - .../ari_add_pvt_id_to_channel_resource.txt | 7 - doc/CHANGES-staging/ast_coredumper.txt | 23 - .../bundled-pjproject-build.txt | 8 - doc/CHANGES-staging/cdr_disable.txt | 8 - doc/CHANGES-staging/chan_dahdi_cadences.txt | 8 - doc/CHANGES-staging/chan_dahdi_polarity.txt | 5 - doc/CHANGES-staging/chan_iax2_ani2.txt | 4 - doc/CHANGES-staging/chan_iax2_dial.txt | 4 - doc/CHANGES-staging/chan_pjsip_180_sdp.txt | 8 - doc/CHANGES-staging/chan_pjsip_flash.txt | 4 - .../chan_sip_pickup_AST_STATE_DOWN.txt | 3 - .../chan_sip_session-timer_on_update.txt | 6 - doc/CHANGES-staging/channel_internal_api.txt | 4 - doc/CHANGES-staging/cli_eval_function.txt | 5 - doc/CHANGES-staging/cli_refresh.txt | 5 - doc/CHANGES-staging/func_channel.txt | 4 - doc/CHANGES-staging/func_db.txt | 6 - doc/CHANGES-staging/func_env.txt | 5 - doc/CHANGES-staging/func_evalexten.txt | 4 - doc/CHANGES-staging/func_framedrop.txt | 5 - doc/CHANGES-staging/func_json.txt | 5 - .../func_odbc_esc_backslashes.txt | 7 - doc/CHANGES-staging/func_scramble.txt | 5 - doc/CHANGES-staging/func_strings.txt | 7 - doc/CHANGES-staging/func_vmcount.txt | 3 - doc/CHANGES-staging/load_realtime_queues.txt | 5 - doc/CHANGES-staging/logger.txt | 5 - doc/CHANGES-staging/manager_disable.txt | 4 - doc/CHANGES-staging/manager_message_send.txt | 6 - doc/CHANGES-staging/mf.txt | 6 - doc/CHANGES-staging/pjsip_read_headers.txt | 5 - doc/CHANGES-staging/queue_withdraw_caller.txt | 14 - doc/CHANGES-staging/res_agi.txt | 5 - doc/CHANGES-staging/res_cliexec.txt | 6 - doc/CHANGES-staging/res_fax_spandsp.txt | 3 - doc/CHANGES-staging/res_geolocation.txt | 4 - doc/CHANGES-staging/res_parking_moh.txt | 4 - doc/CHANGES-staging/res_pjproject.txt | 8 - doc/CHANGES-staging/res_pjsip_geolocation.txt | 4 - .../res_pjsip_header_funcs.txt | 5 - doc/CHANGES-staging/res_pjsip_registrar.txt | 7 - .../res_pjsip_t38_bind_fixes.txt | 9 - ...asterisk_stunaddr_recurring_resolution.txt | 6 - doc/CHANGES-staging/res_statsd.txt | 5 - doc/CHANGES-staging/res_tonedetect.txt | 5 - doc/CHANGES-staging/rls_display_name.txt | 10 - doc/CHANGES-staging/rls_refresh.txt | 7 - doc/CHANGES-staging/say.txt | 7 - doc/CHANGES-staging/tonescan.txt | 6 - doc/UPGRADE-staging/chan_iax2_rsa.txt | 15 - .../http-media-cache-lookup-order.txt | 9 - .../manager_amxml_attribute_fix.txt | 8 - doc/UPGRADE-staging/res_monitor_disabled.txt | 8 - .../res_pjsip_async_operations.txt | 7 - .../stir_shaken_option_split.txt | 7 - 80 files changed, 561 insertions(+), 491 deletions(-) delete mode 100644 doc/CHANGES-staging/add_mix_option_to_playback.txt delete mode 100644 doc/CHANGES-staging/allow_wildcard_certs.txt delete mode 100644 doc/CHANGES-staging/ami_wink.txt delete mode 100644 doc/CHANGES-staging/app_confbridge_channels.txt delete mode 100644 doc/CHANGES-staging/app_confbridge_hear_join.txt delete mode 100644 doc/CHANGES-staging/app_dtmfstore.txt delete mode 100644 doc/CHANGES-staging/app_mf_maxdigits.txt delete mode 100644 doc/CHANGES-staging/app_mf_mf.txt delete mode 100644 doc/CHANGES-staging/app_milliwatt.txt delete mode 100644 doc/CHANGES-staging/app_morsecode.txt delete mode 100644 doc/CHANGES-staging/app_originate_codecs.txt delete mode 100644 doc/CHANGES-staging/app_queue.txt delete mode 100644 doc/CHANGES-staging/app_queue_DIALEDPEERNUMBER.txt delete mode 100644 doc/CHANGES-staging/app_queue_logintime.txt delete mode 100644 doc/CHANGES-staging/app_queue_music.txt delete mode 100644 doc/CHANGES-staging/app_queue_nordic_language.txt delete mode 100644 doc/CHANGES-staging/app_queue_say_thanks.txt delete mode 100644 doc/CHANGES-staging/app_queue_stats.txt delete mode 100644 doc/CHANGES-staging/app_read.txt delete mode 100644 doc/CHANGES-staging/app_sendtext.txt delete mode 100644 doc/CHANGES-staging/app_voicemail.txt delete mode 100644 doc/CHANGES-staging/app_voicemail_danish_syntax.txt delete mode 100644 doc/CHANGES-staging/app_voicemail_nodelete.txt delete mode 100644 doc/CHANGES-staging/ari_add_pvt_id_to_channel_resource.txt delete mode 100644 doc/CHANGES-staging/ast_coredumper.txt delete mode 100644 doc/CHANGES-staging/bundled-pjproject-build.txt delete mode 100644 doc/CHANGES-staging/cdr_disable.txt delete mode 100644 doc/CHANGES-staging/chan_dahdi_cadences.txt delete mode 100644 doc/CHANGES-staging/chan_dahdi_polarity.txt delete mode 100644 doc/CHANGES-staging/chan_iax2_ani2.txt delete mode 100644 doc/CHANGES-staging/chan_iax2_dial.txt delete mode 100644 doc/CHANGES-staging/chan_pjsip_180_sdp.txt delete mode 100644 doc/CHANGES-staging/chan_pjsip_flash.txt delete mode 100644 doc/CHANGES-staging/chan_sip_pickup_AST_STATE_DOWN.txt delete mode 100644 doc/CHANGES-staging/chan_sip_session-timer_on_update.txt delete mode 100644 doc/CHANGES-staging/channel_internal_api.txt delete mode 100644 doc/CHANGES-staging/cli_eval_function.txt delete mode 100644 doc/CHANGES-staging/cli_refresh.txt delete mode 100644 doc/CHANGES-staging/func_channel.txt delete mode 100644 doc/CHANGES-staging/func_db.txt delete mode 100644 doc/CHANGES-staging/func_env.txt delete mode 100644 doc/CHANGES-staging/func_evalexten.txt delete mode 100644 doc/CHANGES-staging/func_framedrop.txt delete mode 100644 doc/CHANGES-staging/func_json.txt delete mode 100644 doc/CHANGES-staging/func_odbc_esc_backslashes.txt delete mode 100644 doc/CHANGES-staging/func_scramble.txt delete mode 100644 doc/CHANGES-staging/func_strings.txt delete mode 100644 doc/CHANGES-staging/func_vmcount.txt delete mode 100644 doc/CHANGES-staging/load_realtime_queues.txt delete mode 100644 doc/CHANGES-staging/logger.txt delete mode 100644 doc/CHANGES-staging/manager_disable.txt delete mode 100644 doc/CHANGES-staging/manager_message_send.txt delete mode 100644 doc/CHANGES-staging/mf.txt delete mode 100644 doc/CHANGES-staging/pjsip_read_headers.txt delete mode 100644 doc/CHANGES-staging/queue_withdraw_caller.txt delete mode 100644 doc/CHANGES-staging/res_agi.txt delete mode 100644 doc/CHANGES-staging/res_cliexec.txt delete mode 100644 doc/CHANGES-staging/res_fax_spandsp.txt delete mode 100644 doc/CHANGES-staging/res_geolocation.txt delete mode 100644 doc/CHANGES-staging/res_parking_moh.txt delete mode 100644 doc/CHANGES-staging/res_pjproject.txt delete mode 100644 doc/CHANGES-staging/res_pjsip_geolocation.txt delete mode 100644 doc/CHANGES-staging/res_pjsip_header_funcs.txt delete mode 100644 doc/CHANGES-staging/res_pjsip_registrar.txt delete mode 100644 doc/CHANGES-staging/res_pjsip_t38_bind_fixes.txt delete mode 100644 doc/CHANGES-staging/res_rtp_asterisk_stunaddr_recurring_resolution.txt delete mode 100644 doc/CHANGES-staging/res_statsd.txt delete mode 100644 doc/CHANGES-staging/res_tonedetect.txt delete mode 100644 doc/CHANGES-staging/rls_display_name.txt delete mode 100644 doc/CHANGES-staging/rls_refresh.txt delete mode 100644 doc/CHANGES-staging/say.txt delete mode 100644 doc/CHANGES-staging/tonescan.txt delete mode 100644 doc/UPGRADE-staging/chan_iax2_rsa.txt delete mode 100644 doc/UPGRADE-staging/http-media-cache-lookup-order.txt delete mode 100644 doc/UPGRADE-staging/manager_amxml_attribute_fix.txt delete mode 100644 doc/UPGRADE-staging/res_monitor_disabled.txt delete mode 100644 doc/UPGRADE-staging/res_pjsip_async_operations.txt delete mode 100644 doc/UPGRADE-staging/stir_shaken_option_split.txt diff --git a/CHANGES b/CHANGES index e65ce50bd3..c10b54e61d 100644 --- a/CHANGES +++ b/CHANGES @@ -12,6 +12,500 @@ === ============================================================================== +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 19.0.0 to Asterisk 20.0.0 ------------ +------------------------------------------------------------------------------ + +Applications +------------------ + * added support for Danish syntax, playing the correct plural sound file + dependen on where you have 1 or multipe messages + based on the existing SE/NO code + + * added that we set DIALEDPEERNUMBER on the outgoing channels + so it is avalible in b(content^extension^line) + this add the same behaviour as Dial + +Channel-agnostic MF support +------------------ + * A SendMF application and PlayMF manager + application are now included to send + arbitrary standard R1 MF tones on the + current channel or another specified channel. + +Core +------------------ + * Bundled PJProject Build + + The build process has been updated to make pjproject troubleshooting + and development easier. See third-party/pjproject/README-hacking.md or + https://wiki.asterisk.org/wiki/display/AST/Bundled+PJProject + for more info. + +Handle non-standard Meter metric type safely +------------------ + * A meter_support flag has been introduced that defaults to true to maintain current behaviour. + If disabled, a counter metric type will be used instead wherever a meter metric type was used, + the counter will have a "_meter" suffix appended to the metric name. + +MessageSend +------------------ + * The MessageSend AMI action has been updated to allow the Destination + and the To addresses to be provided separately. This brings the + MessageSend manager command in line with the capabilities of the + MessageSend dialplan application. + +ToneScan application +------------------ + * A new application, ToneScan, allows for + synchronous detection of call progress + signals such as dial tone, busy tone, + Special Information Tones, and modems. + +ami +------------------ + * An AMI event now exists for "Wink". + + * AMI events can now be globally disabled using + the disabledevents [general] setting. + +app_confbridge +------------------ + * Added the hear_own_join_sound option to the confbridge user profile to + control who hears the sound_join audio file. When set to 'yes' the user + entering the conference and the participants already in the conference + will hear the sound_join audio file. When set to 'no' the user entering + the conference will not hear the sound_join audio file, but the + participants already in the conference will hear the sound_join audio file. + + * Adds the CONFBRIDGE_CHANNELS function which can + be used to retrieve a list of channels in a ConfBridge, + optionally filtered by a particular category. This + list can then be used with functions like SHIFT, POP, + UNSHIFT, etc. + +app_dtmfstore +------------------ + * New application which collects digits + dialed and stores them into + a specified variable. + +app_mf +------------------ + * Adds MF receiver and sender applications to support + the R1 MF signaling protocol, including integration + with the Dial application. + + * Adds an option to ReceiveMF to cap the + number of digits read at a user-specified + maximum. + +app_milliwatt +------------------ + * The Milliwatt application's existing behavior is + incorrect in that it plays a constant tone, which + is not how digital milliwatt test lines actually + work. + + An option is added so that a proper milliwatt test + tone can be provided, including a 1 second silent + interval every 10 seconds. However, for compatability + reasons, the default behavior remains unchanged. + +app_morsecode +------------------ + * Extends the Morsecode application by adding support for + American Morse code and adds a configurable option + for the frequency used in off intervals. + +app_originate +------------------ + * Codecs can now be specified for dialplan-originated + calls, as with call files and the manager action. + By default, only the slin codec is now used, instead + of all the slin* codecs. + +app_playback +------------------ + * A new option 'mix' is added to the Playback application that + will play by filename and say.conf. It will look on the format of the + name, if it is like say format it will play with say.conf if not it + will play the file name. + +app_queue +------------------ + * Reload behavior in app_queue has been changed so + queue and agent stats are not reset during full + app_queue module reloads. The queue reset stats + CLI command may still be used to reset stats while + Asterisk is running. + + * Add field to save the time value when a member enter a queue. + Shows this time in seconds using 'queue show' command and the + field LoginTime for responses for AMI the events. + + The output for the CLI command `queue show` is changed by added a + extra data field for the information of the time login time for each + member. + + * added that we set DIALEDPEERNUMBER on the outgoing channels + so it is avalible in b(content^extension^line) + this add the same behaviour as Dial + + * Load queues and members from Realtime for + AMI actions: QueuePause, QueueStatus and QueueSummary, + Applications: PauseQueueMember and UnpauseQueueMember. + + * Added a new AMI action: QueueWithdrawCaller + This AMI action makes it possible to withdraw a caller from a queue + back to the dialplan. The call will be signaled to leave the queue + whenever it can, hence, it not guaranteed that the call will leave + the queue. + + Optional custom data can be passed in the request, in the WithdrawInfo + parameter. If the call successfully withdrawn the queue, + it can be retrieved using the QUEUE_WITHDRAW_INFO variable. + + This can be useful for certain uses, such as dispatching the call + to a specific extension. + + * The m option now allows an override music on hold + class to be specified for the Queue application + within the dialplan. + +app_queue.c +------------------ + * Allow multiple files to be streamed for agent announcement. + +app_queues +------------------ + * adding support for playing the correct en/et for nordic languages + + * Don't play sound_thanks if there is no leading hold_time message + When the only announcement is hold time, and there is no hold time (0 min, 0 sec), asterisk will say "thank you for your patience" + +app_read +------------------ + * A new option allows the digit '#' to be read literally, + rather than used exclusively as the input terminator + character. + +app_sendtext +------------------ + * A ReceiveText application has been added that can be + used in conjunction with the SendText application. + +app_voicemail +------------------ + * Add a new 'S' option to VoiceMail which prevents the instructions + (vm-intro) from being played if a busy/unavailable/temporary greeting + from the voicemail user is played. This is similar to the existing 's' + option except that instructions will still be played if no user + greeting is available. + + * added support for Danish syntax, playing the correct plural sound file + dependen on where you have 1 or multipe messages + based on the existing SE/NO code + + * The r option has been added, which prevents deletion + of messages from VoiceMailMain, which can be + useful for shared mailboxes. + +apps +------------------ + * A new option 'mix' is added to the Playback application that + will play by filename and say.conf. It will look on the format of the + name, if it is like say format it will play with say.conf if not it + will play the file name. + +ari +------------------ + * Expose channel driver's unique id (which is the Call-ID for SIP/PJSIP) + to ARI channel resources as 'protocol_id'. + + ASTERISK-30027 + +ast_coredumper +------------------ + * New options: + --pid= + Allows specification of an Asterisk instance when trying to + and the script can't determine it itself. + --libdir= + Allows specification of a non-standard installation directory + containing the Asterisk modules. + --(no-)rename + Renames the coredump and the output files with readable + timestamps. This is the default. + Removed unneeded or confusing options: + --append-coredumps + --conffile + --no-default-search + --tarball-uniqueid + Changed Variables: + COREDUMPS is now just "/tmp/core!(*.txt)" + DATEFORMAT is renamed to DATEOPTS and defaults to '-u +%FT%H-%M-%SZ' + Changed behavior: + If you use 'running' or 'RUNNING' you no longer need to specify + '--no-default-search' to ignore existing coredumps. + +cdr +------------------ + * A new CDR option, channeldefaultenabled, allows controlling + whether CDR is enabled or disabled by default on + newly created channels. The default behavior remains + unchanged from previous versions of Asterisk (new + channels will have CDR enabled, as long as CDR is + enabled globally). + +chan_dahdi +------------------ + * Previously, cadences were appended on dahdi restart, + rather than reloaded. This prevented cadences from + being updated and maxed out the available cadences + if reloaded multiple times. This behavior is fixed + so that reloading cadences is idempotent and cadences + can actually be reloaded. + + * A POLARITY function is now available that allows + getting or setting the polarity on a channel + from the dialplan. + +chan_iax2 +------------------ + * ANI2 (OLI) is now transmitted over IAX2 calls + as an information element. + + * Both a secret and an outkey may be specified at dial time, + since encryption is possible with RSA authentication. + +chan_pjsip +------------------ + * Add function PJSIP_HEADERS() to get list of headers by pattern in the same way as SIP_HEADERS() do. + + Add ability to read header by pattern using PJSIP_HEADER(). + + * added global config option "allow_sending_180_after_183" + + Allow Asterisk to send 180 Ringing to an endpoint + after 183 Session Progress has been send. + If disabled Asterisk will instead send only a + 183 Session Progress to the endpoint. + + * Hook flash events can now be sent on a PJSIP channel + if requested to do so. + +chan_sip +------------------ + * Session timers get removed on UPDATE + Fix if Asterisk receives a SIP REFER with Session-Timers UAC + that Asterisk maintains Session-Timers when sending UPDATE request + +chan_sip.c +------------------ + * resolve issue with pickup on device that uses "183" and not "180" + +channel_internal_api +------------------ + * CHANNEL(lastcontext) and CHANNEL(lastexten) + are now available for use in the dialplan. + +cli +------------------ + * The "module refresh" command has been added, + which allows unloading and then loading a + module with a single command. + + * A new CLI command 'dialplan eval function' has been + added which allows users to test the behavior of + dialplan function calls directly from the CLI. + +func_channel +------------------ + * Adds the CHANNEL_EXISTS function to check for the existence + of a channel by name or unique ID. + +func_db +------------------ + * The function DB_KEYCOUNT has been added, which + returns the cardinality of the keys at a specified + prefix in AstDB, i.e. the number of keys at a + given prefix. + +func_env.c +------------------ + * Two new functions, DIRNAME and BASENAME, are now + included which allow users to obtain the directory + or the base filename of any file. + +func_evalexten +------------------ + * This adds the EVAL_EXTEN function which may be + used to evaluate data at dialplan extensions. + +func_framedrop +------------------ + * New function to selectively drop specified frames + in either direction on a channel. + +func_json +------------------ + * The JSON_DECODE dialplan function can now be used + to parse JSON strings, such as in conjunction with + CURL for using API responses. + +func_odbc +------------------ + * A SQL_ESC_BACKSLASHES dialplan function has been added which + escapes backslashes. Usage of this is dependent on whether the + database in use can use backslashes to escape ticks or not. If + it can, then usage of this prevents a broken SQL query depending + on how the SQL query is constructed. + +func_scramble +------------------ + * Adds an audio scrambler function that may be used to + distort voice audio on a channel as a privacy + enhancement. + +func_strings +------------------ + * A new STRBETWEEN function is now included which + allows a substring to be inserted between characters + in a string. This is particularly useful for transforming + dial strings, such as adding pauses between digits + for a string of digits that are sent to another channel. + +func_vmcount +------------------ + * Multiple mailboxes may now be specified instead of just one. + +logger +------------------ + * Added the ability to define custom log levels in logger.conf + and use them in the Log dialplan application. Also adds a + logger show levels CLI command. + +res_agi +------------------ + * Agi command 'exec' can now be enabled + to evaluate dialplan functions and variables + by setting the variable AGIEXECFULL to yes. + +res_cliexec +------------------ + * A new CLI command, dialplan exec application, has + been added which allows dialplan applications to be + executed at the CLI, useful for some quick testing + without needing to write dialplan. + +res_fax_spandsp +------------------ + * Adds support for spandsp 3.0.0. + +res_geolocation +------------------ + * Added res_geolocation which creates the core capabilities + to manipulate Geolocation information on SIP INVITEs. + +res_parking +------------------ + * An m option to Park and ParkAndAnnounce now allows + specifying a music on hold class override. + +res_pjproject +------------------ + * In pjproject.conf you can now map pjproject log levels + to the Asterisk TRACE log level. The default mappings + have therefore changed so that only pjproject levels + 3 and 4 are mapped to DEBUG and 5 and 6 are now mapped + to TRACE. Previously 3, 4, 5, and 6 were all mapped to + DEBUG. + +res_pjsip +------------------ + * A new transport option 'allow_wildcard_certs' has been added that when it + and 'verify_server' are both set to 'yes', enables verification against + wildcards, i.e. '*.' in certs for common, and subject alt names of type DNS + for TLS transport types. Names must start with the wildcard. Partial wildcards, + e.g. 'f*.example.com' and 'foo.*.com' are not allowed. As well, names only + match against a single level meaning '*.example.com' matches 'foo.example.com', + but not 'foo.bar.example.com'. + +res_pjsip_geolocation +------------------ + * Added res_pjsip_geolocation which gives chan_pjsip + the ability to use the core geolocation capabilities. + +res_pjsip_header_funcs +------------------ + * Add function PJSIP_RESPONSE_HEADERS() to get list of header names from 200 response, in the same way as PJSIP_HEADERS() from the request. + + Add function PJSIP_RESPONSE_HEADER() to read header from 200 response, in the same way as PJSIP_HEADER() from the request. + +res_pjsip_pubsub +------------------ + * A new resource_list option, resource_display_name, indicates + whether display name of resource or the resource name being + provided for RLS entries. + If this option is enabled, the Display Name will be provided. + This option is disabled by default to remain the previous behavior. + If the 'event' set to 'presence' or 'dialog' the non-empty HINT name + will be set as the Display Name. + The 'message-summary' is not supported yet. + + * The Resource List Subscriptions (RLS) is dynamic now. + The asterisk now updates current subscriptions to reflect the changes + to the list on subscription refresh. If list items are added, + removed, updated or do not exist anymore, the asterisk regenerates + the resource list. + +res_pjsip_registrar +------------------ + * Adds new PJSIP AOR option remove_unavailable to either + remove unavailable contacts when a REGISTER exceeds + max_contacts when remove_existing is disabled, or + prioritize unavailable contacts over other existing + contacts when remove_existing is enabled. + +res_pjsip_t38 +------------------ + * In res_pjsip_sdp_rtp, the bind_rtp_to_media_address option and the + fallback use of the transport's bind address solve problems sending + media on systems that cannot send ipv4 packets on ipv6 sockets, and + certain other situations. This change extends both of these behaviors + to UDPTL sessions as well in res_pjsip_t38, to fix fax-specific + problems on these systems, introducing a new option + endpoint/t38_bind_udptl_to_media_address. + +res_rtp_asterisk +------------------ + * When the address of the STUN server (stunaddr) is a name resolved via DNS, the + stunaddr will be recurringly resolved when the DNS answer Time-To-Live (TTL) + expires. This allows the STUN server to change its IP address without having to + reload the res_rtp_asterisk module. + +res_tonedetect +------------------ + * Arbitrary tone detection is now available through a + WaitForTone application (blocking) and a TONE_DETECT + function (non-blocking). + +say.c +------------------ + * Adds SAYFILES function to retrieve the file names that would + be played by corresponding Say applications, such as + SayDigits, SayAlpha, etc. + + Additionally adds SayMoney and SayOrdinal applications. + +stasis_channels +------------------ + * Expose channel driver's unique id (which is the Call-ID for SIP/PJSIP) + to ARI channel resources as 'protocol_id'. + + ASTERISK-30027 + ------------------------------------------------------------------------------ --- Functionality changes from Asterisk 18.0.0 to Asterisk 19.0.0 ------------ ------------------------------------------------------------------------------ diff --git a/UPGRADE.txt b/UPGRADE.txt index f76559ec79..9a5d1caeb0 100644 --- a/UPGRADE.txt +++ b/UPGRADE.txt @@ -18,6 +18,73 @@ === =========================================================== +------------------------------------------------------------------------------ +--- New functionality introduced in Asterisk 20.0.0 -------------------------- +------------------------------------------------------------------------------ + +res_monitor +------------------ + * This module is no longer built by default in + accordance with the Module Deprecation Policy. + If you require this functionality you will need + to enable it for building in menuselect. Note + that in the future res_monitor will be removed. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 19.0.0 to Asterisk 20.0.0 ------------ +------------------------------------------------------------------------------ + +AMI +------------------ + * The XML Manager Event Interface (amxml) now generates attribute names + that are compliant with the XML 1.1 specification. Previously, an + attribute name that started with a digit would be rendered as-is, even + though attribute names must not begin with a digit. We now prefix + attribute names that start with a digit with an underscore ('_') to + prevent XML validation failures. + +STIR/SHAKEN +------------------ + * The STIR/SHAKEN configuration option has been split into + 4 different choices: off, attest, verify, and on. Off and + on behave the same way as before. Attest will only perform + attestation on the endpoint, and verify will only perform + verification on the endpoint. + +chan_iax2 +------------------ + * Encryption is now supported for RSA authentication. + + Currently, these auth configurations will cause a crash: + auth = md5,rsa + auth = plaintext,md5,rsa + + With a patched peer, the following will cause a crash: + auth = rsa + auth = md5,rsa + auth = plaintext,md5,rsa + + If both the peer and user are patches, no crash occurs. + Existing good configurations should continue to work. + +res_http_media_cache +------------------ + * When fetching a file for playback from a URL, Asterisk will now first + use the value of the Content-Type header in the HTTP response to + determine the format of the audio data, and only if it is unable to do + that will it attempt to parse the URL and extract the extension from + the path portion. Previously Asterisk would first look at the end of + the URL, which may have included query string parameters or a URL + fragment, which was error prone. + +res_pjsip +------------------ + * The 'async_operations' setting on transports is no longer + obeyed and instead is always set to 1. This is due to the + functionality not being applicable to Asterisk and causing + excess unnecessary memory usage. This setting will now be + ignored but can also be removed from the configuration file. + ------------------------------------------------------------------------------ --- New functionality introduced in Asterisk 19.0.0 -------------------------- ------------------------------------------------------------------------------ diff --git a/doc/CHANGES-staging/add_mix_option_to_playback.txt b/doc/CHANGES-staging/add_mix_option_to_playback.txt deleted file mode 100644 index cfc876ce77..0000000000 --- a/doc/CHANGES-staging/add_mix_option_to_playback.txt +++ /dev/null @@ -1,7 +0,0 @@ -Subject: app_playback -Subject: apps - -A new option 'mix' is added to the Playback application that -will play by filename and say.conf. It will look on the format of the -name, if it is like say format it will play with say.conf if not it -will play the file name. \ No newline at end of file diff --git a/doc/CHANGES-staging/allow_wildcard_certs.txt b/doc/CHANGES-staging/allow_wildcard_certs.txt deleted file mode 100644 index 29a53dd2dc..0000000000 --- a/doc/CHANGES-staging/allow_wildcard_certs.txt +++ /dev/null @@ -1,9 +0,0 @@ -Subject: res_pjsip - -A new transport option 'allow_wildcard_certs' has been added that when it -and 'verify_server' are both set to 'yes', enables verification against -wildcards, i.e. '*.' in certs for common, and subject alt names of type DNS -for TLS transport types. Names must start with the wildcard. Partial wildcards, -e.g. 'f*.example.com' and 'foo.*.com' are not allowed. As well, names only -match against a single level meaning '*.example.com' matches 'foo.example.com', -but not 'foo.bar.example.com'. diff --git a/doc/CHANGES-staging/ami_wink.txt b/doc/CHANGES-staging/ami_wink.txt deleted file mode 100644 index 9d27cca87f..0000000000 --- a/doc/CHANGES-staging/ami_wink.txt +++ /dev/null @@ -1,3 +0,0 @@ -Subject: ami - -An AMI event now exists for "Wink". diff --git a/doc/CHANGES-staging/app_confbridge_channels.txt b/doc/CHANGES-staging/app_confbridge_channels.txt deleted file mode 100644 index 485f664268..0000000000 --- a/doc/CHANGES-staging/app_confbridge_channels.txt +++ /dev/null @@ -1,7 +0,0 @@ -Subject: app_confbridge - -Adds the CONFBRIDGE_CHANNELS function which can -be used to retrieve a list of channels in a ConfBridge, -optionally filtered by a particular category. This -list can then be used with functions like SHIFT, POP, -UNSHIFT, etc. diff --git a/doc/CHANGES-staging/app_confbridge_hear_join.txt b/doc/CHANGES-staging/app_confbridge_hear_join.txt deleted file mode 100644 index 40f23836ff..0000000000 --- a/doc/CHANGES-staging/app_confbridge_hear_join.txt +++ /dev/null @@ -1,8 +0,0 @@ -Subject: app_confbridge - -Added the hear_own_join_sound option to the confbridge user profile to -control who hears the sound_join audio file. When set to 'yes' the user -entering the conference and the participants already in the conference -will hear the sound_join audio file. When set to 'no' the user entering -the conference will not hear the sound_join audio file, but the -participants already in the conference will hear the sound_join audio file. diff --git a/doc/CHANGES-staging/app_dtmfstore.txt b/doc/CHANGES-staging/app_dtmfstore.txt deleted file mode 100644 index a82b5438bd..0000000000 --- a/doc/CHANGES-staging/app_dtmfstore.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: app_dtmfstore - -New application which collects digits -dialed and stores them into -a specified variable. - diff --git a/doc/CHANGES-staging/app_mf_maxdigits.txt b/doc/CHANGES-staging/app_mf_maxdigits.txt deleted file mode 100644 index 429269005e..0000000000 --- a/doc/CHANGES-staging/app_mf_maxdigits.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: app_mf - -Adds an option to ReceiveMF to cap the -number of digits read at a user-specified -maximum. diff --git a/doc/CHANGES-staging/app_mf_mf.txt b/doc/CHANGES-staging/app_mf_mf.txt deleted file mode 100644 index 3168f2adbc..0000000000 --- a/doc/CHANGES-staging/app_mf_mf.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: app_mf - -Adds MF receiver and sender applications to support -the R1 MF signaling protocol, including integration -with the Dial application. diff --git a/doc/CHANGES-staging/app_milliwatt.txt b/doc/CHANGES-staging/app_milliwatt.txt deleted file mode 100644 index 434ace22bb..0000000000 --- a/doc/CHANGES-staging/app_milliwatt.txt +++ /dev/null @@ -1,11 +0,0 @@ -Subject: app_milliwatt - -The Milliwatt application's existing behavior is -incorrect in that it plays a constant tone, which -is not how digital milliwatt test lines actually -work. - -An option is added so that a proper milliwatt test -tone can be provided, including a 1 second silent -interval every 10 seconds. However, for compatability -reasons, the default behavior remains unchanged. diff --git a/doc/CHANGES-staging/app_morsecode.txt b/doc/CHANGES-staging/app_morsecode.txt deleted file mode 100644 index b9e49b63ee..0000000000 --- a/doc/CHANGES-staging/app_morsecode.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: app_morsecode - -Extends the Morsecode application by adding support for -American Morse code and adds a configurable option -for the frequency used in off intervals. - diff --git a/doc/CHANGES-staging/app_originate_codecs.txt b/doc/CHANGES-staging/app_originate_codecs.txt deleted file mode 100644 index a0f52b13c5..0000000000 --- a/doc/CHANGES-staging/app_originate_codecs.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: app_originate - -Codecs can now be specified for dialplan-originated -calls, as with call files and the manager action. -By default, only the slin codec is now used, instead -of all the slin* codecs. diff --git a/doc/CHANGES-staging/app_queue.txt b/doc/CHANGES-staging/app_queue.txt deleted file mode 100644 index 5d677b56b9..0000000000 --- a/doc/CHANGES-staging/app_queue.txt +++ /dev/null @@ -1,4 +0,0 @@ -Subject: app_queue.c - -Allow multiple files to be streamed for agent announcement. - diff --git a/doc/CHANGES-staging/app_queue_DIALEDPEERNUMBER.txt b/doc/CHANGES-staging/app_queue_DIALEDPEERNUMBER.txt deleted file mode 100644 index ef15e9e4ea..0000000000 --- a/doc/CHANGES-staging/app_queue_DIALEDPEERNUMBER.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: app_queue -Subject: Applications - -added that we set DIALEDPEERNUMBER on the outgoing channels -so it is avalible in b(content^extension^line) -this add the same behaviour as Dial diff --git a/doc/CHANGES-staging/app_queue_logintime.txt b/doc/CHANGES-staging/app_queue_logintime.txt deleted file mode 100644 index 5b0eea414f..0000000000 --- a/doc/CHANGES-staging/app_queue_logintime.txt +++ /dev/null @@ -1,9 +0,0 @@ -Subject: app_queue - -Add field to save the time value when a member enter a queue. -Shows this time in seconds using 'queue show' command and the -field LoginTime for responses for AMI the events. - -The output for the CLI command `queue show` is changed by added a -extra data field for the information of the time login time for each -member. diff --git a/doc/CHANGES-staging/app_queue_music.txt b/doc/CHANGES-staging/app_queue_music.txt deleted file mode 100644 index 254a45db45..0000000000 --- a/doc/CHANGES-staging/app_queue_music.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: app_queue - -The m option now allows an override music on hold -class to be specified for the Queue application -within the dialplan. diff --git a/doc/CHANGES-staging/app_queue_nordic_language.txt b/doc/CHANGES-staging/app_queue_nordic_language.txt deleted file mode 100644 index 72efd78001..0000000000 --- a/doc/CHANGES-staging/app_queue_nordic_language.txt +++ /dev/null @@ -1,3 +0,0 @@ -Subject: app_queues - -adding support for playing the correct en/et for nordic languages diff --git a/doc/CHANGES-staging/app_queue_say_thanks.txt b/doc/CHANGES-staging/app_queue_say_thanks.txt deleted file mode 100644 index 7bf7b7b420..0000000000 --- a/doc/CHANGES-staging/app_queue_say_thanks.txt +++ /dev/null @@ -1,4 +0,0 @@ -Subject: app_queues - -Don't play sound_thanks if there is no leading hold_time message -When the only announcement is hold time, and there is no hold time (0 min, 0 sec), asterisk will say "thank you for your patience" diff --git a/doc/CHANGES-staging/app_queue_stats.txt b/doc/CHANGES-staging/app_queue_stats.txt deleted file mode 100644 index 36c0c3da06..0000000000 --- a/doc/CHANGES-staging/app_queue_stats.txt +++ /dev/null @@ -1,7 +0,0 @@ -Subject: app_queue - -Reload behavior in app_queue has been changed so -queue and agent stats are not reset during full -app_queue module reloads. The queue reset stats -CLI command may still be used to reset stats while -Asterisk is running. diff --git a/doc/CHANGES-staging/app_read.txt b/doc/CHANGES-staging/app_read.txt deleted file mode 100644 index df3247c1e1..0000000000 --- a/doc/CHANGES-staging/app_read.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: app_read - -A new option allows the digit '#' to be read literally, -rather than used exclusively as the input terminator -character. diff --git a/doc/CHANGES-staging/app_sendtext.txt b/doc/CHANGES-staging/app_sendtext.txt deleted file mode 100644 index 37dd64bace..0000000000 --- a/doc/CHANGES-staging/app_sendtext.txt +++ /dev/null @@ -1,4 +0,0 @@ -Subject: app_sendtext - -A ReceiveText application has been added that can be -used in conjunction with the SendText application. diff --git a/doc/CHANGES-staging/app_voicemail.txt b/doc/CHANGES-staging/app_voicemail.txt deleted file mode 100644 index c52d1f0666..0000000000 --- a/doc/CHANGES-staging/app_voicemail.txt +++ /dev/null @@ -1,7 +0,0 @@ -Subject: app_voicemail - -Add a new 'S' option to VoiceMail which prevents the instructions -(vm-intro) from being played if a busy/unavailable/temporary greeting -from the voicemail user is played. This is similar to the existing 's' -option except that instructions will still be played if no user -greeting is available. diff --git a/doc/CHANGES-staging/app_voicemail_danish_syntax.txt b/doc/CHANGES-staging/app_voicemail_danish_syntax.txt deleted file mode 100644 index 5e6cdd37bf..0000000000 --- a/doc/CHANGES-staging/app_voicemail_danish_syntax.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: app_voicemail -Subject: Applications - -added support for Danish syntax, playing the correct plural sound file -dependen on where you have 1 or multipe messages -based on the existing SE/NO code diff --git a/doc/CHANGES-staging/app_voicemail_nodelete.txt b/doc/CHANGES-staging/app_voicemail_nodelete.txt deleted file mode 100644 index ef9589652d..0000000000 --- a/doc/CHANGES-staging/app_voicemail_nodelete.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: app_voicemail - -The r option has been added, which prevents deletion -of messages from VoiceMailMain, which can be -useful for shared mailboxes. diff --git a/doc/CHANGES-staging/ari_add_pvt_id_to_channel_resource.txt b/doc/CHANGES-staging/ari_add_pvt_id_to_channel_resource.txt deleted file mode 100644 index a4f008f967..0000000000 --- a/doc/CHANGES-staging/ari_add_pvt_id_to_channel_resource.txt +++ /dev/null @@ -1,7 +0,0 @@ -Subject: ari -Subject: stasis_channels - -Expose channel driver's unique id (which is the Call-ID for SIP/PJSIP) -to ARI channel resources as 'protocol_id'. - -ASTERISK-30027 diff --git a/doc/CHANGES-staging/ast_coredumper.txt b/doc/CHANGES-staging/ast_coredumper.txt deleted file mode 100644 index bbff0da290..0000000000 --- a/doc/CHANGES-staging/ast_coredumper.txt +++ /dev/null @@ -1,23 +0,0 @@ -Subject: ast_coredumper - -New options: - --pid= - Allows specification of an Asterisk instance when trying to - and the script can't determine it itself. - --libdir= - 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. diff --git a/doc/CHANGES-staging/bundled-pjproject-build.txt b/doc/CHANGES-staging/bundled-pjproject-build.txt deleted file mode 100644 index 976c0f5a93..0000000000 --- a/doc/CHANGES-staging/bundled-pjproject-build.txt +++ /dev/null @@ -1,8 +0,0 @@ -Subject: Core - -Bundled PJProject Build - -The build process has been updated to make pjproject troubleshooting -and development easier. See third-party/pjproject/README-hacking.md or -https://wiki.asterisk.org/wiki/display/AST/Bundled+PJProject -for more info. diff --git a/doc/CHANGES-staging/cdr_disable.txt b/doc/CHANGES-staging/cdr_disable.txt deleted file mode 100644 index cae7a7c333..0000000000 --- a/doc/CHANGES-staging/cdr_disable.txt +++ /dev/null @@ -1,8 +0,0 @@ -Subject: cdr - -A new CDR option, channeldefaultenabled, allows controlling -whether CDR is enabled or disabled by default on -newly created channels. The default behavior remains -unchanged from previous versions of Asterisk (new -channels will have CDR enabled, as long as CDR is -enabled globally). diff --git a/doc/CHANGES-staging/chan_dahdi_cadences.txt b/doc/CHANGES-staging/chan_dahdi_cadences.txt deleted file mode 100644 index b888926eee..0000000000 --- a/doc/CHANGES-staging/chan_dahdi_cadences.txt +++ /dev/null @@ -1,8 +0,0 @@ -Subject: chan_dahdi - -Previously, cadences were appended on dahdi restart, -rather than reloaded. This prevented cadences from -being updated and maxed out the available cadences -if reloaded multiple times. This behavior is fixed -so that reloading cadences is idempotent and cadences -can actually be reloaded. diff --git a/doc/CHANGES-staging/chan_dahdi_polarity.txt b/doc/CHANGES-staging/chan_dahdi_polarity.txt deleted file mode 100644 index 365ab200dd..0000000000 --- a/doc/CHANGES-staging/chan_dahdi_polarity.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: chan_dahdi - -A POLARITY function is now available that allows -getting or setting the polarity on a channel -from the dialplan. diff --git a/doc/CHANGES-staging/chan_iax2_ani2.txt b/doc/CHANGES-staging/chan_iax2_ani2.txt deleted file mode 100644 index 37c6fa6cf6..0000000000 --- a/doc/CHANGES-staging/chan_iax2_ani2.txt +++ /dev/null @@ -1,4 +0,0 @@ -Subject: chan_iax2 - -ANI2 (OLI) is now transmitted over IAX2 calls -as an information element. diff --git a/doc/CHANGES-staging/chan_iax2_dial.txt b/doc/CHANGES-staging/chan_iax2_dial.txt deleted file mode 100644 index a95832b0b1..0000000000 --- a/doc/CHANGES-staging/chan_iax2_dial.txt +++ /dev/null @@ -1,4 +0,0 @@ -Subject: chan_iax2 - -Both a secret and an outkey may be specified at dial time, -since encryption is possible with RSA authentication. diff --git a/doc/CHANGES-staging/chan_pjsip_180_sdp.txt b/doc/CHANGES-staging/chan_pjsip_180_sdp.txt deleted file mode 100644 index ffd14af10c..0000000000 --- a/doc/CHANGES-staging/chan_pjsip_180_sdp.txt +++ /dev/null @@ -1,8 +0,0 @@ -Subject: chan_pjsip - -added global config option "allow_sending_180_after_183" - -Allow Asterisk to send 180 Ringing to an endpoint -after 183 Session Progress has been send. -If disabled Asterisk will instead send only a -183 Session Progress to the endpoint. diff --git a/doc/CHANGES-staging/chan_pjsip_flash.txt b/doc/CHANGES-staging/chan_pjsip_flash.txt deleted file mode 100644 index 34da796857..0000000000 --- a/doc/CHANGES-staging/chan_pjsip_flash.txt +++ /dev/null @@ -1,4 +0,0 @@ -Subject: chan_pjsip - -Hook flash events can now be sent on a PJSIP channel -if requested to do so. diff --git a/doc/CHANGES-staging/chan_sip_pickup_AST_STATE_DOWN.txt b/doc/CHANGES-staging/chan_sip_pickup_AST_STATE_DOWN.txt deleted file mode 100644 index e658faa52a..0000000000 --- a/doc/CHANGES-staging/chan_sip_pickup_AST_STATE_DOWN.txt +++ /dev/null @@ -1,3 +0,0 @@ -Subject: chan_sip.c - -resolve issue with pickup on device that uses "183" and not "180" diff --git a/doc/CHANGES-staging/chan_sip_session-timer_on_update.txt b/doc/CHANGES-staging/chan_sip_session-timer_on_update.txt deleted file mode 100644 index 259782f518..0000000000 --- a/doc/CHANGES-staging/chan_sip_session-timer_on_update.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: chan_sip - -Session timers get removed on UPDATE -Fix if Asterisk receives a SIP REFER with Session-Timers UAC -that Asterisk maintains Session-Timers when sending UPDATE request - diff --git a/doc/CHANGES-staging/channel_internal_api.txt b/doc/CHANGES-staging/channel_internal_api.txt deleted file mode 100644 index f40a4c70fe..0000000000 --- a/doc/CHANGES-staging/channel_internal_api.txt +++ /dev/null @@ -1,4 +0,0 @@ -Subject: channel_internal_api - -CHANNEL(lastcontext) and CHANNEL(lastexten) -are now available for use in the dialplan. diff --git a/doc/CHANGES-staging/cli_eval_function.txt b/doc/CHANGES-staging/cli_eval_function.txt deleted file mode 100644 index 9f7873c738..0000000000 --- a/doc/CHANGES-staging/cli_eval_function.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: cli - -A new CLI command 'dialplan eval function' has been -added which allows users to test the behavior of -dialplan function calls directly from the CLI. diff --git a/doc/CHANGES-staging/cli_refresh.txt b/doc/CHANGES-staging/cli_refresh.txt deleted file mode 100644 index 82bcd23f9a..0000000000 --- a/doc/CHANGES-staging/cli_refresh.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: cli - -The "module refresh" command has been added, -which allows unloading and then loading a -module with a single command. diff --git a/doc/CHANGES-staging/func_channel.txt b/doc/CHANGES-staging/func_channel.txt deleted file mode 100644 index 7f92c3e014..0000000000 --- a/doc/CHANGES-staging/func_channel.txt +++ /dev/null @@ -1,4 +0,0 @@ -Subject: func_channel - -Adds the CHANNEL_EXISTS function to check for the existence -of a channel by name or unique ID. diff --git a/doc/CHANGES-staging/func_db.txt b/doc/CHANGES-staging/func_db.txt deleted file mode 100644 index 72e333a547..0000000000 --- a/doc/CHANGES-staging/func_db.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: func_db - -The function DB_KEYCOUNT has been added, which -returns the cardinality of the keys at a specified -prefix in AstDB, i.e. the number of keys at a -given prefix. diff --git a/doc/CHANGES-staging/func_env.txt b/doc/CHANGES-staging/func_env.txt deleted file mode 100644 index af03d5f0d1..0000000000 --- a/doc/CHANGES-staging/func_env.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: func_env.c - -Two new functions, DIRNAME and BASENAME, are now -included which allow users to obtain the directory -or the base filename of any file. diff --git a/doc/CHANGES-staging/func_evalexten.txt b/doc/CHANGES-staging/func_evalexten.txt deleted file mode 100644 index f912bbeb5f..0000000000 --- a/doc/CHANGES-staging/func_evalexten.txt +++ /dev/null @@ -1,4 +0,0 @@ -Subject: func_evalexten - -This adds the EVAL_EXTEN function which may be -used to evaluate data at dialplan extensions. diff --git a/doc/CHANGES-staging/func_framedrop.txt b/doc/CHANGES-staging/func_framedrop.txt deleted file mode 100644 index c17bccd74c..0000000000 --- a/doc/CHANGES-staging/func_framedrop.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: func_framedrop - -New function to selectively drop specified frames -in either direction on a channel. - diff --git a/doc/CHANGES-staging/func_json.txt b/doc/CHANGES-staging/func_json.txt deleted file mode 100644 index 79bb2da87d..0000000000 --- a/doc/CHANGES-staging/func_json.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: func_json - -The JSON_DECODE dialplan function can now be used -to parse JSON strings, such as in conjunction with -CURL for using API responses. diff --git a/doc/CHANGES-staging/func_odbc_esc_backslashes.txt b/doc/CHANGES-staging/func_odbc_esc_backslashes.txt deleted file mode 100644 index 087bb42141..0000000000 --- a/doc/CHANGES-staging/func_odbc_esc_backslashes.txt +++ /dev/null @@ -1,7 +0,0 @@ -Subject: func_odbc - -A SQL_ESC_BACKSLASHES dialplan function has been added which -escapes backslashes. Usage of this is dependent on whether the -database in use can use backslashes to escape ticks or not. If -it can, then usage of this prevents a broken SQL query depending -on how the SQL query is constructed. diff --git a/doc/CHANGES-staging/func_scramble.txt b/doc/CHANGES-staging/func_scramble.txt deleted file mode 100644 index 4c1ffab78b..0000000000 --- a/doc/CHANGES-staging/func_scramble.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: func_scramble - -Adds an audio scrambler function that may be used to -distort voice audio on a channel as a privacy -enhancement. diff --git a/doc/CHANGES-staging/func_strings.txt b/doc/CHANGES-staging/func_strings.txt deleted file mode 100644 index d154464021..0000000000 --- a/doc/CHANGES-staging/func_strings.txt +++ /dev/null @@ -1,7 +0,0 @@ -Subject: func_strings - -A new STRBETWEEN function is now included which -allows a substring to be inserted between characters -in a string. This is particularly useful for transforming -dial strings, such as adding pauses between digits -for a string of digits that are sent to another channel. diff --git a/doc/CHANGES-staging/func_vmcount.txt b/doc/CHANGES-staging/func_vmcount.txt deleted file mode 100644 index ba2a0a1178..0000000000 --- a/doc/CHANGES-staging/func_vmcount.txt +++ /dev/null @@ -1,3 +0,0 @@ -Subject: func_vmcount - -Multiple mailboxes may now be specified instead of just one. diff --git a/doc/CHANGES-staging/load_realtime_queues.txt b/doc/CHANGES-staging/load_realtime_queues.txt deleted file mode 100644 index 68a4a8bcaf..0000000000 --- a/doc/CHANGES-staging/load_realtime_queues.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: app_queue - -Load queues and members from Realtime for -AMI actions: QueuePause, QueueStatus and QueueSummary, -Applications: PauseQueueMember and UnpauseQueueMember. diff --git a/doc/CHANGES-staging/logger.txt b/doc/CHANGES-staging/logger.txt deleted file mode 100644 index d09ebccca2..0000000000 --- a/doc/CHANGES-staging/logger.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: logger - -Added the ability to define custom log levels in logger.conf -and use them in the Log dialplan application. Also adds a -logger show levels CLI command. diff --git a/doc/CHANGES-staging/manager_disable.txt b/doc/CHANGES-staging/manager_disable.txt deleted file mode 100644 index 762ceca19e..0000000000 --- a/doc/CHANGES-staging/manager_disable.txt +++ /dev/null @@ -1,4 +0,0 @@ -Subject: ami - -AMI events can now be globally disabled using -the disabledevents [general] setting. diff --git a/doc/CHANGES-staging/manager_message_send.txt b/doc/CHANGES-staging/manager_message_send.txt deleted file mode 100644 index ab5b58a287..0000000000 --- a/doc/CHANGES-staging/manager_message_send.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: MessageSend - -The MessageSend AMI action has been updated to allow the Destination -and the To addresses to be provided separately. This brings the -MessageSend manager command in line with the capabilities of the -MessageSend dialplan application. diff --git a/doc/CHANGES-staging/mf.txt b/doc/CHANGES-staging/mf.txt deleted file mode 100644 index 644f62a998..0000000000 --- a/doc/CHANGES-staging/mf.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: Channel-agnostic MF support - -A SendMF application and PlayMF manager -application are now included to send -arbitrary standard R1 MF tones on the -current channel or another specified channel. diff --git a/doc/CHANGES-staging/pjsip_read_headers.txt b/doc/CHANGES-staging/pjsip_read_headers.txt deleted file mode 100644 index 4dc641cdae..0000000000 --- a/doc/CHANGES-staging/pjsip_read_headers.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: chan_pjsip - -Add function PJSIP_HEADERS() to get list of headers by pattern in the same way as SIP_HEADERS() do. - -Add ability to read header by pattern using PJSIP_HEADER(). diff --git a/doc/CHANGES-staging/queue_withdraw_caller.txt b/doc/CHANGES-staging/queue_withdraw_caller.txt deleted file mode 100644 index 04e43d0770..0000000000 --- a/doc/CHANGES-staging/queue_withdraw_caller.txt +++ /dev/null @@ -1,14 +0,0 @@ -Subject: app_queue - -Added a new AMI action: QueueWithdrawCaller -This AMI action makes it possible to withdraw a caller from a queue -back to the dialplan. The call will be signaled to leave the queue -whenever it can, hence, it not guaranteed that the call will leave -the queue. - -Optional custom data can be passed in the request, in the WithdrawInfo -parameter. If the call successfully withdrawn the queue, -it can be retrieved using the QUEUE_WITHDRAW_INFO variable. - -This can be useful for certain uses, such as dispatching the call -to a specific extension. diff --git a/doc/CHANGES-staging/res_agi.txt b/doc/CHANGES-staging/res_agi.txt deleted file mode 100644 index eb6132d614..0000000000 --- a/doc/CHANGES-staging/res_agi.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: res_agi - -Agi command 'exec' can now be enabled -to evaluate dialplan functions and variables -by setting the variable AGIEXECFULL to yes. \ No newline at end of file diff --git a/doc/CHANGES-staging/res_cliexec.txt b/doc/CHANGES-staging/res_cliexec.txt deleted file mode 100644 index 2b1fe7679c..0000000000 --- a/doc/CHANGES-staging/res_cliexec.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: res_cliexec - -A new CLI command, dialplan exec application, has -been added which allows dialplan applications to be -executed at the CLI, useful for some quick testing -without needing to write dialplan. diff --git a/doc/CHANGES-staging/res_fax_spandsp.txt b/doc/CHANGES-staging/res_fax_spandsp.txt deleted file mode 100644 index 4ad351fb8e..0000000000 --- a/doc/CHANGES-staging/res_fax_spandsp.txt +++ /dev/null @@ -1,3 +0,0 @@ -Subject: res_fax_spandsp - -Adds support for spandsp 3.0.0. diff --git a/doc/CHANGES-staging/res_geolocation.txt b/doc/CHANGES-staging/res_geolocation.txt deleted file mode 100644 index 5fe7316333..0000000000 --- a/doc/CHANGES-staging/res_geolocation.txt +++ /dev/null @@ -1,4 +0,0 @@ -Subject: res_geolocation - -Added res_geolocation which creates the core capabilities -to manipulate Geolocation information on SIP INVITEs. diff --git a/doc/CHANGES-staging/res_parking_moh.txt b/doc/CHANGES-staging/res_parking_moh.txt deleted file mode 100644 index 50f589ca43..0000000000 --- a/doc/CHANGES-staging/res_parking_moh.txt +++ /dev/null @@ -1,4 +0,0 @@ -Subject: res_parking - -An m option to Park and ParkAndAnnounce now allows -specifying a music on hold class override. diff --git a/doc/CHANGES-staging/res_pjproject.txt b/doc/CHANGES-staging/res_pjproject.txt deleted file mode 100644 index 132c9506b8..0000000000 --- a/doc/CHANGES-staging/res_pjproject.txt +++ /dev/null @@ -1,8 +0,0 @@ -Subject: res_pjproject - -In pjproject.conf you can now map pjproject log levels -to the Asterisk TRACE log level. The default mappings -have therefore changed so that only pjproject levels -3 and 4 are mapped to DEBUG and 5 and 6 are now mapped -to TRACE. Previously 3, 4, 5, and 6 were all mapped to -DEBUG. diff --git a/doc/CHANGES-staging/res_pjsip_geolocation.txt b/doc/CHANGES-staging/res_pjsip_geolocation.txt deleted file mode 100644 index acc49063e0..0000000000 --- a/doc/CHANGES-staging/res_pjsip_geolocation.txt +++ /dev/null @@ -1,4 +0,0 @@ -Subject: res_pjsip_geolocation - -Added res_pjsip_geolocation which gives chan_pjsip -the ability to use the core geolocation capabilities. diff --git a/doc/CHANGES-staging/res_pjsip_header_funcs.txt b/doc/CHANGES-staging/res_pjsip_header_funcs.txt deleted file mode 100644 index 88946e4808..0000000000 --- a/doc/CHANGES-staging/res_pjsip_header_funcs.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: res_pjsip_header_funcs - -Add function PJSIP_RESPONSE_HEADERS() to get list of header names from 200 response, in the same way as PJSIP_HEADERS() from the request. - -Add function PJSIP_RESPONSE_HEADER() to read header from 200 response, in the same way as PJSIP_HEADER() from the request. diff --git a/doc/CHANGES-staging/res_pjsip_registrar.txt b/doc/CHANGES-staging/res_pjsip_registrar.txt deleted file mode 100644 index a80f69ff08..0000000000 --- a/doc/CHANGES-staging/res_pjsip_registrar.txt +++ /dev/null @@ -1,7 +0,0 @@ -Subject: res_pjsip_registrar - -Adds new PJSIP AOR option remove_unavailable to either -remove unavailable contacts when a REGISTER exceeds -max_contacts when remove_existing is disabled, or -prioritize unavailable contacts over other existing -contacts when remove_existing is enabled. diff --git a/doc/CHANGES-staging/res_pjsip_t38_bind_fixes.txt b/doc/CHANGES-staging/res_pjsip_t38_bind_fixes.txt deleted file mode 100644 index d7bc8a1e9f..0000000000 --- a/doc/CHANGES-staging/res_pjsip_t38_bind_fixes.txt +++ /dev/null @@ -1,9 +0,0 @@ -Subject: res_pjsip_t38 - -In res_pjsip_sdp_rtp, the bind_rtp_to_media_address option and the -fallback use of the transport's bind address solve problems sending -media on systems that cannot send ipv4 packets on ipv6 sockets, and -certain other situations. This change extends both of these behaviors -to UDPTL sessions as well in res_pjsip_t38, to fix fax-specific -problems on these systems, introducing a new option -endpoint/t38_bind_udptl_to_media_address. diff --git a/doc/CHANGES-staging/res_rtp_asterisk_stunaddr_recurring_resolution.txt b/doc/CHANGES-staging/res_rtp_asterisk_stunaddr_recurring_resolution.txt deleted file mode 100644 index c78f4f51d4..0000000000 --- a/doc/CHANGES-staging/res_rtp_asterisk_stunaddr_recurring_resolution.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: res_rtp_asterisk - -When the address of the STUN server (stunaddr) is a name resolved via DNS, the -stunaddr will be recurringly resolved when the DNS answer Time-To-Live (TTL) -expires. This allows the STUN server to change its IP address without having to -reload the res_rtp_asterisk module. diff --git a/doc/CHANGES-staging/res_statsd.txt b/doc/CHANGES-staging/res_statsd.txt deleted file mode 100644 index 317c65d00b..0000000000 --- a/doc/CHANGES-staging/res_statsd.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: Handle non-standard Meter metric type safely - -A meter_support flag has been introduced that defaults to true to maintain current behaviour. -If disabled, a counter metric type will be used instead wherever a meter metric type was used, -the counter will have a "_meter" suffix appended to the metric name. \ No newline at end of file diff --git a/doc/CHANGES-staging/res_tonedetect.txt b/doc/CHANGES-staging/res_tonedetect.txt deleted file mode 100644 index ddda8e899e..0000000000 --- a/doc/CHANGES-staging/res_tonedetect.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: res_tonedetect - -Arbitrary tone detection is now available through a -WaitForTone application (blocking) and a TONE_DETECT -function (non-blocking). diff --git a/doc/CHANGES-staging/rls_display_name.txt b/doc/CHANGES-staging/rls_display_name.txt deleted file mode 100644 index 0d95b08fa3..0000000000 --- a/doc/CHANGES-staging/rls_display_name.txt +++ /dev/null @@ -1,10 +0,0 @@ -Subject: res_pjsip_pubsub - -A new resource_list option, resource_display_name, indicates -whether display name of resource or the resource name being -provided for RLS entries. -If this option is enabled, the Display Name will be provided. -This option is disabled by default to remain the previous behavior. -If the 'event' set to 'presence' or 'dialog' the non-empty HINT name -will be set as the Display Name. -The 'message-summary' is not supported yet. diff --git a/doc/CHANGES-staging/rls_refresh.txt b/doc/CHANGES-staging/rls_refresh.txt deleted file mode 100644 index fb36160bef..0000000000 --- a/doc/CHANGES-staging/rls_refresh.txt +++ /dev/null @@ -1,7 +0,0 @@ -Subject: res_pjsip_pubsub - -The Resource List Subscriptions (RLS) is dynamic now. -The asterisk now updates current subscriptions to reflect the changes -to the list on subscription refresh. If list items are added, -removed, updated or do not exist anymore, the asterisk regenerates -the resource list. diff --git a/doc/CHANGES-staging/say.txt b/doc/CHANGES-staging/say.txt deleted file mode 100644 index 115ceea15f..0000000000 --- a/doc/CHANGES-staging/say.txt +++ /dev/null @@ -1,7 +0,0 @@ -Subject: say.c - -Adds SAYFILES function to retrieve the file names that would -be played by corresponding Say applications, such as -SayDigits, SayAlpha, etc. - -Additionally adds SayMoney and SayOrdinal applications. diff --git a/doc/CHANGES-staging/tonescan.txt b/doc/CHANGES-staging/tonescan.txt deleted file mode 100644 index cbed34fa09..0000000000 --- a/doc/CHANGES-staging/tonescan.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: ToneScan application - -A new application, ToneScan, allows for -synchronous detection of call progress -signals such as dial tone, busy tone, -Special Information Tones, and modems. diff --git a/doc/UPGRADE-staging/chan_iax2_rsa.txt b/doc/UPGRADE-staging/chan_iax2_rsa.txt deleted file mode 100644 index d5a9770862..0000000000 --- a/doc/UPGRADE-staging/chan_iax2_rsa.txt +++ /dev/null @@ -1,15 +0,0 @@ -Subject: chan_iax2 - -Encryption is now supported for RSA authentication. - -Currently, these auth configurations will cause a crash: -auth = md5,rsa -auth = plaintext,md5,rsa - -With a patched peer, the following will cause a crash: -auth = rsa -auth = md5,rsa -auth = plaintext,md5,rsa - -If both the peer and user are patches, no crash occurs. -Existing good configurations should continue to work. diff --git a/doc/UPGRADE-staging/http-media-cache-lookup-order.txt b/doc/UPGRADE-staging/http-media-cache-lookup-order.txt deleted file mode 100644 index 83c31dcbcb..0000000000 --- a/doc/UPGRADE-staging/http-media-cache-lookup-order.txt +++ /dev/null @@ -1,9 +0,0 @@ -Subject: res_http_media_cache - -When fetching a file for playback from a URL, Asterisk will now first -use the value of the Content-Type header in the HTTP response to -determine the format of the audio data, and only if it is unable to do -that will it attempt to parse the URL and extract the extension from -the path portion. Previously Asterisk would first look at the end of -the URL, which may have included query string parameters or a URL -fragment, which was error prone. diff --git a/doc/UPGRADE-staging/manager_amxml_attribute_fix.txt b/doc/UPGRADE-staging/manager_amxml_attribute_fix.txt deleted file mode 100644 index 4b15ee92ec..0000000000 --- a/doc/UPGRADE-staging/manager_amxml_attribute_fix.txt +++ /dev/null @@ -1,8 +0,0 @@ -Subject: AMI - -The XML Manager Event Interface (amxml) now generates attribute names -that are compliant with the XML 1.1 specification. Previously, an -attribute name that started with a digit would be rendered as-is, even -though attribute names must not begin with a digit. We now prefix -attribute names that start with a digit with an underscore ('_') to -prevent XML validation failures. diff --git a/doc/UPGRADE-staging/res_monitor_disabled.txt b/doc/UPGRADE-staging/res_monitor_disabled.txt deleted file mode 100644 index 12cc372f54..0000000000 --- a/doc/UPGRADE-staging/res_monitor_disabled.txt +++ /dev/null @@ -1,8 +0,0 @@ -Subject: res_monitor -Master-Only: True - -This module is no longer built by default in -accordance with the Module Deprecation Policy. -If you require this functionality you will need -to enable it for building in menuselect. Note -that in the future res_monitor will be removed. diff --git a/doc/UPGRADE-staging/res_pjsip_async_operations.txt b/doc/UPGRADE-staging/res_pjsip_async_operations.txt deleted file mode 100644 index cf9f9426da..0000000000 --- a/doc/UPGRADE-staging/res_pjsip_async_operations.txt +++ /dev/null @@ -1,7 +0,0 @@ -Subject: res_pjsip - -The 'async_operations' setting on transports is no longer -obeyed and instead is always set to 1. This is due to the -functionality not being applicable to Asterisk and causing -excess unnecessary memory usage. This setting will now be -ignored but can also be removed from the configuration file. diff --git a/doc/UPGRADE-staging/stir_shaken_option_split.txt b/doc/UPGRADE-staging/stir_shaken_option_split.txt deleted file mode 100644 index 79df214a8b..0000000000 --- a/doc/UPGRADE-staging/stir_shaken_option_split.txt +++ /dev/null @@ -1,7 +0,0 @@ -Subject: STIR/SHAKEN - -The STIR/SHAKEN configuration option has been split into -4 different choices: off, attest, verify, and on. Off and -on behave the same way as before. Attest will only perform -attestation on the endpoint, and verify will only perform -verification on the endpoint.