From 1f5e6805bfb6c32371e53b375412514afc7a5b63 Mon Sep 17 00:00:00 2001 From: Asterisk Development Team Date: Wed, 15 Jul 2020 08:59:12 -0500 Subject: [PATCH] Update CHANGES and UPGRADE.txt for 18.0.0 --- CHANGES | 378 ++++++++++++++++++ UPGRADE.txt | 120 ++++++ doc/CHANGES-staging/ACN_streams.txt | 44 -- doc/CHANGES-staging/ARI.txt | 10 - .../ami_sendtext_content_type.txt | 4 - ...app_bridgeaddchan_add_bridgeresult_var.txt | 7 - .../app_chanisavail_empty_device_list.txt | 5 - .../app_confbridge_maximum_sample_rate.txt | 5 - .../app_dial_empty_dial_list.txt | 6 - doc/CHANGES-staging/app_mixmonitor_wav49.txt | 8 - .../app_page_empty_page_list.txt | 5 - .../ari-bridges-inhibit-colp.txt | 5 - .../ari_create_with_variables.txt | 5 - .../chan_pjsip_moh_passthrough.txt | 5 - doc/CHANGES-staging/chan_rtp.txt | 7 - .../confbridge_text_messaging.txt | 7 - doc/CHANGES-staging/feat_audiosocket.txt | 14 - doc/CHANGES-staging/func_curl.txt | 5 - doc/CHANGES-staging/func_curl_headers.txt | 6 - .../func_jitterbuffer_video.txt | 6 - doc/CHANGES-staging/func_volume.txt | 3 - doc/CHANGES-staging/h265-passthrough.txt | 6 - .../hide_messaging_ami_events.txt | 11 - doc/CHANGES-staging/http.txt | 4 - doc/CHANGES-staging/minmemfree.txt | 5 - doc/CHANGES-staging/mixmonitor-s-option.txt | 7 - doc/CHANGES-staging/moh-playlist.txt | 5 - .../pjsip_logger_improvements.txt | 21 - doc/CHANGES-staging/res_musiconhold.txt | 7 - ..._pjsip_add_disable_rport_system_config.txt | 9 - .../res_pjsip_call_offer_pref.txt | 8 - ...jsip_endpoint_identifier_ip_match_port.txt | 8 - .../res_pjsip_session_codecs.txt | 8 - doc/CHANGES-staging/res_rtp_asterisk_cli.txt | 18 - .../res_rtp_asterisk_show_settings_cli.txt | 7 - ...orcery_memory_cache_full_expire_object.txt | 19 - doc/CHANGES-staging/stir_shaken.txt | 20 - .../taskprocessor-like-support.txt | 6 - .../taskprocessor-reset-stats.txt | 7 - .../voicemail_lock_cleanup_revert.txt | 10 - doc/UPGRADE-staging/ACN_streams.txt | 5 - doc/UPGRADE-staging/AMI-Originate.txt | 5 - ...app_bridgeaddchan_add_bridgeresult_var.txt | 7 - .../app_mixmonitor_sync_default.txt | 10 - .../app_queue_consistent_general.txt | 6 - .../app_queue_remove_reason.txt | 5 - doc/UPGRADE-staging/ari_messaging.txt | 26 -- .../res_stir_shaken_directory.txt | 5 - .../stream_immutable_formats.txt | 9 - .../vm_deprecated_removals.txt | 7 - 50 files changed, 498 insertions(+), 428 deletions(-) delete mode 100644 doc/CHANGES-staging/ACN_streams.txt delete mode 100644 doc/CHANGES-staging/ARI.txt delete mode 100644 doc/CHANGES-staging/ami_sendtext_content_type.txt delete mode 100644 doc/CHANGES-staging/app_bridgeaddchan_add_bridgeresult_var.txt delete mode 100644 doc/CHANGES-staging/app_chanisavail_empty_device_list.txt delete mode 100644 doc/CHANGES-staging/app_confbridge_maximum_sample_rate.txt delete mode 100644 doc/CHANGES-staging/app_dial_empty_dial_list.txt delete mode 100644 doc/CHANGES-staging/app_mixmonitor_wav49.txt delete mode 100644 doc/CHANGES-staging/app_page_empty_page_list.txt delete mode 100644 doc/CHANGES-staging/ari-bridges-inhibit-colp.txt delete mode 100644 doc/CHANGES-staging/ari_create_with_variables.txt delete mode 100644 doc/CHANGES-staging/chan_pjsip_moh_passthrough.txt delete mode 100644 doc/CHANGES-staging/chan_rtp.txt delete mode 100644 doc/CHANGES-staging/confbridge_text_messaging.txt delete mode 100644 doc/CHANGES-staging/feat_audiosocket.txt delete mode 100644 doc/CHANGES-staging/func_curl.txt delete mode 100644 doc/CHANGES-staging/func_curl_headers.txt delete mode 100644 doc/CHANGES-staging/func_jitterbuffer_video.txt delete mode 100644 doc/CHANGES-staging/func_volume.txt delete mode 100644 doc/CHANGES-staging/h265-passthrough.txt delete mode 100644 doc/CHANGES-staging/hide_messaging_ami_events.txt delete mode 100644 doc/CHANGES-staging/http.txt delete mode 100644 doc/CHANGES-staging/minmemfree.txt delete mode 100644 doc/CHANGES-staging/mixmonitor-s-option.txt delete mode 100644 doc/CHANGES-staging/moh-playlist.txt delete mode 100644 doc/CHANGES-staging/pjsip_logger_improvements.txt delete mode 100644 doc/CHANGES-staging/res_musiconhold.txt delete mode 100644 doc/CHANGES-staging/res_pjsip_add_disable_rport_system_config.txt delete mode 100644 doc/CHANGES-staging/res_pjsip_call_offer_pref.txt delete mode 100644 doc/CHANGES-staging/res_pjsip_endpoint_identifier_ip_match_port.txt delete mode 100644 doc/CHANGES-staging/res_pjsip_session_codecs.txt delete mode 100644 doc/CHANGES-staging/res_rtp_asterisk_cli.txt delete mode 100644 doc/CHANGES-staging/res_rtp_asterisk_show_settings_cli.txt delete mode 100644 doc/CHANGES-staging/sorcery_memory_cache_full_expire_object.txt delete mode 100644 doc/CHANGES-staging/stir_shaken.txt delete mode 100644 doc/CHANGES-staging/taskprocessor-like-support.txt delete mode 100644 doc/CHANGES-staging/taskprocessor-reset-stats.txt delete mode 100644 doc/CHANGES-staging/voicemail_lock_cleanup_revert.txt delete mode 100644 doc/UPGRADE-staging/ACN_streams.txt delete mode 100644 doc/UPGRADE-staging/AMI-Originate.txt delete mode 100644 doc/UPGRADE-staging/app_bridgeaddchan_add_bridgeresult_var.txt delete mode 100644 doc/UPGRADE-staging/app_mixmonitor_sync_default.txt delete mode 100644 doc/UPGRADE-staging/app_queue_consistent_general.txt delete mode 100644 doc/UPGRADE-staging/app_queue_remove_reason.txt delete mode 100644 doc/UPGRADE-staging/ari_messaging.txt delete mode 100644 doc/UPGRADE-staging/res_stir_shaken_directory.txt delete mode 100644 doc/UPGRADE-staging/stream_immutable_formats.txt delete mode 100644 doc/UPGRADE-staging/vm_deprecated_removals.txt diff --git a/CHANGES b/CHANGES index b59886a917..e05bfe65a7 100644 --- a/CHANGES +++ b/CHANGES @@ -12,6 +12,384 @@ === ============================================================================== +------------------------------------------------------------------------------ +--- New functionality introduced in Asterisk 18.0.0 -------------------------- +------------------------------------------------------------------------------ + +Core +------------------ + * The Streams API becomes the home for the core ACN capabilities. + These include... + + * Parsing and formatting of codec negotation preferences. + * Resolving pending streams and topologies with those configured + using configured preferences. + * Utility functions for creating string representations of + streams, topologies, and negotiation preferences. + + For codec negotiation preferences: + * Added ast_stream_codec_prefs_parse() which takes a string + representation of codec negotiation preferences, which + may come from a pjsip endpoint for example, and populates + a ast_stream_codec_negotiation_prefs structure. + * Added ast_stream_codec_prefs_to_str() which does the reverse. + * Added many functions to parse individual parameter name + and value strings to their respectrive enum values, and the + reverse. + + For streams: + * Added ast_stream_create_resolved() which takes a "live" stream + and resolves it with a configured stream and the negotiation + preferences to create a new stream. + * Added ast_stream_to_str() which create a string representation + of a stream suitable for debug or display purposes. + + For topology: + * Added ast_stream_topology_create_resolved() which takes a "live" + topology and resolves it, stream by stream, with a configured + topology stream and the negotiation preferences to create a new + topology. + * Added ast_stream_topology_to_str() which create a string + representation of a topology suitable for debug or display + purposes. + * Renamed ast_format_caps_from_topology() to + ast_stream_topology_get_formats() to be more consistent with + the existing ast_stream_get_formats(). + + Additional changes: + * A new function ast_format_cap_append_names() appends the results + to the ast_str buffer instead of replacing buffer contents. + +app_bridgeaddchan +------------------ + * The BridgeAdd application now behaves more like the Bridge application. + The application now sets the BRIDGERESULT channel variable to indicate + what happened when the channel resumes in dialplan. This is instead of + hanging up the channel on failure conditions. + +res_pjsip +------------------ + * Two new options, incoming_call_offer_pref and outgoing_call_offer_pref + have been added to res_pjsip endpoints that specify the preferred order + of codecs to use between those received/sent in an SDP offer and those + set in the endpoint configuration. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 17.0.0 to Asterisk 18.0.0 ------------ +------------------------------------------------------------------------------ + +AMI +------------------ + * You can now specify an optional 'Content-Type' as an argument for the Asterisk + SendText manager action. + +ARI +------------------ + * A new parameter 'inhibitConnectedLineUpdates' is now available in the + 'bridges.addChannel' call. This prevents the identity of the newly connected + channel from being presented to other bridge members. + +ARI Channels +------------------ + * The Channel resource has a new sub-resource "externalMedia". + This allows an application to create a channel for the sole purpose + of exchanging media with an external server. Once created, this + channel could be placed into a bridge with existing channels to + allow the external server to inject audio into the bridge or + receive audio from the bridge. + See https://wiki.asterisk.org/wiki/display/AST/External+Media+and+ARI + for more information. + +Core +------------------ + * H.265/HEVC is now a supported video codec and it can be used by + specifying "h265" in the allow line. + Please note however, that handling of the additional SDP parameters + described in RFC 7798 section 7.2 is not yet supported. + +Features +------------------ + * Adds support for AudioSocket, a very simple bidirectional audio streaming + protocol. There are both channel and application interfaces. + + A description of the protocol can be found on the referenced wiki page. A + short talk about the reasons and implementation can be found on YouTube at + the link provided. + + ARI support has also been added via the existing "externalMedia" ARI + functionality. The UUID is specified using the arbitrary "data" field. + + Wiki: https://wiki.asterisk.org/wiki/display/AST/AudioSocket + YouTube: https://www.youtube.com/watch?v=tjduXbZZEgI + +Messaging +------------------ + * In order to reduce the amount of AMI and ARI events generated, + the global "Message/ast_msg_queue" channel can be set to suppress + it's normal channel housekeeping events such as "Newexten", + "VarSet", etc. This can greatly reduce load on the manager + and ARI applications when the Digium Phone Module for Asterisk + is in use. To enable, set "hide_messaging_ami_events" in + asterisk.conf to "yes" In Asterisk versions <18, the default + is "no" preserving existing behavior. Beginning with + Asterisk 18, the option will default to "yes". + +STIR/SHAKEN +------------------ + * STIR/SHAKEN support has been added to Asterisk. Configuration is done in + stir_shaken.conf. There is a sample configuration file to help you get + started (asterisk/configs/samples/stir_shaken.conf.sample). Once that's + set up, you can enable STIR/SHAKEN on any endpoint by setting stir_shaken + to yes on the endpoint configuration object. This will add an Identity + header on outgoing INVITEs, and check for an Identity header on incoming + INVITEs. This option has been added to Alembic as well. + + The information received on an incoming INVITE can be checked using the + STIR_SHAKEN dialplan function. There are two variations: + + STIR_SHAKEN(count) + STIR_SHAKEN(0, verify_result) + + The first variation will tell you how many STIR/SHAKEN results are on the + channel. The second fetches information for a specific result. The first + parameter is the index, followed by what information you want to retrieve. + The available options are 'verify_result', 'identity', and 'attestation'. + +app_chanisavail +------------------ + * The ChanIsAvail application now tolerates empty positions in the supplied + device list. Dialplan can now be simplified by not having to check for + empty positions in the device list. + +app_confbridge +------------------ + * A new bridge profile option, maximum_sample_rate, has been added which sets + a maximum sample rate that the bridge will be mixed at. This allows the bridge + to move below the maximum sample rate as needed but caps it at the maximum. + + * A new option, "text_messaging", has been added to the user profile + which allows control over whether text messaging is enabled or + disabled for a user. If enabled (the default) text messages + will be sent to the user. If disabled no text messages will be + sent to the user. + +app_dial +------------------ + * The Dial application now tolerates empty positions in the supplied + destination list. Dialplan can now be simplified by not having to check + for empty positions in the destination list. If there are no endpoints to + dial then DIALSTATUS is set to CHANUNAVAIL. + +app_mixmonitor +------------------ + * An option 'S' has been added to MixMonitor. If used in combination with + the r() and/or t() options, if a frame is available to write to one of + those files but not the other, a frame of silence if written to the file + that does not have an audio frame. This should prevent the two files + from "drifting" when mixed after the fact. + + * If the 'filename' argument to MixMonitor() ended with '.wav49,' + Asterisk would silently convert the extension to '.WAV' when opening + the file for writing. This caused the MIXMONITOR_FILENAME variable to + reference the wrong file. The MIXMONITOR_FILENAME variable will now + reflect the name of the file that Asterisk actually used instead of + the filename that was passed to the application. + +app_page +------------------ + * The Page application now tolerates empty positions in the supplied + destination list. Dialplan can now be simplified by not having to check + for empty positions in the destination list. + +app_voicemail +------------------ + * A feature was added in Asterisk 13.27.0 and 16.4.0 that removed lock files from + the Asterisk voicemail directory on startup. Some users that store their + voicemails on network storage devices experienced slow startup times due to the + relative expense of traversing the voicemail directory structure looking for + orphaned lock files. This feature has now been removed. + + Users who require the lock files to be removed at startup should modify their + startup scripts to do so before starting the asterisk process. + +chan_pjsip +------------------ + * A new dialplan function, PJSIP_MOH_PASSTRHOUGH, has been added to chan_pjsip. This + allows the behaviour of the moh_passthrough endpoint option to be read or changed + in the dialplan. This allows control on a per-call basis. + +chan_rtp +------------------ + * The UnicastRTP channel driver provided by chan_rtp now accepts + ":" as an alternative to ":" in the destination. + The first AAAA (preferred) or A record resolved will be used as the destination. + The lookup is synchronous so beware of possible dialplan delays if you specify a + hostname. + +func_curl +------------------ + * A new parameter, httpheader, has been added to CURLOPT function. This parameter + allows to set custom http headers for subsequent calls off CURL function. + Any setting of headers will replace the default curl headers + (e.g. "Content-type: application/x-www-form-urlencoded") + + * A new option, followlocation, can now be enabled with the CURLOPT() + dialplan function. Setting this will instruct cURL to follow 3xx + redirects, which it does not by default. + +func_jitterbuffer +------------------ + * The JITTERBUFFER dialplan function now has an option to enable video synchronization + support. When enabled and used with a compatible channel driver (chan_sip, chan_pjsip) + the video is buffered according to the size of the audio jitterbuffer and is + synchronized to the audio. + +func_volume +------------------ + * Accept decimal number as argument. + +http +------------------ + * You can now disable the /httpstatus page served by Asterisk's built-in + HTTP server by setting 'enable_status' to 'no' in http.conf. + +minmemfree +------------------ + * The 'minmemfree' configuration option now counts memory allocated to + the filesystem cache as "free" because it is memory that is available + to the process. + +res_ari_channels +------------------ + * When creating a channel in ARI using the create call + you can now specify dialplan variables to be set as part + of the same operation. + +res_musiconhold +------------------ + * This fix allows a realtime moh class to be unregistered from the command + line. This is useful when the contents of a directory referenced by a + realtime moh class have changed. + The realtime moh class is then reloaded on the next request and uses the + new directory contents. + + * A new mode - playlist - has been added to res_musiconhold. This mode allows the + user to specify the files (or URLs) to play explicitly by putting them directly + in musiconhold.conf. + +res_pjsip +------------------ + * Added a new PJSIP system setting called disable_rport. + Default is no to keep support working as before. + + If it is false (default) it adds the 'rport' parameter in the outgoing request message. + If it is true it does not add the 'rport' parameter in the outgoing request message. + + This is a system option, but working as a global option. + +res_pjsip_endpoint_identifier_ip +------------------ + * In 'type = identify' sections, the addresses specified for the 'match' + clause can now include a port number. For IP addresses, the port is + provided by including a colon after the address, followed by the + desired port number. If supplied, the netmask should follow the port + number. To specify a port for IPv6 addresses, the address itself must + be enclosed in brackets to be parsed correctly. + +res_pjsip_logger +------------------ + * The PJSIP packet logger now has the following CLI commands: + + pjsip set logger pcap + + When used this will create a pcap file containing the incoming + and outgoing SIP packets, in unencrypted form. + + pjsip set logger console + + This allows you to toggle logging to console on and off. + + pjsip set logger host add + + This allows you to add an additional IP address or subnet + mask to logging, allowing you to log multiple instead of + just a single IP address or all traffic. + + The normal "pjsip set logger host" CLI command has also been + expanded to allow subnet masks as well. + +res_pjsip_session +------------------ + * When placing an outgoing call to a PJSIP endpoint the intent + of any requested formats will now be respected. If only an audio + format is requested (such as ulaw) but the underlying endpoint + does not support the format the resulting SDP will still only + contain an audio stream, and not any additional streams such as + video. + + * Two new options, incoming_call_offer_pref and outgoing_call_offer_pref + have been added to res_pjsip endpoints that specify the preferred order + of codecs to use between those received/sent in an SDP offer and those + set in the endpoint configuration. + +res_rtp_asterisk +------------------ + * This change include a new cli command 'rtp show settings' + + The command display by general settings of rtp configuration. For this + point is added the fields: rtpstart, rtpend, dtmftimeout, rtpchecksum, + strictrtp, learning_min_sequential and icesupport. + + * The blacklist mechanism in res_rtp_asterisk for ICE and STUN was converted to + an ACL mechanism. + + As such six now options are now available: + + ice_deny + ice_permit + ice_acl + stun_deny + stun_permit + stun_acl + + These options have their obvious meanings as used elsewhere. + + Backwards compatibility was maintained by adding {stun,ice}_blacklist as + aliases for {stun,ice}_deny. + +res_sorcery_memory_cache +------------------ + * The SorceryMemoryCacheExpireObject AMI action and CLI + command allow expiring of a specific object within the + sorcery memory cache. This is done by removing the + object from the cache with the expectation that the + cache will then re-populate the object when it is next + needed. + + For full backend caching this does not occur. The cache + won't repopulate until an entire refresh is done resulting + in the possibility that objects are missing until that + time. + + The AMI action and CLI command will now not allow + expiring of an object if the cache is configured as a + full backend cache. Instead you must use either the + SorceryMemoryCacheExpire or SorceryMemoryCachePopulate + AMI actions or their associated CLI commands. + +taskprocessor.c +------------------ + * Added two new CLI commands to reset stats for taskprocessors. You can + reset stats for a single, specific taskprocessor ('core reset + taskprocessor '), or you can reset all taskprocessors + ('core reset taskprocessors'). These commands will reset the counter for + the number of tasks processed as well as the max queue size. + + * Added "like" support for 'core show taskprocessors'. Now you + can specify a specific set of taskprocessors (or just one) by + adding the keyword "like" to the above command, followed by + your search criteria. + ------------------------------------------------------------------------------ --- New functionality introduced in Asterisk 17.0.0 -------------------------- ------------------------------------------------------------------------------ diff --git a/UPGRADE.txt b/UPGRADE.txt index 68a9e09f2a..68261ae716 100644 --- a/UPGRADE.txt +++ b/UPGRADE.txt @@ -18,6 +18,126 @@ === =========================================================== +------------------------------------------------------------------------------ +--- New functionality introduced in Asterisk 18.0.0 -------------------------- +------------------------------------------------------------------------------ + +Core +------------------ + * The ast_format_cap_from_stream_topology() function has been renamed + to ast_stream_topology_get_formats(). + +app_bridgeaddchan +------------------ + * The BridgeAdd application now behaves more like the Bridge application. + The application now sets the BRIDGERESULT channel variable to indicate + what happened when the channel resumes in dialplan. This is instead of + hanging up the channel on failure conditions. + +app_mixmonitor +------------------ + * In Asterisk 13.29, a new option flag was added to MixMonitor (the 'S' + option) that when combined with the r() or t() options would inject + silence into these files if audio was going to be written to one and + not that other. This allowed the files specified by r() and t() to + subsequently be mixed outside of Asterisk and be appropriately + synchronized. This behavior is now the default, and a new option has + been added to disable this behavior if desired (the 'n' option). + +app_queue +------------------ + * The 'Reason' header in the QueueMemberPause AMI Event has been + removed. The 'PausedReason' header should be used instead. + + * If they are not specified in [general], "shared_lastcall" and "autofill" + now always default to OFF. Before this version, they would be off ('no') if + queues.conf did not have a [general] section, but on ('yes') if it did. + +app_voicemail +------------------ + * The MessageExists dialplan application and the MESSAGE_EXISTS dialplan + function were removed. The were deprecated in Asterisk 1.6.0 and + Asterisk 11.0.0 respectively. The VM_INFO() dialplan function is the + supported mechanism to query the status of a given mailbox. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 17.0.0 to Asterisk 18.0.0 ------------ +------------------------------------------------------------------------------ + +AMI +------------------ + * The AMI Originate action, which optionally takes a dialplan application as + an argument, no longer accepts "Originate" as the application due to + security concerns. + +ARI +------------------ + * The "TextMessageReceived" event used to include a list of "TextMessageVariable" + objects as part of its output. Due to a couple of bugs in Asterisk a list of + received variables was never included even if ones were available. However, + variables set to send would be (which they should have not been), but would + fail validation due to the bad formatting. + + So basically there was no way to get a "TextMessageReceived" event with + variables. Due to this the API has changed. The "TextMessageVariable" object + no longer exists. "TextMessageReceived" now returns a JSON object of key/value + pairs. So for instance instead of a list of "TextMessageVariable" objects: + + [ TextMessageVariable, TextMessageVariable, TextMessageVariable] + + where a TextMessageVariable was supposed to be: + + { "key": "", "value":, "" } + + The output is now just: + + { "": "" } + + This aligns more with how variables are specified when sending a message, as + well as other variable lists in ARI. + +Core +------------------ + * The streams API function ast_stream_get_formats is + now defined as returning the format capabilities const. + This has always been the case but was never enforced + through the API itself. Any consumer of this API that + is not treating the formats as immutable should update + their code to create a new format capabilities and set + it on the stream instead. + +res_stasis +------------------ + * The "TextMessageReceived" event used to include a list of "TextMessageVariable" + objects as part of its output. Due to a couple of bugs in Asterisk a list of + received variables was never included even if ones were available. However, + variables set to send would be (which they should have not been), but would + fail validation due to the bad formatting. + + So basically there was no way to get a "TextMessageReceived" event with + variables. Due to this the API has changed. The "TextMessageVariable" object + no longer exists. "TextMessageReceived" now returns a JSON object of key/value + pairs. So for instance instead of a list of "TextMessageVariable" objects: + + [ TextMessageVariable, TextMessageVariable, TextMessageVariable] + + where a TextMessageVariable was supposed to be: + + { "key": "", "value":, "" } + + The output is now just: + + { "": "" } + + This aligns more with how variables are specified when sending a message, as + well as other variable lists in ARI. + +res_stir_shaken +------------------ + * A new directory has been added under the default (e.g., /var/lib/asterisk) - + inside the 'keys' directory - named 'stir_shaken'. This directory will + hold public keys that have been downloaded for STIR/SHAKEN verification. + ------------------------------------------------------------------------------ --- New functionality introduced in Asterisk 17.0.0 -------------------------- ------------------------------------------------------------------------------ diff --git a/doc/CHANGES-staging/ACN_streams.txt b/doc/CHANGES-staging/ACN_streams.txt deleted file mode 100644 index 6b54d71992..0000000000 --- a/doc/CHANGES-staging/ACN_streams.txt +++ /dev/null @@ -1,44 +0,0 @@ -Subject: Core -Master-Only: True - -The Streams API becomes the home for the core ACN capabilities. -These include... - - * Parsing and formatting of codec negotation preferences. - * Resolving pending streams and topologies with those configured - using configured preferences. - * Utility functions for creating string representations of - streams, topologies, and negotiation preferences. - -For codec negotiation preferences: - * Added ast_stream_codec_prefs_parse() which takes a string - representation of codec negotiation preferences, which - may come from a pjsip endpoint for example, and populates - a ast_stream_codec_negotiation_prefs structure. - * Added ast_stream_codec_prefs_to_str() which does the reverse. - * Added many functions to parse individual parameter name - and value strings to their respectrive enum values, and the - reverse. - -For streams: - * Added ast_stream_create_resolved() which takes a "live" stream - and resolves it with a configured stream and the negotiation - preferences to create a new stream. - * Added ast_stream_to_str() which create a string representation - of a stream suitable for debug or display purposes. - -For topology: - * Added ast_stream_topology_create_resolved() which takes a "live" - topology and resolves it, stream by stream, with a configured - topology stream and the negotiation preferences to create a new - topology. - * Added ast_stream_topology_to_str() which create a string - representation of a topology suitable for debug or display - purposes. - * Renamed ast_format_caps_from_topology() to - ast_stream_topology_get_formats() to be more consistent with - the existing ast_stream_get_formats(). - -Additional changes: - * A new function ast_format_cap_append_names() appends the results - to the ast_str buffer instead of replacing buffer contents. diff --git a/doc/CHANGES-staging/ARI.txt b/doc/CHANGES-staging/ARI.txt deleted file mode 100644 index 06ac4ab0fe..0000000000 --- a/doc/CHANGES-staging/ARI.txt +++ /dev/null @@ -1,10 +0,0 @@ -Subject: ARI Channels - -The Channel resource has a new sub-resource "externalMedia". -This allows an application to create a channel for the sole purpose -of exchanging media with an external server. Once created, this -channel could be placed into a bridge with existing channels to -allow the external server to inject audio into the bridge or -receive audio from the bridge. -See https://wiki.asterisk.org/wiki/display/AST/External+Media+and+ARI -for more information. \ No newline at end of file diff --git a/doc/CHANGES-staging/ami_sendtext_content_type.txt b/doc/CHANGES-staging/ami_sendtext_content_type.txt deleted file mode 100644 index 45037ff7a0..0000000000 --- a/doc/CHANGES-staging/ami_sendtext_content_type.txt +++ /dev/null @@ -1,4 +0,0 @@ -Subject: AMI - -You can now specify an optional 'Content-Type' as an argument for the Asterisk -SendText manager action. diff --git a/doc/CHANGES-staging/app_bridgeaddchan_add_bridgeresult_var.txt b/doc/CHANGES-staging/app_bridgeaddchan_add_bridgeresult_var.txt deleted file mode 100644 index 784e502ebe..0000000000 --- a/doc/CHANGES-staging/app_bridgeaddchan_add_bridgeresult_var.txt +++ /dev/null @@ -1,7 +0,0 @@ -Subject: app_bridgeaddchan -Master-Only: true - -The BridgeAdd application now behaves more like the Bridge application. -The application now sets the BRIDGERESULT channel variable to indicate -what happened when the channel resumes in dialplan. This is instead of -hanging up the channel on failure conditions. diff --git a/doc/CHANGES-staging/app_chanisavail_empty_device_list.txt b/doc/CHANGES-staging/app_chanisavail_empty_device_list.txt deleted file mode 100644 index bf06c3130b..0000000000 --- a/doc/CHANGES-staging/app_chanisavail_empty_device_list.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: app_chanisavail - -The ChanIsAvail application now tolerates empty positions in the supplied -device list. Dialplan can now be simplified by not having to check for -empty positions in the device list. diff --git a/doc/CHANGES-staging/app_confbridge_maximum_sample_rate.txt b/doc/CHANGES-staging/app_confbridge_maximum_sample_rate.txt deleted file mode 100644 index 1c584fa8f0..0000000000 --- a/doc/CHANGES-staging/app_confbridge_maximum_sample_rate.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: app_confbridge - -A new bridge profile option, maximum_sample_rate, has been added which sets -a maximum sample rate that the bridge will be mixed at. This allows the bridge -to move below the maximum sample rate as needed but caps it at the maximum. diff --git a/doc/CHANGES-staging/app_dial_empty_dial_list.txt b/doc/CHANGES-staging/app_dial_empty_dial_list.txt deleted file mode 100644 index dc68ee664a..0000000000 --- a/doc/CHANGES-staging/app_dial_empty_dial_list.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: app_dial - -The Dial application now tolerates empty positions in the supplied -destination list. Dialplan can now be simplified by not having to check -for empty positions in the destination list. If there are no endpoints to -dial then DIALSTATUS is set to CHANUNAVAIL. diff --git a/doc/CHANGES-staging/app_mixmonitor_wav49.txt b/doc/CHANGES-staging/app_mixmonitor_wav49.txt deleted file mode 100644 index f3218d70d2..0000000000 --- a/doc/CHANGES-staging/app_mixmonitor_wav49.txt +++ /dev/null @@ -1,8 +0,0 @@ -Subject: app_mixmonitor - -If the 'filename' argument to MixMonitor() ended with '.wav49,' -Asterisk would silently convert the extension to '.WAV' when opening -the file for writing. This caused the MIXMONITOR_FILENAME variable to -reference the wrong file. The MIXMONITOR_FILENAME variable will now -reflect the name of the file that Asterisk actually used instead of -the filename that was passed to the application. diff --git a/doc/CHANGES-staging/app_page_empty_page_list.txt b/doc/CHANGES-staging/app_page_empty_page_list.txt deleted file mode 100644 index 73e8420e45..0000000000 --- a/doc/CHANGES-staging/app_page_empty_page_list.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: app_page - -The Page application now tolerates empty positions in the supplied -destination list. Dialplan can now be simplified by not having to check -for empty positions in the destination list. diff --git a/doc/CHANGES-staging/ari-bridges-inhibit-colp.txt b/doc/CHANGES-staging/ari-bridges-inhibit-colp.txt deleted file mode 100644 index cdc9ffb80a..0000000000 --- a/doc/CHANGES-staging/ari-bridges-inhibit-colp.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: ARI - -A new parameter 'inhibitConnectedLineUpdates' is now available in the -'bridges.addChannel' call. This prevents the identity of the newly connected -channel from being presented to other bridge members. diff --git a/doc/CHANGES-staging/ari_create_with_variables.txt b/doc/CHANGES-staging/ari_create_with_variables.txt deleted file mode 100644 index a9d28dd4e5..0000000000 --- a/doc/CHANGES-staging/ari_create_with_variables.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: res_ari_channels - -When creating a channel in ARI using the create call -you can now specify dialplan variables to be set as part -of the same operation. diff --git a/doc/CHANGES-staging/chan_pjsip_moh_passthrough.txt b/doc/CHANGES-staging/chan_pjsip_moh_passthrough.txt deleted file mode 100644 index cb874a580a..0000000000 --- a/doc/CHANGES-staging/chan_pjsip_moh_passthrough.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: chan_pjsip - -A new dialplan function, PJSIP_MOH_PASSTRHOUGH, has been added to chan_pjsip. This -allows the behaviour of the moh_passthrough endpoint option to be read or changed -in the dialplan. This allows control on a per-call basis. diff --git a/doc/CHANGES-staging/chan_rtp.txt b/doc/CHANGES-staging/chan_rtp.txt deleted file mode 100644 index ce908ee037..0000000000 --- a/doc/CHANGES-staging/chan_rtp.txt +++ /dev/null @@ -1,7 +0,0 @@ -Subject: chan_rtp - -The UnicastRTP channel driver provided by chan_rtp now accepts -":" as an alternative to ":" in the destination. -The first AAAA (preferred) or A record resolved will be used as the destination. -The lookup is synchronous so beware of possible dialplan delays if you specify a -hostname. diff --git a/doc/CHANGES-staging/confbridge_text_messaging.txt b/doc/CHANGES-staging/confbridge_text_messaging.txt deleted file mode 100644 index 24315afa6f..0000000000 --- a/doc/CHANGES-staging/confbridge_text_messaging.txt +++ /dev/null @@ -1,7 +0,0 @@ -Subject: app_confbridge - -A new option, "text_messaging", has been added to the user profile -which allows control over whether text messaging is enabled or -disabled for a user. If enabled (the default) text messages -will be sent to the user. If disabled no text messages will be -sent to the user. diff --git a/doc/CHANGES-staging/feat_audiosocket.txt b/doc/CHANGES-staging/feat_audiosocket.txt deleted file mode 100644 index cc7b352e0d..0000000000 --- a/doc/CHANGES-staging/feat_audiosocket.txt +++ /dev/null @@ -1,14 +0,0 @@ -Subject: Features - -Adds support for AudioSocket, a very simple bidirectional audio streaming -protocol. There are both channel and application interfaces. - -A description of the protocol can be found on the referenced wiki page. A -short talk about the reasons and implementation can be found on YouTube at -the link provided. - -ARI support has also been added via the existing "externalMedia" ARI -functionality. The UUID is specified using the arbitrary "data" field. - -Wiki: https://wiki.asterisk.org/wiki/display/AST/AudioSocket -YouTube: https://www.youtube.com/watch?v=tjduXbZZEgI diff --git a/doc/CHANGES-staging/func_curl.txt b/doc/CHANGES-staging/func_curl.txt deleted file mode 100644 index fa9ad97462..0000000000 --- a/doc/CHANGES-staging/func_curl.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: func_curl - -A new option, followlocation, can now be enabled with the CURLOPT() -dialplan function. Setting this will instruct cURL to follow 3xx -redirects, which it does not by default. diff --git a/doc/CHANGES-staging/func_curl_headers.txt b/doc/CHANGES-staging/func_curl_headers.txt deleted file mode 100644 index 9d5c8c3e18..0000000000 --- a/doc/CHANGES-staging/func_curl_headers.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: func_curl - -A new parameter, httpheader, has been added to CURLOPT function. This parameter -allows to set custom http headers for subsequent calls off CURL function. -Any setting of headers will replace the default curl headers -(e.g. "Content-type: application/x-www-form-urlencoded") diff --git a/doc/CHANGES-staging/func_jitterbuffer_video.txt b/doc/CHANGES-staging/func_jitterbuffer_video.txt deleted file mode 100644 index 02f9b0b5df..0000000000 --- a/doc/CHANGES-staging/func_jitterbuffer_video.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: func_jitterbuffer - -The JITTERBUFFER dialplan function now has an option to enable video synchronization -support. When enabled and used with a compatible channel driver (chan_sip, chan_pjsip) -the video is buffered according to the size of the audio jitterbuffer and is -synchronized to the audio. diff --git a/doc/CHANGES-staging/func_volume.txt b/doc/CHANGES-staging/func_volume.txt deleted file mode 100644 index e73295b99a..0000000000 --- a/doc/CHANGES-staging/func_volume.txt +++ /dev/null @@ -1,3 +0,0 @@ -Subject: func_volume - -Accept decimal number as argument. diff --git a/doc/CHANGES-staging/h265-passthrough.txt b/doc/CHANGES-staging/h265-passthrough.txt deleted file mode 100644 index b2c449a81b..0000000000 --- a/doc/CHANGES-staging/h265-passthrough.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: Core - -H.265/HEVC is now a supported video codec and it can be used by -specifying "h265" in the allow line. -Please note however, that handling of the additional SDP parameters -described in RFC 7798 section 7.2 is not yet supported. diff --git a/doc/CHANGES-staging/hide_messaging_ami_events.txt b/doc/CHANGES-staging/hide_messaging_ami_events.txt deleted file mode 100644 index 0afbeecdf7..0000000000 --- a/doc/CHANGES-staging/hide_messaging_ami_events.txt +++ /dev/null @@ -1,11 +0,0 @@ -Subject: Messaging - -In order to reduce the amount of AMI and ARI events generated, -the global "Message/ast_msg_queue" channel can be set to suppress -it's normal channel housekeeping events such as "Newexten", -"VarSet", etc. This can greatly reduce load on the manager -and ARI applications when the Digium Phone Module for Asterisk -is in use. To enable, set "hide_messaging_ami_events" in -asterisk.conf to "yes" In Asterisk versions <18, the default -is "no" preserving existing behavior. Beginning with -Asterisk 18, the option will default to "yes". diff --git a/doc/CHANGES-staging/http.txt b/doc/CHANGES-staging/http.txt deleted file mode 100644 index ad778ecf82..0000000000 --- a/doc/CHANGES-staging/http.txt +++ /dev/null @@ -1,4 +0,0 @@ -Subject: http - -You can now disable the /httpstatus page served by Asterisk's built-in -HTTP server by setting 'enable_status' to 'no' in http.conf. diff --git a/doc/CHANGES-staging/minmemfree.txt b/doc/CHANGES-staging/minmemfree.txt deleted file mode 100644 index 5762c15d16..0000000000 --- a/doc/CHANGES-staging/minmemfree.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: minmemfree - -The 'minmemfree' configuration option now counts memory allocated to -the filesystem cache as "free" because it is memory that is available -to the process. diff --git a/doc/CHANGES-staging/mixmonitor-s-option.txt b/doc/CHANGES-staging/mixmonitor-s-option.txt deleted file mode 100644 index d08b86d3fc..0000000000 --- a/doc/CHANGES-staging/mixmonitor-s-option.txt +++ /dev/null @@ -1,7 +0,0 @@ -Subject: app_mixmonitor - -An option 'S' has been added to MixMonitor. If used in combination with -the r() and/or t() options, if a frame is available to write to one of -those files but not the other, a frame of silence if written to the file -that does not have an audio frame. This should prevent the two files -from "drifting" when mixed after the fact. diff --git a/doc/CHANGES-staging/moh-playlist.txt b/doc/CHANGES-staging/moh-playlist.txt deleted file mode 100644 index 14cb0224ac..0000000000 --- a/doc/CHANGES-staging/moh-playlist.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: res_musiconhold - -A new mode - playlist - has been added to res_musiconhold. This mode allows the -user to specify the files (or URLs) to play explicitly by putting them directly -in musiconhold.conf. diff --git a/doc/CHANGES-staging/pjsip_logger_improvements.txt b/doc/CHANGES-staging/pjsip_logger_improvements.txt deleted file mode 100644 index 1a16be9a44..0000000000 --- a/doc/CHANGES-staging/pjsip_logger_improvements.txt +++ /dev/null @@ -1,21 +0,0 @@ -Subject: res_pjsip_logger - -The PJSIP packet logger now has the following CLI commands: - -pjsip set logger pcap - -When used this will create a pcap file containing the incoming -and outgoing SIP packets, in unencrypted form. - -pjsip set logger console - -This allows you to toggle logging to console on and off. - -pjsip set logger host add - -This allows you to add an additional IP address or subnet -mask to logging, allowing you to log multiple instead of -just a single IP address or all traffic. - -The normal "pjsip set logger host" CLI command has also been -expanded to allow subnet masks as well. diff --git a/doc/CHANGES-staging/res_musiconhold.txt b/doc/CHANGES-staging/res_musiconhold.txt deleted file mode 100644 index 47ef397412..0000000000 --- a/doc/CHANGES-staging/res_musiconhold.txt +++ /dev/null @@ -1,7 +0,0 @@ -Subject: res_musiconhold - -This fix allows a realtime moh class to be unregistered from the command -line. This is useful when the contents of a directory referenced by a -realtime moh class have changed. -The realtime moh class is then reloaded on the next request and uses the -new directory contents. diff --git a/doc/CHANGES-staging/res_pjsip_add_disable_rport_system_config.txt b/doc/CHANGES-staging/res_pjsip_add_disable_rport_system_config.txt deleted file mode 100644 index a565e2011e..0000000000 --- a/doc/CHANGES-staging/res_pjsip_add_disable_rport_system_config.txt +++ /dev/null @@ -1,9 +0,0 @@ -Subject: res_pjsip - -Added a new PJSIP system setting called disable_rport. -Default is no to keep support working as before. - -If it is false (default) it adds the 'rport' parameter in the outgoing request message. -If it is true it does not add the 'rport' parameter in the outgoing request message. - -This is a system option, but working as a global option. \ No newline at end of file diff --git a/doc/CHANGES-staging/res_pjsip_call_offer_pref.txt b/doc/CHANGES-staging/res_pjsip_call_offer_pref.txt deleted file mode 100644 index c8b8747c27..0000000000 --- a/doc/CHANGES-staging/res_pjsip_call_offer_pref.txt +++ /dev/null @@ -1,8 +0,0 @@ -Subject: res_pjsip -Subject: res_pjsip_session -Master-Only: True - -Two new options, incoming_call_offer_pref and outgoing_call_offer_pref -have been added to res_pjsip endpoints that specify the preferred order -of codecs to use between those received/sent in an SDP offer and those -set in the endpoint configuration. diff --git a/doc/CHANGES-staging/res_pjsip_endpoint_identifier_ip_match_port.txt b/doc/CHANGES-staging/res_pjsip_endpoint_identifier_ip_match_port.txt deleted file mode 100644 index 3881d64ab1..0000000000 --- a/doc/CHANGES-staging/res_pjsip_endpoint_identifier_ip_match_port.txt +++ /dev/null @@ -1,8 +0,0 @@ -Subject: res_pjsip_endpoint_identifier_ip - -In 'type = identify' sections, the addresses specified for the 'match' -clause can now include a port number. For IP addresses, the port is -provided by including a colon after the address, followed by the -desired port number. If supplied, the netmask should follow the port -number. To specify a port for IPv6 addresses, the address itself must -be enclosed in brackets to be parsed correctly. diff --git a/doc/CHANGES-staging/res_pjsip_session_codecs.txt b/doc/CHANGES-staging/res_pjsip_session_codecs.txt deleted file mode 100644 index 847eb41b32..0000000000 --- a/doc/CHANGES-staging/res_pjsip_session_codecs.txt +++ /dev/null @@ -1,8 +0,0 @@ -Subject: res_pjsip_session - -When placing an outgoing call to a PJSIP endpoint the intent -of any requested formats will now be respected. If only an audio -format is requested (such as ulaw) but the underlying endpoint -does not support the format the resulting SDP will still only -contain an audio stream, and not any additional streams such as -video. diff --git a/doc/CHANGES-staging/res_rtp_asterisk_cli.txt b/doc/CHANGES-staging/res_rtp_asterisk_cli.txt deleted file mode 100644 index 7b5516d06c..0000000000 --- a/doc/CHANGES-staging/res_rtp_asterisk_cli.txt +++ /dev/null @@ -1,18 +0,0 @@ -Subject: res_rtp_asterisk - -The blacklist mechanism in res_rtp_asterisk for ICE and STUN was converted to -an ACL mechanism. - -As such six now options are now available: - -ice_deny -ice_permit -ice_acl -stun_deny -stun_permit -stun_acl - -These options have their obvious meanings as used elsewhere. - -Backwards compatibility was maintained by adding {stun,ice}_blacklist as -aliases for {stun,ice}_deny. diff --git a/doc/CHANGES-staging/res_rtp_asterisk_show_settings_cli.txt b/doc/CHANGES-staging/res_rtp_asterisk_show_settings_cli.txt deleted file mode 100644 index 4f636bbe42..0000000000 --- a/doc/CHANGES-staging/res_rtp_asterisk_show_settings_cli.txt +++ /dev/null @@ -1,7 +0,0 @@ -Subject: res_rtp_asterisk - -This change include a new cli command 'rtp show settings' - -The command display by general settings of rtp configuration. For this -point is added the fields: rtpstart, rtpend, dtmftimeout, rtpchecksum, -strictrtp, learning_min_sequential and icesupport. diff --git a/doc/CHANGES-staging/sorcery_memory_cache_full_expire_object.txt b/doc/CHANGES-staging/sorcery_memory_cache_full_expire_object.txt deleted file mode 100644 index d568e38f16..0000000000 --- a/doc/CHANGES-staging/sorcery_memory_cache_full_expire_object.txt +++ /dev/null @@ -1,19 +0,0 @@ -Subject: res_sorcery_memory_cache - -The SorceryMemoryCacheExpireObject AMI action and CLI -command allow expiring of a specific object within the -sorcery memory cache. This is done by removing the -object from the cache with the expectation that the -cache will then re-populate the object when it is next -needed. - -For full backend caching this does not occur. The cache -won't repopulate until an entire refresh is done resulting -in the possibility that objects are missing until that -time. - -The AMI action and CLI command will now not allow -expiring of an object if the cache is configured as a -full backend cache. Instead you must use either the -SorceryMemoryCacheExpire or SorceryMemoryCachePopulate -AMI actions or their associated CLI commands. diff --git a/doc/CHANGES-staging/stir_shaken.txt b/doc/CHANGES-staging/stir_shaken.txt deleted file mode 100644 index 3ad1784bdf..0000000000 --- a/doc/CHANGES-staging/stir_shaken.txt +++ /dev/null @@ -1,20 +0,0 @@ -Subject: STIR/SHAKEN - -STIR/SHAKEN support has been added to Asterisk. Configuration is done in -stir_shaken.conf. There is a sample configuration file to help you get -started (asterisk/configs/samples/stir_shaken.conf.sample). Once that's -set up, you can enable STIR/SHAKEN on any endpoint by setting stir_shaken -to yes on the endpoint configuration object. This will add an Identity -header on outgoing INVITEs, and check for an Identity header on incoming -INVITEs. This option has been added to Alembic as well. - -The information received on an incoming INVITE can be checked using the -STIR_SHAKEN dialplan function. There are two variations: - -STIR_SHAKEN(count) -STIR_SHAKEN(0, verify_result) - -The first variation will tell you how many STIR/SHAKEN results are on the -channel. The second fetches information for a specific result. The first -parameter is the index, followed by what information you want to retrieve. -The available options are 'verify_result', 'identity', and 'attestation'. diff --git a/doc/CHANGES-staging/taskprocessor-like-support.txt b/doc/CHANGES-staging/taskprocessor-like-support.txt deleted file mode 100644 index 8f61d39e56..0000000000 --- a/doc/CHANGES-staging/taskprocessor-like-support.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: taskprocessor.c - -Added "like" support for 'core show taskprocessors'. Now you -can specify a specific set of taskprocessors (or just one) by -adding the keyword "like" to the above command, followed by -your search criteria. diff --git a/doc/CHANGES-staging/taskprocessor-reset-stats.txt b/doc/CHANGES-staging/taskprocessor-reset-stats.txt deleted file mode 100644 index b5ebb86917..0000000000 --- a/doc/CHANGES-staging/taskprocessor-reset-stats.txt +++ /dev/null @@ -1,7 +0,0 @@ -Subject: taskprocessor.c - -Added two new CLI commands to reset stats for taskprocessors. You can -reset stats for a single, specific taskprocessor ('core reset -taskprocessor '), or you can reset all taskprocessors -('core reset taskprocessors'). These commands will reset the counter for -the number of tasks processed as well as the max queue size. diff --git a/doc/CHANGES-staging/voicemail_lock_cleanup_revert.txt b/doc/CHANGES-staging/voicemail_lock_cleanup_revert.txt deleted file mode 100644 index 500c9a4d7a..0000000000 --- a/doc/CHANGES-staging/voicemail_lock_cleanup_revert.txt +++ /dev/null @@ -1,10 +0,0 @@ -Subject: app_voicemail - -A feature was added in Asterisk 13.27.0 and 16.4.0 that removed lock files from -the Asterisk voicemail directory on startup. Some users that store their -voicemails on network storage devices experienced slow startup times due to the -relative expense of traversing the voicemail directory structure looking for -orphaned lock files. This feature has now been removed. - -Users who require the lock files to be removed at startup should modify their -startup scripts to do so before starting the asterisk process. diff --git a/doc/UPGRADE-staging/ACN_streams.txt b/doc/UPGRADE-staging/ACN_streams.txt deleted file mode 100644 index 400a6d2cdf..0000000000 --- a/doc/UPGRADE-staging/ACN_streams.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: Core -Master-Only: True - -The ast_format_cap_from_stream_topology() function has been renamed -to ast_stream_topology_get_formats(). diff --git a/doc/UPGRADE-staging/AMI-Originate.txt b/doc/UPGRADE-staging/AMI-Originate.txt deleted file mode 100644 index f2d3133098..0000000000 --- a/doc/UPGRADE-staging/AMI-Originate.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: AMI - -The AMI Originate action, which optionally takes a dialplan application as -an argument, no longer accepts "Originate" as the application due to -security concerns. diff --git a/doc/UPGRADE-staging/app_bridgeaddchan_add_bridgeresult_var.txt b/doc/UPGRADE-staging/app_bridgeaddchan_add_bridgeresult_var.txt deleted file mode 100644 index 784e502ebe..0000000000 --- a/doc/UPGRADE-staging/app_bridgeaddchan_add_bridgeresult_var.txt +++ /dev/null @@ -1,7 +0,0 @@ -Subject: app_bridgeaddchan -Master-Only: true - -The BridgeAdd application now behaves more like the Bridge application. -The application now sets the BRIDGERESULT channel variable to indicate -what happened when the channel resumes in dialplan. This is instead of -hanging up the channel on failure conditions. diff --git a/doc/UPGRADE-staging/app_mixmonitor_sync_default.txt b/doc/UPGRADE-staging/app_mixmonitor_sync_default.txt deleted file mode 100644 index 33a55e1673..0000000000 --- a/doc/UPGRADE-staging/app_mixmonitor_sync_default.txt +++ /dev/null @@ -1,10 +0,0 @@ -Subject: app_mixmonitor -Master-Only: true - -In Asterisk 13.29, a new option flag was added to MixMonitor (the 'S' -option) that when combined with the r() or t() options would inject -silence into these files if audio was going to be written to one and -not that other. This allowed the files specified by r() and t() to -subsequently be mixed outside of Asterisk and be appropriately -synchronized. This behavior is now the default, and a new option has -been added to disable this behavior if desired (the 'n' option). diff --git a/doc/UPGRADE-staging/app_queue_consistent_general.txt b/doc/UPGRADE-staging/app_queue_consistent_general.txt deleted file mode 100644 index 0a98bf9dc2..0000000000 --- a/doc/UPGRADE-staging/app_queue_consistent_general.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: app_queue -Master-Only: true - -If they are not specified in [general], "shared_lastcall" and "autofill" -now always default to OFF. Before this version, they would be off ('no') if -queues.conf did not have a [general] section, but on ('yes') if it did. diff --git a/doc/UPGRADE-staging/app_queue_remove_reason.txt b/doc/UPGRADE-staging/app_queue_remove_reason.txt deleted file mode 100644 index e333e2ce83..0000000000 --- a/doc/UPGRADE-staging/app_queue_remove_reason.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: app_queue -Master-Only: True - -The 'Reason' header in the QueueMemberPause AMI Event has been -removed. The 'PausedReason' header should be used instead. diff --git a/doc/UPGRADE-staging/ari_messaging.txt b/doc/UPGRADE-staging/ari_messaging.txt deleted file mode 100644 index 199a8a2382..0000000000 --- a/doc/UPGRADE-staging/ari_messaging.txt +++ /dev/null @@ -1,26 +0,0 @@ -Subject: ARI -Subject: res_stasis - -The "TextMessageReceived" event used to include a list of "TextMessageVariable" -objects as part of its output. Due to a couple of bugs in Asterisk a list of -received variables was never included even if ones were available. However, -variables set to send would be (which they should have not been), but would -fail validation due to the bad formatting. - -So basically there was no way to get a "TextMessageReceived" event with -variables. Due to this the API has changed. The "TextMessageVariable" object -no longer exists. "TextMessageReceived" now returns a JSON object of key/value -pairs. So for instance instead of a list of "TextMessageVariable" objects: - -[ TextMessageVariable, TextMessageVariable, TextMessageVariable] - -where a TextMessageVariable was supposed to be: - -{ "key": "", "value":, "" } - -The output is now just: - -{ "": "" } - -This aligns more with how variables are specified when sending a message, as -well as other variable lists in ARI. diff --git a/doc/UPGRADE-staging/res_stir_shaken_directory.txt b/doc/UPGRADE-staging/res_stir_shaken_directory.txt deleted file mode 100644 index 160241e742..0000000000 --- a/doc/UPGRADE-staging/res_stir_shaken_directory.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: res_stir_shaken - -A new directory has been added under the default (e.g., /var/lib/asterisk) - -inside the 'keys' directory - named 'stir_shaken'. This directory will -hold public keys that have been downloaded for STIR/SHAKEN verification. diff --git a/doc/UPGRADE-staging/stream_immutable_formats.txt b/doc/UPGRADE-staging/stream_immutable_formats.txt deleted file mode 100644 index 2af5d9b271..0000000000 --- a/doc/UPGRADE-staging/stream_immutable_formats.txt +++ /dev/null @@ -1,9 +0,0 @@ -Subject: Core - -The streams API function ast_stream_get_formats is -now defined as returning the format capabilities const. -This has always been the case but was never enforced -through the API itself. Any consumer of this API that -is not treating the formats as immutable should update -their code to create a new format capabilities and set -it on the stream instead. diff --git a/doc/UPGRADE-staging/vm_deprecated_removals.txt b/doc/UPGRADE-staging/vm_deprecated_removals.txt deleted file mode 100644 index 684a012b61..0000000000 --- a/doc/UPGRADE-staging/vm_deprecated_removals.txt +++ /dev/null @@ -1,7 +0,0 @@ -Subject: app_voicemail -Master-Only: True - -The MessageExists dialplan application and the MESSAGE_EXISTS dialplan -function were removed. The were deprecated in Asterisk 1.6.0 and -Asterisk 11.0.0 respectively. The VM_INFO() dialplan function is the -supported mechanism to query the status of a given mailbox. -- 2.47.2