From c3c82441a2faf35f281c9ee6424751fdc9b976fc Mon Sep 17 00:00:00 2001 From: George Joseph Date: Wed, 9 Aug 2023 11:55:11 -0600 Subject: [PATCH] Prepare master for Asterisk 22 --- doc/CHANGES-staging/README.md | 37 -------------- doc/CHANGES-staging/ami_hook_flash.txt | 4 -- doc/CHANGES-staging/answer.txt | 5 -- doc/CHANGES-staging/app_amd.txt | 5 -- doc/CHANGES-staging/app_bridgewait.txt | 4 -- doc/CHANGES-staging/app_broadcast.txt | 4 -- .../app_confbridge_marked_any.txt | 5 -- .../app_directory_skip_call.txt | 5 -- doc/CHANGES-staging/app_if.txt | 4 -- doc/CHANGES-staging/app_mixmonitor_clid.txt | 5 -- doc/CHANGES-staging/app_mixmonitor_delete.txt | 6 --- .../app_mixmonitor_mute_by_id.txt | 17 ------- .../app_read_return_terminator.txt | 5 -- doc/CHANGES-staging/app_senddtmf_answer.txt | 5 -- doc/CHANGES-staging/app_signal.txt | 5 -- .../app_voicemail_attachext.txt | 5 -- ...ridge_builtin_features_beep_on_monitor.txt | 12 ----- doc/CHANGES-staging/cdr_ignore.txt | 6 --- .../cli_channel_display_length_increase.txt | 14 ------ doc/CHANGES-staging/db_prefix.txt | 5 -- doc/CHANGES-staging/dundi.txt | 5 -- .../features_bridge_noanswer.txt | 5 -- .../format_sln_support_for_slin.txt | 5 -- doc/CHANGES-staging/func_export.txt | 5 -- doc/CHANGES-staging/func_json_additions.txt | 5 -- doc/CHANGES-staging/func_strings_trim.txt | 5 -- doc/CHANGES-staging/fxo_immediate.txt | 6 --- doc/CHANGES-staging/http_bindaddr.txt | 6 --- doc/CHANGES-staging/lock_deadlock.txt | 5 -- doc/CHANGES-staging/manager_aoc.txt | 3 -- doc/CHANGES-staging/res_geolocation.txt | 49 ------------------- doc/CHANGES-staging/res_hep.txt | 5 -- doc/CHANGES-staging/res_http_media_cache.txt | 12 ----- .../res_musiconhold_answeredonly.txt | 4 -- .../res_phoneprov_multihomed_server.txt | 5 -- .../res_pjsip_100rel_option.txt | 6 --- ...ip_all_codecs_on_empty_reinvite_option.txt | 8 --- doc/CHANGES-staging/res_pjsip_aoc.txt | 4 -- .../res_pjsip_logger_method.txt | 7 --- .../res_pjsip_notify_options.txt | 4 -- doc/CHANGES-staging/res_pjsip_parameters.txt | 5 -- doc/CHANGES-staging/res_pjsip_rfc3329.txt | 6 --- .../res_pjsip_session_overlap.txt | 4 -- .../res_pjsip_tls_cert_key_reload.txt | 5 -- doc/CHANGES-staging/res_pjsip_usereqphone.txt | 4 -- doc/CHANGES-staging/res_rtp_asterisk.txt | 9 ---- doc/CHANGES-staging/res_rtp_rfc3326_sip.txt | 5 -- doc/CHANGES-staging/res_tonedetect_ring.txt | 5 -- doc/CHANGES-staging/test.txt | 11 ----- doc/CHANGES-staging/transfer.txt | 14 ------ doc/CHANGES-staging/xmldoc.txt | 5 -- doc/UPGRADE-staging/README.md | 37 -------------- doc/UPGRADE-staging/app_cdr.txt | 6 --- doc/UPGRADE-staging/app_macro_removal.txt | 37 -------------- doc/UPGRADE-staging/app_osplookup_removal.txt | 6 --- .../app_playback_playbackstatus.txt | 8 --- doc/UPGRADE-staging/chan_alsa_removal.txt | 6 --- doc/UPGRADE-staging/chan_mgcp_removal.txt | 7 --- doc/UPGRADE-staging/chan_sip_removal.txt | 6 --- doc/UPGRADE-staging/chan_skinny_removal.txt | 6 --- .../manager_config_live_dangerously.txt | 8 --- doc/UPGRADE-staging/pbx_builtins.txt | 5 -- .../res_crypto-regular-file-keys.txt | 5 -- doc/UPGRADE-staging/res_monitor_removal.txt | 13 ----- doc/UPGRADE-staging/translate.txt | 6 --- include/asterisk/manager.h | 2 +- res/ari/resource_endpoints.h | 2 - rest-api/resources.json | 2 +- 68 files changed, 2 insertions(+), 545 deletions(-) delete mode 100644 doc/CHANGES-staging/README.md delete mode 100644 doc/CHANGES-staging/ami_hook_flash.txt delete mode 100644 doc/CHANGES-staging/answer.txt delete mode 100644 doc/CHANGES-staging/app_amd.txt delete mode 100644 doc/CHANGES-staging/app_bridgewait.txt delete mode 100644 doc/CHANGES-staging/app_broadcast.txt delete mode 100644 doc/CHANGES-staging/app_confbridge_marked_any.txt delete mode 100644 doc/CHANGES-staging/app_directory_skip_call.txt delete mode 100644 doc/CHANGES-staging/app_if.txt delete mode 100644 doc/CHANGES-staging/app_mixmonitor_clid.txt delete mode 100644 doc/CHANGES-staging/app_mixmonitor_delete.txt delete mode 100644 doc/CHANGES-staging/app_mixmonitor_mute_by_id.txt delete mode 100644 doc/CHANGES-staging/app_read_return_terminator.txt delete mode 100644 doc/CHANGES-staging/app_senddtmf_answer.txt delete mode 100644 doc/CHANGES-staging/app_signal.txt delete mode 100644 doc/CHANGES-staging/app_voicemail_attachext.txt delete mode 100644 doc/CHANGES-staging/bridge_builtin_features_beep_on_monitor.txt delete mode 100644 doc/CHANGES-staging/cdr_ignore.txt delete mode 100644 doc/CHANGES-staging/cli_channel_display_length_increase.txt delete mode 100644 doc/CHANGES-staging/db_prefix.txt delete mode 100644 doc/CHANGES-staging/dundi.txt delete mode 100644 doc/CHANGES-staging/features_bridge_noanswer.txt delete mode 100644 doc/CHANGES-staging/format_sln_support_for_slin.txt delete mode 100644 doc/CHANGES-staging/func_export.txt delete mode 100644 doc/CHANGES-staging/func_json_additions.txt delete mode 100644 doc/CHANGES-staging/func_strings_trim.txt delete mode 100644 doc/CHANGES-staging/fxo_immediate.txt delete mode 100644 doc/CHANGES-staging/http_bindaddr.txt delete mode 100644 doc/CHANGES-staging/lock_deadlock.txt delete mode 100644 doc/CHANGES-staging/manager_aoc.txt delete mode 100644 doc/CHANGES-staging/res_geolocation.txt delete mode 100644 doc/CHANGES-staging/res_hep.txt delete mode 100644 doc/CHANGES-staging/res_http_media_cache.txt delete mode 100644 doc/CHANGES-staging/res_musiconhold_answeredonly.txt delete mode 100644 doc/CHANGES-staging/res_phoneprov_multihomed_server.txt delete mode 100644 doc/CHANGES-staging/res_pjsip_100rel_option.txt delete mode 100644 doc/CHANGES-staging/res_pjsip_all_codecs_on_empty_reinvite_option.txt delete mode 100644 doc/CHANGES-staging/res_pjsip_aoc.txt delete mode 100644 doc/CHANGES-staging/res_pjsip_logger_method.txt delete mode 100644 doc/CHANGES-staging/res_pjsip_notify_options.txt delete mode 100644 doc/CHANGES-staging/res_pjsip_parameters.txt delete mode 100644 doc/CHANGES-staging/res_pjsip_rfc3329.txt delete mode 100644 doc/CHANGES-staging/res_pjsip_session_overlap.txt delete mode 100644 doc/CHANGES-staging/res_pjsip_tls_cert_key_reload.txt delete mode 100644 doc/CHANGES-staging/res_pjsip_usereqphone.txt delete mode 100644 doc/CHANGES-staging/res_rtp_asterisk.txt delete mode 100644 doc/CHANGES-staging/res_rtp_rfc3326_sip.txt delete mode 100644 doc/CHANGES-staging/res_tonedetect_ring.txt delete mode 100644 doc/CHANGES-staging/test.txt delete mode 100644 doc/CHANGES-staging/transfer.txt delete mode 100644 doc/CHANGES-staging/xmldoc.txt delete mode 100644 doc/UPGRADE-staging/README.md delete mode 100644 doc/UPGRADE-staging/app_cdr.txt delete mode 100644 doc/UPGRADE-staging/app_macro_removal.txt delete mode 100644 doc/UPGRADE-staging/app_osplookup_removal.txt delete mode 100644 doc/UPGRADE-staging/app_playback_playbackstatus.txt delete mode 100644 doc/UPGRADE-staging/chan_alsa_removal.txt delete mode 100644 doc/UPGRADE-staging/chan_mgcp_removal.txt delete mode 100644 doc/UPGRADE-staging/chan_sip_removal.txt delete mode 100644 doc/UPGRADE-staging/chan_skinny_removal.txt delete mode 100644 doc/UPGRADE-staging/manager_config_live_dangerously.txt delete mode 100644 doc/UPGRADE-staging/pbx_builtins.txt delete mode 100644 doc/UPGRADE-staging/res_crypto-regular-file-keys.txt delete mode 100644 doc/UPGRADE-staging/res_monitor_removal.txt delete mode 100644 doc/UPGRADE-staging/translate.txt diff --git a/doc/CHANGES-staging/README.md b/doc/CHANGES-staging/README.md deleted file mode 100644 index 8fc51bce28..0000000000 --- a/doc/CHANGES-staging/README.md +++ /dev/null @@ -1,37 +0,0 @@ -## **DO NOT REMOVE THIS FILE!** - -The only files that should be added to this directory are ones that will be -used by the release script to update the CHANGES file automatically. The only -time that it is necessary to add something to the CHANGES-staging directory is -if you are either adding a new feature to Asterisk or adding new functionality -to an existing feature. The file does not need to have a meaningful name, but -it probably should. If there are multiple items that need documenting, you can -add multiple files, each with their own description. If the message is going to -be the same for each subject, then you can add multiple subject headers to one -file. The "Subject: xxx" line is case sensitive! For example, if you are making -a change to PJSIP, then you might add the file "res_pjsip_my_cool_feature.txt" to -this directory, with a short description of what it does. The files must have -the ".txt" suffix. If you are adding multiple entries, they should be done in -the same commit to avoid merge conflicts. Here's an example: - -> Subject: res_pjsip -> Subject: Core -> -> Here's a pretty good description of my new feature that explains exactly what -> it does and how to use it. - -Here's a master-only example: - -> Subject: res_ari -> Master-Only: True -> -> This change will only go into the master branch. The "Master-Only" header -> will never be in a change not in master. - -Note that the second subject has another header: "Master-Only". Changes that go -into the master branch and ONLY the master branch are the only ones that should -have this header. Also, the value can only be "true" or "True". The -"Master-Only" part of the header IS case-sensitive, however! - -For more information, check out the wiki page: -https://wiki.asterisk.org/wiki/display/AST/CHANGES+and+UPGRADE.txt diff --git a/doc/CHANGES-staging/ami_hook_flash.txt b/doc/CHANGES-staging/ami_hook_flash.txt deleted file mode 100644 index 5bf1e3455e..0000000000 --- a/doc/CHANGES-staging/ami_hook_flash.txt +++ /dev/null @@ -1,4 +0,0 @@ -Subject: app_senddtmf - -The SendFlash AMI action now allows sending -a hook flash event on a channel. diff --git a/doc/CHANGES-staging/answer.txt b/doc/CHANGES-staging/answer.txt deleted file mode 100644 index 7e047015b8..0000000000 --- a/doc/CHANGES-staging/answer.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: pbx_builtins - -It is now possible to not wait for media on -a channel when answering it using Answer, -by specifying the i option. diff --git a/doc/CHANGES-staging/app_amd.txt b/doc/CHANGES-staging/app_amd.txt deleted file mode 100644 index ffccd8cd48..0000000000 --- a/doc/CHANGES-staging/app_amd.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: app_amd - -An audio file to play during AMD processing can -now be specified to the AMD application or configured -in the amd.conf configuration file. diff --git a/doc/CHANGES-staging/app_bridgewait.txt b/doc/CHANGES-staging/app_bridgewait.txt deleted file mode 100644 index aa2e00be9a..0000000000 --- a/doc/CHANGES-staging/app_bridgewait.txt +++ /dev/null @@ -1,4 +0,0 @@ -Subject: app_bridgewait - -Adds the n option to not answer the channel when -the BridgeWait application is called. diff --git a/doc/CHANGES-staging/app_broadcast.txt b/doc/CHANGES-staging/app_broadcast.txt deleted file mode 100644 index 03e6848362..0000000000 --- a/doc/CHANGES-staging/app_broadcast.txt +++ /dev/null @@ -1,4 +0,0 @@ -Subject: app_broadcast - -A Broadcast application is now available which allows -for asynchronous one-to-many and many-to-one channel audio. diff --git a/doc/CHANGES-staging/app_confbridge_marked_any.txt b/doc/CHANGES-staging/app_confbridge_marked_any.txt deleted file mode 100644 index cbc1bbffdf..0000000000 --- a/doc/CHANGES-staging/app_confbridge_marked_any.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: app_confbridge - -Adds the end_marked_any option which can be used -to kick users from a conference after any -marked user leaves (including marked users). diff --git a/doc/CHANGES-staging/app_directory_skip_call.txt b/doc/CHANGES-staging/app_directory_skip_call.txt deleted file mode 100644 index 83687fe3f9..0000000000 --- a/doc/CHANGES-staging/app_directory_skip_call.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: app_directory - -A new option 's' has been added to the Directory() application that -will skip calling the extension and instead set the extension as -DIRECTORY_EXTEN channel variable. diff --git a/doc/CHANGES-staging/app_if.txt b/doc/CHANGES-staging/app_if.txt deleted file mode 100644 index 855f15aaf6..0000000000 --- a/doc/CHANGES-staging/app_if.txt +++ /dev/null @@ -1,4 +0,0 @@ -Subject: app_if - -Adds the If, ElseIf, Else, EndIf, and ExitIf applications -for conditional execution of a block of code. diff --git a/doc/CHANGES-staging/app_mixmonitor_clid.txt b/doc/CHANGES-staging/app_mixmonitor_clid.txt deleted file mode 100644 index a8331ec673..0000000000 --- a/doc/CHANGES-staging/app_mixmonitor_clid.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: app_mixmonitor - -Adds the c option to use the real Caller ID on -the channel in voicemail recordings as opposed -to the Connected Line. diff --git a/doc/CHANGES-staging/app_mixmonitor_delete.txt b/doc/CHANGES-staging/app_mixmonitor_delete.txt deleted file mode 100644 index 924c9c08a1..0000000000 --- a/doc/CHANGES-staging/app_mixmonitor_delete.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: app_mixmonitor - -The d option for MixMonitor now allows deleting -the original recording when MixMonitor exits, -which can be useful when MixMonitor copies it -somewhere else before exiting. diff --git a/doc/CHANGES-staging/app_mixmonitor_mute_by_id.txt b/doc/CHANGES-staging/app_mixmonitor_mute_by_id.txt deleted file mode 100644 index 958a914ba9..0000000000 --- a/doc/CHANGES-staging/app_mixmonitor_mute_by_id.txt +++ /dev/null @@ -1,17 +0,0 @@ -Subject: app_mixmonitor -Subject: audiohook -Subject: manager - -It is now possible to specify the MixMonitorID when calling -the manager action: MixMonitorMute. This will allow an -individual MixMonitor instance to be muted via ID. - -The MixMonitorID can be stored as a channel variable using -the 'i' MixMonitor option and is returned upon creation if -this option is used. - -As part of this change, if no MixMonitorID is specified in -the manager action MixMonitorMute, Asterisk will set the mute -flag on all MixMonitor audiohooks on the channel. Previous -behavior would set the flag on the first MixMonitor audiohook -found. diff --git a/doc/CHANGES-staging/app_read_return_terminator.txt b/doc/CHANGES-staging/app_read_return_terminator.txt deleted file mode 100644 index 2987f77ea7..0000000000 --- a/doc/CHANGES-staging/app_read_return_terminator.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: app_read - -A new option 'e' has been added to allow Read() to return the -terminator as the dialed digits in the case where only the terminator -is entered. diff --git a/doc/CHANGES-staging/app_senddtmf_answer.txt b/doc/CHANGES-staging/app_senddtmf_answer.txt deleted file mode 100644 index 76811e3a7f..0000000000 --- a/doc/CHANGES-staging/app_senddtmf_answer.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: app_senddtmf - -A new option has been added to SendDTMF() which will answer the -specified channel if it is not already up. If no channel is specified, -the current channel will be answered instead. diff --git a/doc/CHANGES-staging/app_signal.txt b/doc/CHANGES-staging/app_signal.txt deleted file mode 100644 index b3b108d821..0000000000 --- a/doc/CHANGES-staging/app_signal.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: app_signal - -Adds Signal and WaitForSignal applications -which can be used for signaling or as a -simple message queue in the dialplan. diff --git a/doc/CHANGES-staging/app_voicemail_attachext.txt b/doc/CHANGES-staging/app_voicemail_attachext.txt deleted file mode 100644 index c56f04a105..0000000000 --- a/doc/CHANGES-staging/app_voicemail_attachext.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: app_voicemail - -The voicemail user option attachextrecs can -now be set to control whether external recordings -trigger voicemail email notifications. diff --git a/doc/CHANGES-staging/bridge_builtin_features_beep_on_monitor.txt b/doc/CHANGES-staging/bridge_builtin_features_beep_on_monitor.txt deleted file mode 100644 index 39bf9a72c0..0000000000 --- a/doc/CHANGES-staging/bridge_builtin_features_beep_on_monitor.txt +++ /dev/null @@ -1,12 +0,0 @@ -Subject: bridge_builtin_features - -Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval) - -Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid -interval in seconds will result in a periodic beep being -played to the monitored channel upon MixMontior/Monitor -feature start. - -If an interval less than 5 seconds is specified, the interval -will default to 5 seconds. If the value is set to an invalid -interval, the default of 15 seconds will be used. diff --git a/doc/CHANGES-staging/cdr_ignore.txt b/doc/CHANGES-staging/cdr_ignore.txt deleted file mode 100644 index e82f40415a..0000000000 --- a/doc/CHANGES-staging/cdr_ignore.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: cdr - -Two new options have been added which allow -bridging and dial state changes to be ignored -in CDRs, which can be useful if a single CDR -is desired for a channel. diff --git a/doc/CHANGES-staging/cli_channel_display_length_increase.txt b/doc/CHANGES-staging/cli_channel_display_length_increase.txt deleted file mode 100644 index d9fc77cd88..0000000000 --- a/doc/CHANGES-staging/cli_channel_display_length_increase.txt +++ /dev/null @@ -1,14 +0,0 @@ -Subject: cli -Subject: core - -This change increases the display width on 'core show channels' -amd 'core show channels verbose' - -For 'core show channels', the Channel name field is increased to -64 characters and the Location name field is increased to 32 -characters. - -For 'core show channels verbose', the Channel name field is -increased to 80 characters, the Context is increased to 24 -characters and the Extension is increased to 24 characters. - diff --git a/doc/CHANGES-staging/db_prefix.txt b/doc/CHANGES-staging/db_prefix.txt deleted file mode 100644 index 41268155c3..0000000000 --- a/doc/CHANGES-staging/db_prefix.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: db - -The DBPrefixGet AMI action now allows retrieving -all of the DB keys beginning with a particular -prefix. diff --git a/doc/CHANGES-staging/dundi.txt b/doc/CHANGES-staging/dundi.txt deleted file mode 100644 index e71f726743..0000000000 --- a/doc/CHANGES-staging/dundi.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: DUNDi - -DUNDi now supports chan_pjsip. Outgoing calls using -PJSIP require the pjsip_outgoing_endpoint option -to be set in dundi.conf. diff --git a/doc/CHANGES-staging/features_bridge_noanswer.txt b/doc/CHANGES-staging/features_bridge_noanswer.txt deleted file mode 100644 index 7399ad142b..0000000000 --- a/doc/CHANGES-staging/features_bridge_noanswer.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: features - -The Bridge application now has the n "no answer" option -that can be used to prevent the channel from being -automatically answered prior to bridging. diff --git a/doc/CHANGES-staging/format_sln_support_for_slin.txt b/doc/CHANGES-staging/format_sln_support_for_slin.txt deleted file mode 100644 index 3d66536ccd..0000000000 --- a/doc/CHANGES-staging/format_sln_support_for_slin.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: format_sln - -format_sln now recognizes '.slin' as a valid -file extension in addition to the existing -'.sln' and '.raw'. diff --git a/doc/CHANGES-staging/func_export.txt b/doc/CHANGES-staging/func_export.txt deleted file mode 100644 index d9f95c0123..0000000000 --- a/doc/CHANGES-staging/func_export.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: New EXPORT function - -A new function, EXPORT, allows writing variables -and functions on other channels, the complement -of the IMPORT function. diff --git a/doc/CHANGES-staging/func_json_additions.txt b/doc/CHANGES-staging/func_json_additions.txt deleted file mode 100644 index 963f0b18e2..0000000000 --- a/doc/CHANGES-staging/func_json_additions.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: func_json - -Additional parsing capabilities have been added to the -JSON_DECODE function, including support for arrays -and recursive indexing. diff --git a/doc/CHANGES-staging/func_strings_trim.txt b/doc/CHANGES-staging/func_strings_trim.txt deleted file mode 100644 index ade7a4b61e..0000000000 --- a/doc/CHANGES-staging/func_strings_trim.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: func_strings - -Three new functions, TRIM, LTRIM, and RTRIM, are -now available for trimming leading and trailing -whitespace. diff --git a/doc/CHANGES-staging/fxo_immediate.txt b/doc/CHANGES-staging/fxo_immediate.txt deleted file mode 100644 index 01f9ec5eb5..0000000000 --- a/doc/CHANGES-staging/fxo_immediate.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: chan_dahdi - -FXO channels (FXS signaled) that don't use callerid or -distinctive ring detection can now be configured -to enter the dialplan immediately using immediate=yes, -instead of waiting for at least one ring. diff --git a/doc/CHANGES-staging/http_bindaddr.txt b/doc/CHANGES-staging/http_bindaddr.txt deleted file mode 100644 index e83312bc7f..0000000000 --- a/doc/CHANGES-staging/http_bindaddr.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: http -Master-Only: True - -For bound addresses, the HTTP status page now combines the bound -address and bound port in a single line. Additionally, the SSL bind -address has been renamed to TLS. diff --git a/doc/CHANGES-staging/lock_deadlock.txt b/doc/CHANGES-staging/lock_deadlock.txt deleted file mode 100644 index 5fad3ee078..0000000000 --- a/doc/CHANGES-staging/lock_deadlock.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: locks - -A new AMI event, DeadlockStart, is now available -when Asterisk is compiled with DETECT_DEADLOCKS, -and can indicate that a deadlock has occured. diff --git a/doc/CHANGES-staging/manager_aoc.txt b/doc/CHANGES-staging/manager_aoc.txt deleted file mode 100644 index ed90cf609f..0000000000 --- a/doc/CHANGES-staging/manager_aoc.txt +++ /dev/null @@ -1,3 +0,0 @@ -Subject: AMI - -The AOCMessage action can now be used to generate AOC-S messages. diff --git a/doc/CHANGES-staging/res_geolocation.txt b/doc/CHANGES-staging/res_geolocation.txt deleted file mode 100644 index 4d290ba94f..0000000000 --- a/doc/CHANGES-staging/res_geolocation.txt +++ /dev/null @@ -1,49 +0,0 @@ -Subject: res_geolocation - -* Added processing for the 'confidence' element. -* Added documentation to some APIs. -* removed a lot of complex code related to the very-off-nominal - case of needing to process multiple location info sources. -* Create a new 'ast_geoloc_eprofile_to_pidf' API that just takes - one eprofile instead of a datastore of multiples. -* Plugged a huge leak in XML processing that arose from - insufficient documentation by the libxml/libxslt authors. -* Refactored stylesheets to be more efficient. -* Renamed 'profile_action' to 'profile_precedence' to better - reflect it's purpose. -* Added the config option for 'allow_routing_use' which - sets the value of the 'Geolocation-Routing' header. -* Removed the GeolocProfileCreate and GeolocProfileDelete - dialplan apps. -* Changed the GEOLOC_PROFILE dialplan function as follows: - * Removed the 'profile' argument. - * Automatically create a profile if it doesn't exist. - * Delete a profile if 'inheritable' is set to no. -* Fixed various bugs and leaks -* Updated Asterisk WiKi documentation. - -Added 4 built-in profiles: - "" - "" - "" - "" -The profiles are empty except for having their precedence -set. - -Added profile parameter "suppress_empty_ca_elements" that -will cause Civic Address elements that are empty to be -suppressed from the outgoing PIDF-LO document. - -You can now specify the location object's format, location_info, -method, location_source and confidence parameters directly on -a profile object for simple scenarios where the location -information isn't common with any other profiles. This is -mutually exclusive with setting location_reference on the -profile. - -Added an 'a' option to the GEOLOC_PROFILE function to allow -variable lists like location_info_refinement to be appended -to instead of replacing the entire list. - -Added an 'r' option to the GEOLOC_PROFILE function to resolve all -variables before a read operation and after a Set operation. diff --git a/doc/CHANGES-staging/res_hep.txt b/doc/CHANGES-staging/res_hep.txt deleted file mode 100644 index fb386a11d4..0000000000 --- a/doc/CHANGES-staging/res_hep.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: Add support for named capture agent. - -A name for the capture agent can now be specified -using the capture_name option which, if specified, -will be sent to the HEP server. diff --git a/doc/CHANGES-staging/res_http_media_cache.txt b/doc/CHANGES-staging/res_http_media_cache.txt deleted file mode 100644 index 79223c0339..0000000000 --- a/doc/CHANGES-staging/res_http_media_cache.txt +++ /dev/null @@ -1,12 +0,0 @@ -Subject: res_http_media_cache - -The res_http_media_cache module now attempts to load -configuration from the res_http_media_cache.conf file. -The following options were added: - * timeout_secs - * user_agent - * follow_location - * max_redirects - * protocols - * redirect_protocols - * dns_cache_timeout_secs diff --git a/doc/CHANGES-staging/res_musiconhold_answeredonly.txt b/doc/CHANGES-staging/res_musiconhold_answeredonly.txt deleted file mode 100644 index c335184c22..0000000000 --- a/doc/CHANGES-staging/res_musiconhold_answeredonly.txt +++ /dev/null @@ -1,4 +0,0 @@ -Subject: res_musiconhold_answeredonly - -This change adds an option, answeredonly, that will prevent music -on hold on channels that are not answered. diff --git a/doc/CHANGES-staging/res_phoneprov_multihomed_server.txt b/doc/CHANGES-staging/res_phoneprov_multihomed_server.txt deleted file mode 100644 index ff68014570..0000000000 --- a/doc/CHANGES-staging/res_phoneprov_multihomed_server.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: res_phoneprov - -On multihomed Asterisk servers with dynamic SERVER template variables, -reloading this module is no longer required when re-provisioning your -phone to another interface address (e.g. when moving between VLANs.) diff --git a/doc/CHANGES-staging/res_pjsip_100rel_option.txt b/doc/CHANGES-staging/res_pjsip_100rel_option.txt deleted file mode 100644 index 81af80ac03..0000000000 --- a/doc/CHANGES-staging/res_pjsip_100rel_option.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: res_pjsip - -A new option named "peer_supported" has been added to the endpoint option -100rel. When set to this option, Asterisk sends provisional responses -reliably if the peer supports it. If the peer does not support reliable -provisional responses, Asterisk sends them normally. diff --git a/doc/CHANGES-staging/res_pjsip_all_codecs_on_empty_reinvite_option.txt b/doc/CHANGES-staging/res_pjsip_all_codecs_on_empty_reinvite_option.txt deleted file mode 100644 index 99eccbb512..0000000000 --- a/doc/CHANGES-staging/res_pjsip_all_codecs_on_empty_reinvite_option.txt +++ /dev/null @@ -1,8 +0,0 @@ -Subject: res_pjsip - -A new option named "all_codecs_on_empty_reinvite" has been added to the -global section. When this option is enabled, on reception of a re-INVITE -without SDP, Asterisk will send an SDP offer in the 200 OK response containing -all configured codecs on the endpoint, instead of simply those that have -already been negotiated. RFC 3261 specifies this as a SHOULD requirement. -The default value is "off". \ No newline at end of file diff --git a/doc/CHANGES-staging/res_pjsip_aoc.txt b/doc/CHANGES-staging/res_pjsip_aoc.txt deleted file mode 100644 index 496bd0b385..0000000000 --- a/doc/CHANGES-staging/res_pjsip_aoc.txt +++ /dev/null @@ -1,4 +0,0 @@ -Subject: res_pjsip_aoc - -Added res_pjsip_aoc which gives chan_pjsip the ability to send Advice-of-Charge messages. -A new endpoint option, send_aoc, controls this. diff --git a/doc/CHANGES-staging/res_pjsip_logger_method.txt b/doc/CHANGES-staging/res_pjsip_logger_method.txt deleted file mode 100644 index a1f774edb6..0000000000 --- a/doc/CHANGES-staging/res_pjsip_logger_method.txt +++ /dev/null @@ -1,7 +0,0 @@ -Subject: res_pjsip_logger - -SIP messages can now be filtered by SIP request method -(INVITE, CANCEL, ACK, BYE, REGISTER, OPTION, -SUBSCRIBE, NOTIFY, PUBLISH, INFO, and MESSAGE), -allowing for more granular debugging to be done -in the CLI. This applies to requests but not responses. diff --git a/doc/CHANGES-staging/res_pjsip_notify_options.txt b/doc/CHANGES-staging/res_pjsip_notify_options.txt deleted file mode 100644 index 0a500f67fa..0000000000 --- a/doc/CHANGES-staging/res_pjsip_notify_options.txt +++ /dev/null @@ -1,4 +0,0 @@ -Subject: res_pjsip_notify - -Allows using the config options in pjsip_notify.conf -from AMI actions as with the existing CLI commands. diff --git a/doc/CHANGES-staging/res_pjsip_parameters.txt b/doc/CHANGES-staging/res_pjsip_parameters.txt deleted file mode 100644 index c95b43dedf..0000000000 --- a/doc/CHANGES-staging/res_pjsip_parameters.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: res_pjsip_header_funcs - -The new PJSIP_HEADER_PARAM function now fully supports both -URI and header parameters. Both reading and writing -parameters are supported. diff --git a/doc/CHANGES-staging/res_pjsip_rfc3329.txt b/doc/CHANGES-staging/res_pjsip_rfc3329.txt deleted file mode 100644 index 06510b5661..0000000000 --- a/doc/CHANGES-staging/res_pjsip_rfc3329.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: res_pjsip - -Added options "security_negotiation" and "security_mechanisms" to pjsip -endpoints and registrations. "security_negotiation" can be set to "no" (default) -or "mediasec", and "security_mechanisms" can be a list of comma-separated -security_mechanisms in the form defined by RFC 3329 section 2.2. diff --git a/doc/CHANGES-staging/res_pjsip_session_overlap.txt b/doc/CHANGES-staging/res_pjsip_session_overlap.txt deleted file mode 100644 index 5523f3c086..0000000000 --- a/doc/CHANGES-staging/res_pjsip_session_overlap.txt +++ /dev/null @@ -1,4 +0,0 @@ -Subject: res_pjsip_session - -The overlap_context option now allows explicitly -specifying a context to use for overlap dialing matches. diff --git a/doc/CHANGES-staging/res_pjsip_tls_cert_key_reload.txt b/doc/CHANGES-staging/res_pjsip_tls_cert_key_reload.txt deleted file mode 100644 index 60c9dcd700..0000000000 --- a/doc/CHANGES-staging/res_pjsip_tls_cert_key_reload.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: res_pjsip - -TLS transports in res_pjsip can now reload their TLS certificate -and private key files, provided the filename of them has not -changed. diff --git a/doc/CHANGES-staging/res_pjsip_usereqphone.txt b/doc/CHANGES-staging/res_pjsip_usereqphone.txt deleted file mode 100644 index 01d59a7ad9..0000000000 --- a/doc/CHANGES-staging/res_pjsip_usereqphone.txt +++ /dev/null @@ -1,4 +0,0 @@ -subject: res_pjsip - -user_eq_phone=yes flag on a pjsip endpoint will now set user=phone on -the From and Prviacy headers in addition to the existing To and RURI diff --git a/doc/CHANGES-staging/res_rtp_asterisk.txt b/doc/CHANGES-staging/res_rtp_asterisk.txt deleted file mode 100644 index 9c8e05f0b6..0000000000 --- a/doc/CHANGES-staging/res_rtp_asterisk.txt +++ /dev/null @@ -1,9 +0,0 @@ -Subject: res_rtp_asterisk - -This module has been updated to provide additional -quality statistics in the form of an Asterisk -Media Experience Score. The score is available using -the same mechanisms you'd use to retrieve jitter, loss, -and rtt statistics. For more information about the -score and how to retrieve it, see -https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score diff --git a/doc/CHANGES-staging/res_rtp_rfc3326_sip.txt b/doc/CHANGES-staging/res_rtp_rfc3326_sip.txt deleted file mode 100644 index 62a73925ca..0000000000 --- a/doc/CHANGES-staging/res_rtp_rfc3326_sip.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: res_pjsip_rfc3326 - -Add ability to set HANGUPCAUSE when SIP causecode received in BYE Reason header (in -addition to currently supported Q.850). The first header found will be used to set -the HANGUPCAUSE variable. diff --git a/doc/CHANGES-staging/res_tonedetect_ring.txt b/doc/CHANGES-staging/res_tonedetect_ring.txt deleted file mode 100644 index e5e4c2e232..0000000000 --- a/doc/CHANGES-staging/res_tonedetect_ring.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: res_tonedetect - -The TONE_DETECT function now supports -detection of audible ringback tone -using the p option. diff --git a/doc/CHANGES-staging/test.txt b/doc/CHANGES-staging/test.txt deleted file mode 100644 index 05a2694b31..0000000000 --- a/doc/CHANGES-staging/test.txt +++ /dev/null @@ -1,11 +0,0 @@ -Subject: test.c - -The "tests" attribute of the "testsuite" element in the -output XML now reflects only the tests actually requested -to be executed instead of all the tests registered. - -The "failures" attribute was added to the "testsuite" -element. - -Also added two new unit tests that just pass and fail -to be used for testing CI itself. diff --git a/doc/CHANGES-staging/transfer.txt b/doc/CHANGES-staging/transfer.txt deleted file mode 100644 index 962272fcf7..0000000000 --- a/doc/CHANGES-staging/transfer.txt +++ /dev/null @@ -1,14 +0,0 @@ -Subject: Transfer feature - -The following capabilities have been added to the -transfer feature: - -- The transfer initiation announcement prompt can -now be customized in features.conf. - -- The TRANSFER_EXTEN variable now can be set on the -transferer's channel in order to allow the transfer -function to automatically attempt to go to the extension -contained in this variable, if it exists. The transfer -context behavior is not changed (TRANSFER_CONTEXT is used -if it exists; otherwise the default context is used). diff --git a/doc/CHANGES-staging/xmldoc.txt b/doc/CHANGES-staging/xmldoc.txt deleted file mode 100644 index 50324e46f5..0000000000 --- a/doc/CHANGES-staging/xmldoc.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: xmldocs - -The XML documentation can now be reloaded without restarting -Asterisk, which makes it possible to load new modules that -enforce documentation without restarting Asterisk. diff --git a/doc/UPGRADE-staging/README.md b/doc/UPGRADE-staging/README.md deleted file mode 100644 index 21cbe78c59..0000000000 --- a/doc/UPGRADE-staging/README.md +++ /dev/null @@ -1,37 +0,0 @@ -## **DO NOT REMOVE THIS FILE!** - -The only files that should be added to this directory are ones that will be -used by the release script to update the UPGRADE.txt file automatically. The -only time that it is necessary to add something to the UPGRADE-staging directory -is if you are making a breaking change to an existing feature in Asterisk. The -file does not need to have a meaningful name, but it probably should. If there -are multiple items that need documenting, you can add multiple files, each with -their own description. If the message is going to be the same for each subject, -then you can add multiple subject headers to one file. The "Subject: xxx" line -is case sensitive! For example, if you are making a change to PJSIP, then you -might add the file "res_pjsip_my_cool_feature.txt" to this directory, with a -short description of what it does. The files must have the ".txt" suffix. -If you are adding multiple entries, they should be done in the same commit -to avoid merge conflicts. Here's an example: - -> Subject: res_pjsip -> Subject: Core -> -> Here's a pretty good description of my new feature that explains exactly what -> it does and how to use it. - -Here's a master-only example: - -> Subject: res_ari -> Master-Only: True -> -> This change will only go into the master branch. The "Master-Only" header -> will never be in a change not in master. - -Note that the second subject has another header: "Master-Only". Changes that go -into the master branch and ONLY the master branch are the only ones that should -have this header. Also, the value can only be "true" or "True". The -"Master-Only" part of the header IS case-sensitive, however! - -For more information, check out the wiki page: -https://wiki.asterisk.org/wiki/display/AST/CHANGES+and+UPGRADE.txt diff --git a/doc/UPGRADE-staging/app_cdr.txt b/doc/UPGRADE-staging/app_cdr.txt deleted file mode 100644 index 114f0ad2a2..0000000000 --- a/doc/UPGRADE-staging/app_cdr.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: app_cdr -Master-Only: True - -The previously deprecated NoCDR application has been removed. -Additionally, the previously deprecated 'e' option to the ResetCDR -application has been removed. diff --git a/doc/UPGRADE-staging/app_macro_removal.txt b/doc/UPGRADE-staging/app_macro_removal.txt deleted file mode 100644 index 77f852fa89..0000000000 --- a/doc/UPGRADE-staging/app_macro_removal.txt +++ /dev/null @@ -1,37 +0,0 @@ -Subject: app_macro -Master-Only: True - -This module was deprecated in Asterisk 16 -and is now being removed in accordance with -the Asterisk Module Deprecation policy. - - -For most modules that interacted with app_macro, -this change is limited to no longer looking for -the current context from the macrocontext when set. - -The following modules have additional impacts: - -app_dial - no longer supports M^ connected/redirecting macro - -app_minivm - samples written using macro will no longer work. -The sample needs to be re-written - -app_queue - can no longer call a macro on the called party's -channel. Use gosub which is currently supported - -ccss - no callback macro, gosub only - -app_voicemail - no macro support - -channel - remove macrocontext and priority, no connected -line or redirection macro options - -options - stdexten is deprecated to gosub as the default -and only options - -pbx - removed macrolock - -pbx_dundi - no longer look for macro - -snmp - removed macro context, exten, and priority diff --git a/doc/UPGRADE-staging/app_osplookup_removal.txt b/doc/UPGRADE-staging/app_osplookup_removal.txt deleted file mode 100644 index 6900e4913d..0000000000 --- a/doc/UPGRADE-staging/app_osplookup_removal.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: app_osplookup -Master-Only: True - -This module was deprecated in Asterisk 19 -and is now being removed in accordance with -the Asterisk Module Deprecation policy. diff --git a/doc/UPGRADE-staging/app_playback_playbackstatus.txt b/doc/UPGRADE-staging/app_playback_playbackstatus.txt deleted file mode 100644 index 49302b7966..0000000000 --- a/doc/UPGRADE-staging/app_playback_playbackstatus.txt +++ /dev/null @@ -1,8 +0,0 @@ -Subject: app_playback - -In Asterisk 11, if a channel was redirected away during Playback(), -the PLAYBACKSTATUS variable would be set to SUCCESS. In Asterisk 12 -(specifically commit 7d9871b3940fa50e85039aef6a8fb9870a7615b9) that -behavior was inadvertently changed and the same operation would result -in the PLAYBACKSTATUS variable being set to FAILED. The Asterisk 11 -behavior has been restored. diff --git a/doc/UPGRADE-staging/chan_alsa_removal.txt b/doc/UPGRADE-staging/chan_alsa_removal.txt deleted file mode 100644 index baf91af5eb..0000000000 --- a/doc/UPGRADE-staging/chan_alsa_removal.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: chan_alsa -Master-Only: True - -This module was deprecated in Asterisk 19 -and is now being removed in accordance with -the Asterisk Module Deprecation policy. diff --git a/doc/UPGRADE-staging/chan_mgcp_removal.txt b/doc/UPGRADE-staging/chan_mgcp_removal.txt deleted file mode 100644 index 11bb4863b3..0000000000 --- a/doc/UPGRADE-staging/chan_mgcp_removal.txt +++ /dev/null @@ -1,7 +0,0 @@ -Subject: chan_mgcp -Master-Only: True - -This module was deprecated in Asterisk 19 -and is now being removed in accordance with -the Asterisk Module Deprecation policy. - diff --git a/doc/UPGRADE-staging/chan_sip_removal.txt b/doc/UPGRADE-staging/chan_sip_removal.txt deleted file mode 100644 index d73e8d6571..0000000000 --- a/doc/UPGRADE-staging/chan_sip_removal.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: chan_sip -Master-Only: True - -This module was deprecated in Asterisk 17 -and is now being removed in accordance with -the Asterisk Module Deprecation policy. diff --git a/doc/UPGRADE-staging/chan_skinny_removal.txt b/doc/UPGRADE-staging/chan_skinny_removal.txt deleted file mode 100644 index 2fcc5d1ef1..0000000000 --- a/doc/UPGRADE-staging/chan_skinny_removal.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: chan_skinny -Master-Only: True - -This module was deprecated in Asterisk 19 -and is now being removed in accordance with -the Asterisk Module Deprecation policy. diff --git a/doc/UPGRADE-staging/manager_config_live_dangerously.txt b/doc/UPGRADE-staging/manager_config_live_dangerously.txt deleted file mode 100644 index 56f39f9c8d..0000000000 --- a/doc/UPGRADE-staging/manager_config_live_dangerously.txt +++ /dev/null @@ -1,8 +0,0 @@ -Subject: AMI (Asterisk Manager Interface) - -Previously, GetConfig and UpdateConfig were able to access files outside of -the Asterisk configuration directory. Now this access is put behind the -live_dangerously configuration option in asterisk.conf, which is disabled by -default. If access to configuration files outside of the Asterisk configuation -directory is required via AMI, then the live_dangerously configuration option -must be set to yes. diff --git a/doc/UPGRADE-staging/pbx_builtins.txt b/doc/UPGRADE-staging/pbx_builtins.txt deleted file mode 100644 index 9aeb70c769..0000000000 --- a/doc/UPGRADE-staging/pbx_builtins.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: pbx_builtins -Master-Only: True - -The previously deprecated ImportVar and SetAMAFlags -applications have now been removed. diff --git a/doc/UPGRADE-staging/res_crypto-regular-file-keys.txt b/doc/UPGRADE-staging/res_crypto-regular-file-keys.txt deleted file mode 100644 index a2d8d81da0..0000000000 --- a/doc/UPGRADE-staging/res_crypto-regular-file-keys.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: res_crypto - -In addition to only paying attention to files ending with .key or .pub -in the keys directory, we now also ignore any files which aren't regular -files. diff --git a/doc/UPGRADE-staging/res_monitor_removal.txt b/doc/UPGRADE-staging/res_monitor_removal.txt deleted file mode 100644 index b5c7eee547..0000000000 --- a/doc/UPGRADE-staging/res_monitor_removal.txt +++ /dev/null @@ -1,13 +0,0 @@ -Subject: res_monitor -Master-Only: True - -This module was deprecated in Asterisk 16 -and is now being removed in accordance with -the Asterisk Module Deprecation policy. - -This also removes the 'w' and 'W' options -for app_queue. - -MixMonitor should be default and only option -for all settings that previously used either -Monitor or MixMonitor. diff --git a/doc/UPGRADE-staging/translate.txt b/doc/UPGRADE-staging/translate.txt deleted file mode 100644 index 6b26998976..0000000000 --- a/doc/UPGRADE-staging/translate.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: translate.c -Master-Only: True - -When setting up translation between two codecs the quality was not taken into account, -resulting in suboptimal translation. The quality is now taken into account, -which can reduce the number of translation steps required, and improve the resulting quality. diff --git a/include/asterisk/manager.h b/include/asterisk/manager.h index b531956bbf..89d39ea7b6 100644 --- a/include/asterisk/manager.h +++ b/include/asterisk/manager.h @@ -54,7 +54,7 @@ - \ref manager.c Main manager code file */ -#define AMI_VERSION "10.0.0" +#define AMI_VERSION "11.0.0" #define DEFAULT_MANAGER_PORT 5038 /* Default port for Asterisk management via TCP */ #define DEFAULT_MANAGER_TLS_PORT 5039 /* Default port for Asterisk management via TCP */ diff --git a/res/ari/resource_endpoints.h b/res/ari/resource_endpoints.h index 3c212b93fa..dd15e1b8e4 100644 --- a/res/ari/resource_endpoints.h +++ b/res/ari/resource_endpoints.h @@ -58,7 +58,6 @@ struct ast_ari_endpoints_send_message_args { const char *from; /*! The body of the message */ const char *body; - /*! The "variables" key in the body object holds technology specific key/value pairs to append to the message. These can be interpreted and used by the various resource types; for example, pjsip and sip resource types will add the key/value pairs as SIP headers, */ struct ast_json *variables; }; /*! @@ -150,7 +149,6 @@ struct ast_ari_endpoints_send_message_to_endpoint_args { const char *from; /*! The body of the message */ const char *body; - /*! The "variables" key in the body object holds technology specific key/value pairs to append to the message. These can be interpreted and used by the various resource types; for example, pjsip and sip resource types will add the key/value pairs as SIP headers, */ struct ast_json *variables; }; /*! diff --git a/rest-api/resources.json b/rest-api/resources.json index 147d96bbc1..e223c5e41c 100644 --- a/rest-api/resources.json +++ b/rest-api/resources.json @@ -2,7 +2,7 @@ "_copyright": "Copyright (C) 2012 - 2013, Digium, Inc.", "_author": "David M. Lee, II ", "_svn_revision": "$Revision$", - "apiVersion": "9.0.0", + "apiVersion": "10.0.0", "swaggerVersion": "1.1", "basePath": "http://localhost:8088/ari", "apis": [