From 9ff955f4d104c7ab89eaeedc9f6d48510388654c Mon Sep 17 00:00:00 2001 From: Asterisk Development Team Date: Wed, 13 Oct 2021 05:21:51 -0500 Subject: [PATCH] Update CHANGES and UPGRADE.txt for 19.0.0 --- CHANGES | 436 ++++++++++++++++++ UPGRADE.txt | 224 +++++++++ doc/CHANGES-staging/app_confbridge.txt | 7 - doc/CHANGES-staging/app_confbridge_answer.txt | 6 - doc/CHANGES-staging/app_confkick.txt | 6 - doc/CHANGES-staging/app_dial_announcement.txt | 6 - doc/CHANGES-staging/app_dtmfstore.txt | 6 - doc/CHANGES-staging/app_milliwatt.txt | 11 - doc/CHANGES-staging/app_morsecode.txt | 6 - doc/CHANGES-staging/app_originate_codecs.txt | 6 - doc/CHANGES-staging/app_originate_vars.txt | 6 - doc/CHANGES-staging/app_queue.txt | 4 - doc/CHANGES-staging/app_queue_stats.txt | 7 - doc/CHANGES-staging/app_read.txt | 5 - doc/CHANGES-staging/app_reload.txt | 4 - doc/CHANGES-staging/app_transferprotocol.txt | 6 - doc/CHANGES-staging/app_voicemail.txt | 7 - doc/CHANGES-staging/app_waitforcond.txt | 5 - doc/CHANGES-staging/chan_iax2.txt | 4 - doc/CHANGES-staging/chan_iax2_ani2.txt | 4 - doc/CHANGES-staging/flash_ami_event.txt | 3 - doc/CHANGES-staging/func_channel.txt | 4 - doc/CHANGES-staging/func_env.txt | 5 - doc/CHANGES-staging/func_framedrop.txt | 5 - doc/CHANGES-staging/func_min_max.txt | 4 - .../func_odbc_ARGC_minargs.txt | 20 - doc/CHANGES-staging/func_scramble.txt | 5 - doc/CHANGES-staging/func_strings.txt | 7 - doc/CHANGES-staging/func_vmcount.txt | 3 - doc/CHANGES-staging/func_volume_read.txt | 4 - doc/CHANGES-staging/logger.txt | 5 - doc/CHANGES-staging/logger_category.txt | 18 - doc/CHANGES-staging/logger_dateformat.txt | 47 -- doc/CHANGES-staging/logger_format.txt | 12 - doc/CHANGES-staging/manager_message_send.txt | 6 - doc/CHANGES-staging/media_cache_cachedir.txt | 9 - doc/CHANGES-staging/messagesend.txt | 16 - doc/CHANGES-staging/mf.txt | 6 - .../mixmonitor_manager_events.txt | 5 - ...pjsip_endpoint_unauthenticated_options.txt | 5 - doc/CHANGES-staging/pjsip_read_headers.txt | 5 - .../pjsip_send_session_refresh.txt | 4 - .../pjsip_transport_partial_reload.txt | 4 - doc/CHANGES-staging/res_pjproject.txt | 8 - doc/CHANGES-staging/res_pjsip.txt | 5 - .../res_pjsip_dialog_info_body_generator.txt | 5 - doc/CHANGES-staging/res_pjsip_dtmf.txt | 5 - doc/CHANGES-staging/res_pjsip_messaging.txt | 7 - doc/CHANGES-staging/res_pjsip_registrar.txt | 7 - .../res_pjsip_t38_bind_fixes.txt | 9 - ...s_rtp_asterisk_stun_software_attribute.txt | 8 - ...asterisk_stunaddr_recurring_resolution.txt | 6 - doc/CHANGES-staging/res_stasis_playback.txt | 9 - doc/CHANGES-staging/res_statsd.txt | 5 - doc/CHANGES-staging/res_tonedetect.txt | 5 - doc/CHANGES-staging/say.txt | 7 - .../srtp_replay_protection.txt | 9 - doc/CHANGES-staging/voicemail_beep.txt | 3 - doc/CHANGES-staging/voicemail_early_media.txt | 6 - doc/UPGRADE-staging/app_dahdiras_removal.txt | 6 - doc/UPGRADE-staging/app_fax_removal.txt | 6 - doc/UPGRADE-staging/app_ices_removal.txt | 6 - doc/UPGRADE-staging/app_image_removal.txt | 6 - .../app_meetme_deprecation.txt | 6 - doc/UPGRADE-staging/app_mysql_removal.txt | 6 - doc/UPGRADE-staging/app_nbscat_removal.txt | 6 - .../app_osplookup_deprecation.txt | 6 - doc/UPGRADE-staging/app_url_removal.txt | 6 - doc/UPGRADE-staging/asterisk_logrotate.txt | 9 - doc/UPGRADE-staging/cdr_mysql_removal.txt | 6 - doc/UPGRADE-staging/cdr_syslog_removal.txt | 6 - doc/UPGRADE-staging/chan_alsa_deprecation.txt | 6 - doc/UPGRADE-staging/chan_iax2_rsa.txt | 15 - doc/UPGRADE-staging/chan_mgcp_deprecation.txt | 6 - doc/UPGRADE-staging/chan_misdn_removal.txt | 6 - doc/UPGRADE-staging/chan_nbs_removal.txt | 6 - doc/UPGRADE-staging/chan_oss_removal.txt | 6 - doc/UPGRADE-staging/chan_phone_removal.txt | 6 - .../chan_skinny_deprecation.txt | 6 - doc/UPGRADE-staging/chan_vpb_removal.txt | 6 - doc/UPGRADE-staging/conf2ael_removal.txt | 6 - .../do-not-build-chan-sip-by-default.txt | 5 - .../http-media-cache-lookup-order.txt | 9 - doc/UPGRADE-staging/menuselect-could-fail.txt | 5 - doc/UPGRADE-staging/muted_removal.txt | 6 - .../res_config_sqlite_removal.txt | 6 - doc/UPGRADE-staging/res_monitor_disabled.txt | 8 - .../res_pktccops_deprecation.txt | 6 - .../srtp_replay_protection.txt | 9 - .../stir-shaken-public-key-url.txt | 6 - doc/UPGRADE-staging/stir_shaken_origid.txt | 8 - 91 files changed, 660 insertions(+), 624 deletions(-) delete mode 100644 doc/CHANGES-staging/app_confbridge.txt delete mode 100644 doc/CHANGES-staging/app_confbridge_answer.txt delete mode 100644 doc/CHANGES-staging/app_confkick.txt delete mode 100644 doc/CHANGES-staging/app_dial_announcement.txt delete mode 100644 doc/CHANGES-staging/app_dtmfstore.txt delete mode 100644 doc/CHANGES-staging/app_milliwatt.txt delete mode 100644 doc/CHANGES-staging/app_morsecode.txt delete mode 100644 doc/CHANGES-staging/app_originate_codecs.txt delete mode 100644 doc/CHANGES-staging/app_originate_vars.txt delete mode 100644 doc/CHANGES-staging/app_queue.txt delete mode 100644 doc/CHANGES-staging/app_queue_stats.txt delete mode 100644 doc/CHANGES-staging/app_read.txt delete mode 100644 doc/CHANGES-staging/app_reload.txt delete mode 100644 doc/CHANGES-staging/app_transferprotocol.txt delete mode 100644 doc/CHANGES-staging/app_voicemail.txt delete mode 100644 doc/CHANGES-staging/app_waitforcond.txt delete mode 100644 doc/CHANGES-staging/chan_iax2.txt delete mode 100644 doc/CHANGES-staging/chan_iax2_ani2.txt delete mode 100644 doc/CHANGES-staging/flash_ami_event.txt delete mode 100644 doc/CHANGES-staging/func_channel.txt delete mode 100644 doc/CHANGES-staging/func_env.txt delete mode 100644 doc/CHANGES-staging/func_framedrop.txt delete mode 100644 doc/CHANGES-staging/func_min_max.txt delete mode 100644 doc/CHANGES-staging/func_odbc_ARGC_minargs.txt delete mode 100644 doc/CHANGES-staging/func_scramble.txt delete mode 100644 doc/CHANGES-staging/func_strings.txt delete mode 100644 doc/CHANGES-staging/func_vmcount.txt delete mode 100644 doc/CHANGES-staging/func_volume_read.txt delete mode 100644 doc/CHANGES-staging/logger.txt delete mode 100644 doc/CHANGES-staging/logger_category.txt delete mode 100644 doc/CHANGES-staging/logger_dateformat.txt delete mode 100644 doc/CHANGES-staging/logger_format.txt delete mode 100644 doc/CHANGES-staging/manager_message_send.txt delete mode 100644 doc/CHANGES-staging/media_cache_cachedir.txt delete mode 100644 doc/CHANGES-staging/messagesend.txt delete mode 100644 doc/CHANGES-staging/mf.txt delete mode 100644 doc/CHANGES-staging/mixmonitor_manager_events.txt delete mode 100644 doc/CHANGES-staging/pjsip_endpoint_unauthenticated_options.txt delete mode 100644 doc/CHANGES-staging/pjsip_read_headers.txt delete mode 100644 doc/CHANGES-staging/pjsip_send_session_refresh.txt delete mode 100644 doc/CHANGES-staging/pjsip_transport_partial_reload.txt delete mode 100644 doc/CHANGES-staging/res_pjproject.txt delete mode 100644 doc/CHANGES-staging/res_pjsip.txt delete mode 100644 doc/CHANGES-staging/res_pjsip_dialog_info_body_generator.txt delete mode 100644 doc/CHANGES-staging/res_pjsip_dtmf.txt delete mode 100644 doc/CHANGES-staging/res_pjsip_messaging.txt delete mode 100644 doc/CHANGES-staging/res_pjsip_registrar.txt delete mode 100644 doc/CHANGES-staging/res_pjsip_t38_bind_fixes.txt delete mode 100644 doc/CHANGES-staging/res_rtp_asterisk_stun_software_attribute.txt delete mode 100644 doc/CHANGES-staging/res_rtp_asterisk_stunaddr_recurring_resolution.txt delete mode 100644 doc/CHANGES-staging/res_stasis_playback.txt delete mode 100644 doc/CHANGES-staging/res_statsd.txt delete mode 100644 doc/CHANGES-staging/res_tonedetect.txt delete mode 100644 doc/CHANGES-staging/say.txt delete mode 100644 doc/CHANGES-staging/srtp_replay_protection.txt delete mode 100644 doc/CHANGES-staging/voicemail_beep.txt delete mode 100644 doc/CHANGES-staging/voicemail_early_media.txt delete mode 100644 doc/UPGRADE-staging/app_dahdiras_removal.txt delete mode 100644 doc/UPGRADE-staging/app_fax_removal.txt delete mode 100644 doc/UPGRADE-staging/app_ices_removal.txt delete mode 100644 doc/UPGRADE-staging/app_image_removal.txt delete mode 100644 doc/UPGRADE-staging/app_meetme_deprecation.txt delete mode 100644 doc/UPGRADE-staging/app_mysql_removal.txt delete mode 100644 doc/UPGRADE-staging/app_nbscat_removal.txt delete mode 100644 doc/UPGRADE-staging/app_osplookup_deprecation.txt delete mode 100644 doc/UPGRADE-staging/app_url_removal.txt delete mode 100644 doc/UPGRADE-staging/asterisk_logrotate.txt delete mode 100644 doc/UPGRADE-staging/cdr_mysql_removal.txt delete mode 100644 doc/UPGRADE-staging/cdr_syslog_removal.txt delete mode 100644 doc/UPGRADE-staging/chan_alsa_deprecation.txt delete mode 100644 doc/UPGRADE-staging/chan_iax2_rsa.txt delete mode 100644 doc/UPGRADE-staging/chan_mgcp_deprecation.txt delete mode 100644 doc/UPGRADE-staging/chan_misdn_removal.txt delete mode 100644 doc/UPGRADE-staging/chan_nbs_removal.txt delete mode 100644 doc/UPGRADE-staging/chan_oss_removal.txt delete mode 100644 doc/UPGRADE-staging/chan_phone_removal.txt delete mode 100644 doc/UPGRADE-staging/chan_skinny_deprecation.txt delete mode 100644 doc/UPGRADE-staging/chan_vpb_removal.txt delete mode 100644 doc/UPGRADE-staging/conf2ael_removal.txt delete mode 100644 doc/UPGRADE-staging/do-not-build-chan-sip-by-default.txt delete mode 100644 doc/UPGRADE-staging/http-media-cache-lookup-order.txt delete mode 100644 doc/UPGRADE-staging/menuselect-could-fail.txt delete mode 100644 doc/UPGRADE-staging/muted_removal.txt delete mode 100644 doc/UPGRADE-staging/res_config_sqlite_removal.txt delete mode 100644 doc/UPGRADE-staging/res_monitor_disabled.txt delete mode 100644 doc/UPGRADE-staging/res_pktccops_deprecation.txt delete mode 100644 doc/UPGRADE-staging/srtp_replay_protection.txt delete mode 100644 doc/UPGRADE-staging/stir-shaken-public-key-url.txt delete mode 100644 doc/UPGRADE-staging/stir_shaken_origid.txt diff --git a/CHANGES b/CHANGES index e05bfe65a7..9d599ea840 100644 --- a/CHANGES +++ b/CHANGES @@ -12,6 +12,442 @@ === ============================================================================== +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 18.0.0 to Asterisk 19.0.0 ------------ +------------------------------------------------------------------------------ + +AMI Flash event +------------------ + * Hook flash events are now exposed as AMI events. + +Add variable support to Originate +------------------ + * The Originate application now allows + variables to be set on the new channel + through a new option. + +Channel-agnostic MF support +------------------ + * A SendMF application and PlayMF manager + application are now included to send + arbitrary standard R1 MF tones on the + current channel or another specified channel. + +Core +------------------ + * Added debug logging categories that allow a user to output debug information + based on a specified category. This lets the user limit, and filter debug + output to data relevant to a particular context, or topic. For instance the + following categories are now available for debug logging purposes: + + dtls, dtls_packet, ice, rtcp, rtcp_packet, rtp, rtp_packet, stun, stun_packet + + These debug categories can be enable/disable via an Asterisk CLI command: + + core set debug category [:] [category[: [] ...] + + If no sub-level is associated all debug statements for a given category are + output. If a sub-level is given then only those statements assigned a value + at or below the associated sub-level are output. + + * The location where the media cache stores its temporary files + is no longer hardcoded to /tmp but can now be configured separately + via the astcachedir config variable in asterisk.conf. + + The default location for astcachedir is now /var/cache/asterisk + instead of /tmp, please make sure to manually cleanup and/or + migrate the temporary files in /tmp after upgrading. + +Handle non-standard Meter metric type safely +------------------ + * A meter_support flag has been introduced that defaults to true to maintain current behaviour. + If disabled, a counter metric type will be used instead wherever a meter metric type was used, + the counter will have a "_meter" suffix appended to the metric name. + +MessageSend +------------------ + * The MessageSend dialplan application now takes an + optional third argument that can set the message's + "To" field on outgoing messages. It's an alternative + to using the MESSAGE(to) dialplan function. + + To prevent confusion with the first argument, currently + named "to", it's been renamed to "destination". + Its function, creating the request URI, hasn't changed. + + The online documentation has also been enhanced to + explain the behavior. + + Despite the changes in this commit, there should be + no impact to current users of MessageSend. + + * The MessageSend AMI action has been updated to allow the Destination + and the To addresses to be provided separately. This brings the + MessageSend manager command in line with the capabilities of the + MessageSend dialplan application. + +New ConfKick application +------------------ + * Adds a ConfKick() application, which allows + a specific channel, all users, or all non-admin + users to be kicked from a conference bridge. + +New Reload application +------------------ + * Adds an application to reload modules + +PlaybackFinished has a new error state +------------------ + * The PlaybackFinished event now has a new state "failed" + that is used when the sound file was not played due to an error. + Before the state on PlaybackFinished was always "done". + + In case of multiple sound files to be played, + the PlaybackFinished is sent only once in the end of the list, + even in case of error. + +WaitForCondition application +------------------ + * This application provides a way to halt + dialplan execution until a provided + condition evaluates to true. + +app_confbridge +------------------ + * app_confbridge now has the ability to force the estimated bitrate on an SFU + bridge. To use it, set a bridge profile's remb_behavior to "force" and + set remb_estimated_bitrate to a rate in bits per second. The + remb_estimated_bitrate parameter is ignored if remb_behavior is something + other than "force". + +app_confbridge answer supervision control +------------------ + * app_confbridge now provides a user option to prevent + answer supervision if the channel hasn't been + answered yet. To use it, set a user profile's + answer_channel option to no. + +app_dial announcement option +------------------ + * The A option for Dial now supports + playing audio to the caller as well + as the called party. + +app_dtmfstore +------------------ + * New application which collects digits + dialed and stores them into + a specified variable. + +app_milliwatt +------------------ + * The Milliwatt application's existing behavior is + incorrect in that it plays a constant tone, which + is not how digital milliwatt test lines actually + work. + + An option is added so that a proper milliwatt test + tone can be provided, including a 1 second silent + interval every 10 seconds. However, for compatability + reasons, the default behavior remains unchanged. + +app_mixmonitor +------------------ + * app_mixmonitor now sends manager events MixMonitorStart, MixMonitorStop and + MixMonitorMute when the channel monitoring is started, stopped and muted (or + unmuted) respectively. + +app_morsecode +------------------ + * Extends the Morsecode application by adding support for + American Morse code and adds a configurable option + for the frequency used in off intervals. + +app_originate +------------------ + * Codecs can now be specified for dialplan-originated + calls, as with call files and the manager action. + By default, only the slin codec is now used, instead + of all the slin* codecs. + +app_queue +------------------ + * Reload behavior in app_queue has been changed so + queue and agent stats are not reset during full + app_queue module reloads. The queue reset stats + CLI command may still be used to reset stats while + Asterisk is running. + +app_queue.c +------------------ + * Allow multiple files to be streamed for agent announcement. + +app_read +------------------ + * A new option allows the digit '#' to be read literally, + rather than used exclusively as the input terminator + character. + +app_voicemail +------------------ + * The VoiceMail application can now be configured to send greetings and + instructions via early media and only answering the channel when it is + time for the caller to record their message. This behavior can be + activated by passing the new 'e' option to VoiceMail. + + * You can now customize the "beep" tone or omit it entirely. + + * Add a new 'S' option to VoiceMail which prevents the instructions + (vm-intro) from being played if a busy/unavailable/temporary greeting + from the voicemail user is played. This is similar to the existing 's' + option except that instructions will still be played if no user + greeting is available. + +chan_iax2 +------------------ + * You can now specify a default "auth" method in the + [general] section of iax.conf + + * ANI2 (OLI) is now transmitted over IAX2 calls + as an information element. + +chan_pjsip +------------------ + * The PJSIP_SEND_SESSION_REFRESH dialplan function now issues a warning, and + returns unsuccessful if it's used on a channel prior to answering. + + * Add function PJSIP_HEADERS() to get list of headers by pattern in the same way as SIP_HEADERS() do. + + Add ability to read header by pattern using PJSIP_HEADER(). + +chan_pjsip, app_transfer +------------------ + * Added TRANSFERSTATUSPROTOCOL variable. When transfer is performed, + transfers can pass a protocol specific error code. + Example, in SIP 3xx-6xx represent any SIP specific error received when + performing a REFER. + +func_channel +------------------ + * Adds the CHANNEL_EXISTS function to check for the existence + of a channel by name or unique ID. + +func_env.c +------------------ + * Two new functions, DIRNAME and BASENAME, are now + included which allow users to obtain the directory + or the base filename of any file. + +func_framedrop +------------------ + * New function to selectively drop specified frames + in either direction on a channel. + +func_math: Three new dialplan functions +------------------ + * Introduce three new functions, MIN, MAX, and ABS, which can be used to + obtain the minimum or maximum of up to two integers or absolute value. + +func_odbc +------------------ + * Introduce an ARGC variable for func_odbc functions, along with a minargs + per-function configuration option. + + minargs enables enforcing of minimum count of arguments to pass to + func_odbc, so if you're unconditionally using ARG1 through ARG4 then + this should be set to 4. func_odbc will generate an error in this case, + so for example + + [FOO] + minargs = 4 + + and ODBC_FOO(a,b,c) in dialplan will now error out instead of using a + potentially leaked ARG4 from Gosub(). + + ARGC is needed if you're using optional argument, to verify whether or + not an argument has been passed, else it's possible to use a leaked ARGn + from Gosub (app_stack). So now you can safely do + ${IF($[${ARGC}>3]?${ARGV}:default value)} kind of thing. + +func_scramble +------------------ + * Adds an audio scrambler function that may be used to + distort voice audio on a channel as a privacy + enhancement. + +func_strings +------------------ + * A new STRBETWEEN function is now included which + allows a substring to be inserted between characters + in a string. This is particularly useful for transforming + dial strings, such as adding pauses between digits + for a string of digits that are sent to another channel. + +func_vmcount +------------------ + * Multiple mailboxes may now be specified instead of just one. + +func_volume now can be read +------------------ + * The VOLUME function can now also be used + to read existing values previously set. + +logger +------------------ + * Added a new log formatter called "plain" that always prints + file, function and line number if available (even for verbose + messages) and never prints color control characters. Most + suitable for file output but can be used for other channels + as well. + + You use it in logger.conf like so: + debug => [plain]debug + console => [plain]error,warning,debug,notice,pjsip_history + messages => [plain]warning,error,verbose + + * The dateformat option in logger.conf will now control the remote + console (asterisk -r -T) timestamp format. Previously, dateformat only + controlled the formatting of the timestamp going to log files and the + main console (asterisk -c) but only for non-verbose messages. + + Internally, Asterisk does not send the logging timestamp with verbose + messages to console clients. It's up to the Asterisk remote consoles + to format verbose messages. Asterisk remote consoles previously did + not load dateformat from logger.conf. + + Previously there was a non-configurable and hard-coded "%b %e %T" + dateformat that would be used no matter what on all verbose console + messages printed on remote consoles. + + Example: + logger.conf + dateformat=%F %T.%3q + + # asterisk -rvvv -T + [2021-03-19 09:54:19.760-0400] Loading res_stasis_answer.so. + [Mar 19 09:55:43] -- Goto (dialExten,s,1) + + Given the following example configuration in logger.conf, Asterisk log + files and the console, will log verbose messages using the given + timestamp. Now ensuring that all remote console messages are logged + with the same dateformat as other log streams. + + --- + [general] + dateformat=%F %T.%3q + + [logfiles] + console => notice,warning,error,verbose + full => notice,warning,error,debug,verbose + --- + + Now we have a globally-defined dateformat that will be used + consistently across the Asterisk main console, remote consoles, and + log files. + + Now we have consistent logging: + + # asterisk -rvvv -T + [2021-03-19 09:54:19.760-0400] Loading res_stasis_answer.so. + [2021-03-19 09:55:43.920-0400] -- Goto (dialExten,s,1) + + * Added the ability to define custom log levels in logger.conf + and use them in the Log dialplan application. Also adds a + logger show levels CLI command. + +res_pjproject +------------------ + * In pjproject.conf you can now map pjproject log levels + to the Asterisk TRACE log level. The default mappings + have therefore changed so that only pjproject levels + 3 and 4 are mapped to DEBUG and 5 and 6 are now mapped + to TRACE. Previously 3, 4, 5, and 6 were all mapped to + DEBUG. + +res_pjsip +------------------ + * PJSIP transports can now be partially reloaded safely. This allows the + local_net and external_* options to be updated without restarting Asterisk. + + * PJSIP endpoints can now be configured to skip authentication when + handling OPTIONS requests by setting the allow_unauthenticated_options + configuration property to 'yes.' + + * PJSIP support of registrations of endpoints in multidomain + scenarios, where the endpoint contains the domain info + in pjsip_endpoint.conf + +res_pjsip_dialog_info_body_generator +------------------ + * PJSIP now supports RFC 4235 Section 4.1.6 dialog-info+xml local and + remote elements by iterating through ringing channels and inserting + that info into NOTIFY packet sent to the endpoint. + +res_pjsip_messaging +------------------ + * Implemented the new "to" parameter of the MessageSend() + dialplan application. This allows a user to specify + a complete SIP "To" header separate from the Request URI. + We now also accept a destination in the same format + as Dial()... PJSIP/number@endpoint + +res_pjsip_registrar +------------------ + * Adds new PJSIP AOR option remove_unavailable to either + remove unavailable contacts when a REGISTER exceeds + max_contacts when remove_existing is disabled, or + prioritize unavailable contacts over other existing + contacts when remove_existing is enabled. + +res_pjsip_t38 +------------------ + * In res_pjsip_sdp_rtp, the bind_rtp_to_media_address option and the + fallback use of the transport's bind address solve problems sending + media on systems that cannot send ipv4 packets on ipv6 sockets, and + certain other situations. This change extends both of these behaviors + to UDPTL sessions as well in res_pjsip_t38, to fix fax-specific + problems on these systems, introducing a new option + endpoint/t38_bind_udptl_to_media_address. + +res_rtp_asterisk +------------------ + * By default Asterisk reports the PJSIP version in all + STUN packets it sends. + + This behaviour may not be desired in a production + environment and can now be disabled by setting the + stun_software_attribute option to 'no' in rtp.conf. + + * When the address of the STUN server (stunaddr) is a name resolved via DNS, the + stunaddr will be recurringly resolved when the DNS answer Time-To-Live (TTL) + expires. This allows the STUN server to change its IP address without having to + reload the res_rtp_asterisk module. + +res_srtp +------------------ + * SRTP replay protection has been added to res_srtp and + a new configuration option "srtpreplayprotection" has + been added to the rtp.conf config file. For security + reasons, the default setting is "yes". Buggy clients + may not handle this correctly which could result in + no, or one way, audio and Asterisk error messages like + "replay check failed". + +res_tonedetect +------------------ + * Arbitrary tone detection is now available through a + WaitForTone application (blocking) and a TONE_DETECT + function (non-blocking). + +say.c +------------------ + * Adds SAYFILES function to retrieve the file names that would + be played by corresponding Say applications, such as + SayDigits, SayAlpha, etc. + + Additionally adds SayMoney and SayOrdinal applications. + ------------------------------------------------------------------------------ --- New functionality introduced in Asterisk 18.0.0 -------------------------- ------------------------------------------------------------------------------ diff --git a/UPGRADE.txt b/UPGRADE.txt index 68261ae716..88e3359979 100644 --- a/UPGRADE.txt +++ b/UPGRADE.txt @@ -18,6 +18,230 @@ === =========================================================== +------------------------------------------------------------------------------ +--- New functionality introduced in Asterisk 19.0.0 -------------------------- +------------------------------------------------------------------------------ + +Log Rotate +------------------ + * The sample logger files have been changed to have .log as their file + extension. This was done so that when attached to issues on the issue + tracker, they are able to be opened in the browser for convenience. + Because of this, the asterisk.logrotate script has been updated to look + for .log extensions instead of no extension for files such as full + and messages. + +app_dahdiras +------------------ + * This module was deprecated in Asterisk 16 + and is now being removed in accordance with + the Asterisk Module Deprecation policy. + +app_fax +------------------ + * This module was deprecated in Asterisk 16 + and is now being removed in accordance with + the Asterisk Module Deprecation policy. + +app_ices +------------------ + * This module was deprecated in Asterisk 16 + and is now being removed in accordance with + the Asterisk Module Deprecation policy. + +app_image +------------------ + * This module was deprecated in Asterisk 16 + and is now being removed in accordance with + the Asterisk Module Deprecation policy. + +app_meetme +------------------ + * This module is now deprecated and will no + longer be built by default. It is scheduled + to be removed as of Asterisk 21. + +app_mysql +------------------ + * This module was deprecated in Asterisk 1.8 + and is now being removed in accordance with + the Asterisk Module Deprecation policy. + +app_nbscat +------------------ + * This module was deprecated in Asterisk 16 + and is now being removed in accordance with + the Asterisk Module Deprecation policy. + +app_osplookup +------------------ + * This module is now deprecated and will no + longer be built by default. It is scheduled + to be removed as of Asterisk 21. + +app_url +------------------ + * This module was deprecated in Asterisk 16 + and is now being removed in accordance with + the Asterisk Module Deprecation policy. + +cdr_mysql +------------------ + * This module was deprecated in Asterisk 1.8 + and is now being removed in accordance with + the Asterisk Module Deprecation policy. + +cdr_syslog +------------------ + * This module was deprecated in Asterisk 16 + and is now being removed in accordance with + the Asterisk Module Deprecation policy. + +chan_alsa +------------------ + * This module is now deprecated and will no + longer be built by default. It is scheduled + to be removed as of Asterisk 21. + +chan_mgcp +------------------ + * This module is now deprecated and will no + longer be built by default. It is scheduled + to be removed as of Asterisk 21. + +chan_misdn +------------------ + * This module was deprecated in Asterisk 16 + and is now being removed in accordance with + the Asterisk Module Deprecation policy. + +chan_nbs +------------------ + * This module was deprecated in Asterisk 16 + and is now being removed in accordance with + the Asterisk Module Deprecation policy. + +chan_oss +------------------ + * This module was deprecated in Asterisk 16 + and is now being removed in accordance with + the Asterisk Module Deprecation policy. + +chan_phone +------------------ + * This module was deprecated in Asterisk 16 + and is now being removed in accordance with + the Asterisk Module Deprecation policy. + +chan_sip +------------------ + * chan_sip is no longer built by default. To build it, make sure to + enable it when running 'make menuselect' + +chan_skinny +------------------ + * This module is now deprecated and will no + longer be built by default. It is scheduled + to be removed as of Asterisk 21. + +chan_vpb +------------------ + * This module was deprecated in Asterisk 16 + and is now being removed in accordance with + the Asterisk Module Deprecation policy. + +conf2ael +------------------ + * This application was deprecated in Asterisk 16 + and is now being removed in accordance with + the Asterisk Module Deprecation policy. + +muted +------------------ + * This application was deprecated in Asterisk 16 + and is now being removed in accordance with + the Asterisk Module Deprecation policy. + +res_config_sqlite +------------------ + * This module was deprecated in Asterisk 16 + and is now being removed in accordance with + the Asterisk Module Deprecation policy. + +res_monitor +------------------ + * This module is no longer built by default in + accordance with the Module Deprecation Policy. + If you require this functionality you will need + to enable it for building in menuselect. Note + that in the future res_monitor will be removed. + +res_pktccops +------------------ + * This module is now deprecated and will no + longer be built by default. It is scheduled + to be removed as of Asterisk 21. + +------------------------------------------------------------------------------ +--- Functionality changes from Asterisk 18.0.0 to Asterisk 19.0.0 ------------ +------------------------------------------------------------------------------ + +STIR/SHAKEN +------------------ + * The configuration option public_key_url in stir_shaken.conf + has been renamed to public_cert_url to better fit what it + contains. Only the name has changed - functionality is the + same. + + * STIR/SHAKEN originally needed an origid to be specified in + stir_shaken.conf under the certificate config object in + order to work. Now, one is automatically created by + generating a UUID, as recommended by RFC8588. Any origid + you have in your stir_shaken.conf will need to be removed + for the module to read in certificates. + +chan_iax2 +------------------ + * Encryption is now supported for RSA authentication. + + Currently, these auth configurations will cause a crash: + auth = md5,rsa + auth = plaintext,md5,rsa + + With a patched peer, the following will cause a crash: + auth = rsa + auth = md5,rsa + auth = plaintext,md5,rsa + + If both the peer and user are patches, no crash occurs. + Existing good configurations should continue to work. + +menuselect +------------------ + * menuselect --enable, --disable, --enable-category and --disable-category will + now fail with a non-zero exit code instead of silently failing if an invalid + option or category is specified. + +res_http_media_cache +------------------ + * When fetching a file for playback from a URL, Asterisk will now first + use the value of the Content-Type header in the HTTP response to + determine the format of the audio data, and only if it is unable to do + that will it attempt to parse the URL and extract the extension from + the path portion. Previously Asterisk would first look at the end of + the URL, which may have included query string parameters or a URL + fragment, which was error prone. + +res_srtp +------------------ + * SRTP replay protection has been added to res_srtp and + a new configuration option "srtpreplayprotection" has + been added to the rtp.conf config file. For security + reasons, the default setting is "yes". Buggy clients + may not handle this correctly which could result in + no, or one way, audio and Asterisk error messages like + "replay check failed". + ------------------------------------------------------------------------------ --- New functionality introduced in Asterisk 18.0.0 -------------------------- ------------------------------------------------------------------------------ diff --git a/doc/CHANGES-staging/app_confbridge.txt b/doc/CHANGES-staging/app_confbridge.txt deleted file mode 100644 index 092e392f5d..0000000000 --- a/doc/CHANGES-staging/app_confbridge.txt +++ /dev/null @@ -1,7 +0,0 @@ -Subject: app_confbridge - -app_confbridge now has the ability to force the estimated bitrate on an SFU -bridge. To use it, set a bridge profile's remb_behavior to "force" and -set remb_estimated_bitrate to a rate in bits per second. The -remb_estimated_bitrate parameter is ignored if remb_behavior is something -other than "force". diff --git a/doc/CHANGES-staging/app_confbridge_answer.txt b/doc/CHANGES-staging/app_confbridge_answer.txt deleted file mode 100644 index b975f48f4e..0000000000 --- a/doc/CHANGES-staging/app_confbridge_answer.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: app_confbridge answer supervision control - -app_confbridge now provides a user option to prevent -answer supervision if the channel hasn't been -answered yet. To use it, set a user profile's -answer_channel option to no. diff --git a/doc/CHANGES-staging/app_confkick.txt b/doc/CHANGES-staging/app_confkick.txt deleted file mode 100644 index 4250c7d6ba..0000000000 --- a/doc/CHANGES-staging/app_confkick.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: New ConfKick application - -Adds a ConfKick() application, which allows -a specific channel, all users, or all non-admin -users to be kicked from a conference bridge. - diff --git a/doc/CHANGES-staging/app_dial_announcement.txt b/doc/CHANGES-staging/app_dial_announcement.txt deleted file mode 100644 index 3947b0e4e0..0000000000 --- a/doc/CHANGES-staging/app_dial_announcement.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: app_dial announcement option - -The A option for Dial now supports -playing audio to the caller as well -as the called party. - diff --git a/doc/CHANGES-staging/app_dtmfstore.txt b/doc/CHANGES-staging/app_dtmfstore.txt deleted file mode 100644 index a82b5438bd..0000000000 --- a/doc/CHANGES-staging/app_dtmfstore.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: app_dtmfstore - -New application which collects digits -dialed and stores them into -a specified variable. - diff --git a/doc/CHANGES-staging/app_milliwatt.txt b/doc/CHANGES-staging/app_milliwatt.txt deleted file mode 100644 index 434ace22bb..0000000000 --- a/doc/CHANGES-staging/app_milliwatt.txt +++ /dev/null @@ -1,11 +0,0 @@ -Subject: app_milliwatt - -The Milliwatt application's existing behavior is -incorrect in that it plays a constant tone, which -is not how digital milliwatt test lines actually -work. - -An option is added so that a proper milliwatt test -tone can be provided, including a 1 second silent -interval every 10 seconds. However, for compatability -reasons, the default behavior remains unchanged. diff --git a/doc/CHANGES-staging/app_morsecode.txt b/doc/CHANGES-staging/app_morsecode.txt deleted file mode 100644 index b9e49b63ee..0000000000 --- a/doc/CHANGES-staging/app_morsecode.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: app_morsecode - -Extends the Morsecode application by adding support for -American Morse code and adds a configurable option -for the frequency used in off intervals. - diff --git a/doc/CHANGES-staging/app_originate_codecs.txt b/doc/CHANGES-staging/app_originate_codecs.txt deleted file mode 100644 index a0f52b13c5..0000000000 --- a/doc/CHANGES-staging/app_originate_codecs.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: app_originate - -Codecs can now be specified for dialplan-originated -calls, as with call files and the manager action. -By default, only the slin codec is now used, instead -of all the slin* codecs. diff --git a/doc/CHANGES-staging/app_originate_vars.txt b/doc/CHANGES-staging/app_originate_vars.txt deleted file mode 100644 index 4e08ae61f8..0000000000 --- a/doc/CHANGES-staging/app_originate_vars.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: Add variable support to Originate - -The Originate application now allows -variables to be set on the new channel -through a new option. - diff --git a/doc/CHANGES-staging/app_queue.txt b/doc/CHANGES-staging/app_queue.txt deleted file mode 100644 index 5d677b56b9..0000000000 --- a/doc/CHANGES-staging/app_queue.txt +++ /dev/null @@ -1,4 +0,0 @@ -Subject: app_queue.c - -Allow multiple files to be streamed for agent announcement. - diff --git a/doc/CHANGES-staging/app_queue_stats.txt b/doc/CHANGES-staging/app_queue_stats.txt deleted file mode 100644 index 36c0c3da06..0000000000 --- a/doc/CHANGES-staging/app_queue_stats.txt +++ /dev/null @@ -1,7 +0,0 @@ -Subject: app_queue - -Reload behavior in app_queue has been changed so -queue and agent stats are not reset during full -app_queue module reloads. The queue reset stats -CLI command may still be used to reset stats while -Asterisk is running. diff --git a/doc/CHANGES-staging/app_read.txt b/doc/CHANGES-staging/app_read.txt deleted file mode 100644 index df3247c1e1..0000000000 --- a/doc/CHANGES-staging/app_read.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: app_read - -A new option allows the digit '#' to be read literally, -rather than used exclusively as the input terminator -character. diff --git a/doc/CHANGES-staging/app_reload.txt b/doc/CHANGES-staging/app_reload.txt deleted file mode 100644 index 308db15c7c..0000000000 --- a/doc/CHANGES-staging/app_reload.txt +++ /dev/null @@ -1,4 +0,0 @@ -Subject: New Reload application - -Adds an application to reload modules - diff --git a/doc/CHANGES-staging/app_transferprotocol.txt b/doc/CHANGES-staging/app_transferprotocol.txt deleted file mode 100644 index 5d3521bbd4..0000000000 --- a/doc/CHANGES-staging/app_transferprotocol.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: chan_pjsip, app_transfer - -Added TRANSFERSTATUSPROTOCOL variable. When transfer is performed, -transfers can pass a protocol specific error code. -Example, in SIP 3xx-6xx represent any SIP specific error received when -performing a REFER. diff --git a/doc/CHANGES-staging/app_voicemail.txt b/doc/CHANGES-staging/app_voicemail.txt deleted file mode 100644 index c52d1f0666..0000000000 --- a/doc/CHANGES-staging/app_voicemail.txt +++ /dev/null @@ -1,7 +0,0 @@ -Subject: app_voicemail - -Add a new 'S' option to VoiceMail which prevents the instructions -(vm-intro) from being played if a busy/unavailable/temporary greeting -from the voicemail user is played. This is similar to the existing 's' -option except that instructions will still be played if no user -greeting is available. diff --git a/doc/CHANGES-staging/app_waitforcond.txt b/doc/CHANGES-staging/app_waitforcond.txt deleted file mode 100644 index a7ab60028d..0000000000 --- a/doc/CHANGES-staging/app_waitforcond.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: WaitForCondition application - -This application provides a way to halt -dialplan execution until a provided -condition evaluates to true. diff --git a/doc/CHANGES-staging/chan_iax2.txt b/doc/CHANGES-staging/chan_iax2.txt deleted file mode 100644 index 4e1d844204..0000000000 --- a/doc/CHANGES-staging/chan_iax2.txt +++ /dev/null @@ -1,4 +0,0 @@ -Subject: chan_iax2 - -You can now specify a default "auth" method in the -[general] section of iax.conf diff --git a/doc/CHANGES-staging/chan_iax2_ani2.txt b/doc/CHANGES-staging/chan_iax2_ani2.txt deleted file mode 100644 index 37c6fa6cf6..0000000000 --- a/doc/CHANGES-staging/chan_iax2_ani2.txt +++ /dev/null @@ -1,4 +0,0 @@ -Subject: chan_iax2 - -ANI2 (OLI) is now transmitted over IAX2 calls -as an information element. diff --git a/doc/CHANGES-staging/flash_ami_event.txt b/doc/CHANGES-staging/flash_ami_event.txt deleted file mode 100644 index 4cbea80683..0000000000 --- a/doc/CHANGES-staging/flash_ami_event.txt +++ /dev/null @@ -1,3 +0,0 @@ -Subject: AMI Flash event - -Hook flash events are now exposed as AMI events. diff --git a/doc/CHANGES-staging/func_channel.txt b/doc/CHANGES-staging/func_channel.txt deleted file mode 100644 index 7f92c3e014..0000000000 --- a/doc/CHANGES-staging/func_channel.txt +++ /dev/null @@ -1,4 +0,0 @@ -Subject: func_channel - -Adds the CHANNEL_EXISTS function to check for the existence -of a channel by name or unique ID. diff --git a/doc/CHANGES-staging/func_env.txt b/doc/CHANGES-staging/func_env.txt deleted file mode 100644 index af03d5f0d1..0000000000 --- a/doc/CHANGES-staging/func_env.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: func_env.c - -Two new functions, DIRNAME and BASENAME, are now -included which allow users to obtain the directory -or the base filename of any file. diff --git a/doc/CHANGES-staging/func_framedrop.txt b/doc/CHANGES-staging/func_framedrop.txt deleted file mode 100644 index c17bccd74c..0000000000 --- a/doc/CHANGES-staging/func_framedrop.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: func_framedrop - -New function to selectively drop specified frames -in either direction on a channel. - diff --git a/doc/CHANGES-staging/func_min_max.txt b/doc/CHANGES-staging/func_min_max.txt deleted file mode 100644 index df2b6653e0..0000000000 --- a/doc/CHANGES-staging/func_min_max.txt +++ /dev/null @@ -1,4 +0,0 @@ -Subject: func_math: Three new dialplan functions - -Introduce three new functions, MIN, MAX, and ABS, which can be used to -obtain the minimum or maximum of up to two integers or absolute value. diff --git a/doc/CHANGES-staging/func_odbc_ARGC_minargs.txt b/doc/CHANGES-staging/func_odbc_ARGC_minargs.txt deleted file mode 100644 index 0984b5022d..0000000000 --- a/doc/CHANGES-staging/func_odbc_ARGC_minargs.txt +++ /dev/null @@ -1,20 +0,0 @@ -Subject: func_odbc - -Introduce an ARGC variable for func_odbc functions, along with a minargs -per-function configuration option. - -minargs enables enforcing of minimum count of arguments to pass to -func_odbc, so if you're unconditionally using ARG1 through ARG4 then -this should be set to 4. func_odbc will generate an error in this case, -so for example - -[FOO] -minargs = 4 - -and ODBC_FOO(a,b,c) in dialplan will now error out instead of using a -potentially leaked ARG4 from Gosub(). - -ARGC is needed if you're using optional argument, to verify whether or -not an argument has been passed, else it's possible to use a leaked ARGn -from Gosub (app_stack). So now you can safely do -${IF($[${ARGC}>3]?${ARGV}:default value)} kind of thing. diff --git a/doc/CHANGES-staging/func_scramble.txt b/doc/CHANGES-staging/func_scramble.txt deleted file mode 100644 index 4c1ffab78b..0000000000 --- a/doc/CHANGES-staging/func_scramble.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: func_scramble - -Adds an audio scrambler function that may be used to -distort voice audio on a channel as a privacy -enhancement. diff --git a/doc/CHANGES-staging/func_strings.txt b/doc/CHANGES-staging/func_strings.txt deleted file mode 100644 index d154464021..0000000000 --- a/doc/CHANGES-staging/func_strings.txt +++ /dev/null @@ -1,7 +0,0 @@ -Subject: func_strings - -A new STRBETWEEN function is now included which -allows a substring to be inserted between characters -in a string. This is particularly useful for transforming -dial strings, such as adding pauses between digits -for a string of digits that are sent to another channel. diff --git a/doc/CHANGES-staging/func_vmcount.txt b/doc/CHANGES-staging/func_vmcount.txt deleted file mode 100644 index ba2a0a1178..0000000000 --- a/doc/CHANGES-staging/func_vmcount.txt +++ /dev/null @@ -1,3 +0,0 @@ -Subject: func_vmcount - -Multiple mailboxes may now be specified instead of just one. diff --git a/doc/CHANGES-staging/func_volume_read.txt b/doc/CHANGES-staging/func_volume_read.txt deleted file mode 100644 index 8ea27cdce3..0000000000 --- a/doc/CHANGES-staging/func_volume_read.txt +++ /dev/null @@ -1,4 +0,0 @@ -Subject: func_volume now can be read - -The VOLUME function can now also be used -to read existing values previously set. diff --git a/doc/CHANGES-staging/logger.txt b/doc/CHANGES-staging/logger.txt deleted file mode 100644 index d09ebccca2..0000000000 --- a/doc/CHANGES-staging/logger.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: logger - -Added the ability to define custom log levels in logger.conf -and use them in the Log dialplan application. Also adds a -logger show levels CLI command. diff --git a/doc/CHANGES-staging/logger_category.txt b/doc/CHANGES-staging/logger_category.txt deleted file mode 100644 index 67cc3ec7ad..0000000000 --- a/doc/CHANGES-staging/logger_category.txt +++ /dev/null @@ -1,18 +0,0 @@ -Subject: Core - -Added debug logging categories that allow a user to output debug information -based on a specified category. This lets the user limit, and filter debug -output to data relevant to a particular context, or topic. For instance the -following categories are now available for debug logging purposes: - - dtls, dtls_packet, ice, rtcp, rtcp_packet, rtp, rtp_packet, stun, stun_packet - -These debug categories can be enable/disable via an Asterisk CLI command: - - core set debug category [:] [category[: [] ...] - -If no sub-level is associated all debug statements for a given category are -output. If a sub-level is given then only those statements assigned a value -at or below the associated sub-level are output. - diff --git a/doc/CHANGES-staging/logger_dateformat.txt b/doc/CHANGES-staging/logger_dateformat.txt deleted file mode 100644 index efeb11803d..0000000000 --- a/doc/CHANGES-staging/logger_dateformat.txt +++ /dev/null @@ -1,47 +0,0 @@ -Subject: logger - -The dateformat option in logger.conf will now control the remote -console (asterisk -r -T) timestamp format. Previously, dateformat only -controlled the formatting of the timestamp going to log files and the -main console (asterisk -c) but only for non-verbose messages. - -Internally, Asterisk does not send the logging timestamp with verbose -messages to console clients. It's up to the Asterisk remote consoles -to format verbose messages. Asterisk remote consoles previously did -not load dateformat from logger.conf. - -Previously there was a non-configurable and hard-coded "%b %e %T" -dateformat that would be used no matter what on all verbose console -messages printed on remote consoles. - -Example: -logger.conf - dateformat=%F %T.%3q - -# asterisk -rvvv -T -[2021-03-19 09:54:19.760-0400] Loading res_stasis_answer.so. -[Mar 19 09:55:43] -- Goto (dialExten,s,1) - -Given the following example configuration in logger.conf, Asterisk log -files and the console, will log verbose messages using the given -timestamp. Now ensuring that all remote console messages are logged -with the same dateformat as other log streams. - ---- -[general] -dateformat=%F %T.%3q - -[logfiles] -console => notice,warning,error,verbose -full => notice,warning,error,debug,verbose ---- - -Now we have a globally-defined dateformat that will be used -consistently across the Asterisk main console, remote consoles, and -log files. - -Now we have consistent logging: - -# asterisk -rvvv -T -[2021-03-19 09:54:19.760-0400] Loading res_stasis_answer.so. -[2021-03-19 09:55:43.920-0400] -- Goto (dialExten,s,1) diff --git a/doc/CHANGES-staging/logger_format.txt b/doc/CHANGES-staging/logger_format.txt deleted file mode 100644 index 58d864d673..0000000000 --- a/doc/CHANGES-staging/logger_format.txt +++ /dev/null @@ -1,12 +0,0 @@ -Subject: logger - -Added a new log formatter called "plain" that always prints -file, function and line number if available (even for verbose -messages) and never prints color control characters. Most -suitable for file output but can be used for other channels -as well. - -You use it in logger.conf like so: -debug => [plain]debug -console => [plain]error,warning,debug,notice,pjsip_history -messages => [plain]warning,error,verbose diff --git a/doc/CHANGES-staging/manager_message_send.txt b/doc/CHANGES-staging/manager_message_send.txt deleted file mode 100644 index ab5b58a287..0000000000 --- a/doc/CHANGES-staging/manager_message_send.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: MessageSend - -The MessageSend AMI action has been updated to allow the Destination -and the To addresses to be provided separately. This brings the -MessageSend manager command in line with the capabilities of the -MessageSend dialplan application. diff --git a/doc/CHANGES-staging/media_cache_cachedir.txt b/doc/CHANGES-staging/media_cache_cachedir.txt deleted file mode 100644 index e30543fb29..0000000000 --- a/doc/CHANGES-staging/media_cache_cachedir.txt +++ /dev/null @@ -1,9 +0,0 @@ -Subject: Core - -The location where the media cache stores its temporary files -is no longer hardcoded to /tmp but can now be configured separately -via the astcachedir config variable in asterisk.conf. - -The default location for astcachedir is now /var/cache/asterisk -instead of /tmp, please make sure to manually cleanup and/or -migrate the temporary files in /tmp after upgrading. diff --git a/doc/CHANGES-staging/messagesend.txt b/doc/CHANGES-staging/messagesend.txt deleted file mode 100644 index 7977ff15c8..0000000000 --- a/doc/CHANGES-staging/messagesend.txt +++ /dev/null @@ -1,16 +0,0 @@ -Subject: MessageSend - -The MessageSend dialplan application now takes an -optional third argument that can set the message's -"To" field on outgoing messages. It's an alternative -to using the MESSAGE(to) dialplan function. - -To prevent confusion with the first argument, currently -named "to", it's been renamed to "destination". -Its function, creating the request URI, hasn't changed. - -The online documentation has also been enhanced to -explain the behavior. - -Despite the changes in this commit, there should be -no impact to current users of MessageSend. diff --git a/doc/CHANGES-staging/mf.txt b/doc/CHANGES-staging/mf.txt deleted file mode 100644 index 644f62a998..0000000000 --- a/doc/CHANGES-staging/mf.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: Channel-agnostic MF support - -A SendMF application and PlayMF manager -application are now included to send -arbitrary standard R1 MF tones on the -current channel or another specified channel. diff --git a/doc/CHANGES-staging/mixmonitor_manager_events.txt b/doc/CHANGES-staging/mixmonitor_manager_events.txt deleted file mode 100644 index 64b63e52e7..0000000000 --- a/doc/CHANGES-staging/mixmonitor_manager_events.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: app_mixmonitor - -app_mixmonitor now sends manager events MixMonitorStart, MixMonitorStop and -MixMonitorMute when the channel monitoring is started, stopped and muted (or -unmuted) respectively. diff --git a/doc/CHANGES-staging/pjsip_endpoint_unauthenticated_options.txt b/doc/CHANGES-staging/pjsip_endpoint_unauthenticated_options.txt deleted file mode 100644 index 9c8d32cb0e..0000000000 --- a/doc/CHANGES-staging/pjsip_endpoint_unauthenticated_options.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: res_pjsip - -PJSIP endpoints can now be configured to skip authentication when -handling OPTIONS requests by setting the allow_unauthenticated_options -configuration property to 'yes.' diff --git a/doc/CHANGES-staging/pjsip_read_headers.txt b/doc/CHANGES-staging/pjsip_read_headers.txt deleted file mode 100644 index 4dc641cdae..0000000000 --- a/doc/CHANGES-staging/pjsip_read_headers.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: chan_pjsip - -Add function PJSIP_HEADERS() to get list of headers by pattern in the same way as SIP_HEADERS() do. - -Add ability to read header by pattern using PJSIP_HEADER(). diff --git a/doc/CHANGES-staging/pjsip_send_session_refresh.txt b/doc/CHANGES-staging/pjsip_send_session_refresh.txt deleted file mode 100644 index 0705c293d7..0000000000 --- a/doc/CHANGES-staging/pjsip_send_session_refresh.txt +++ /dev/null @@ -1,4 +0,0 @@ -Subject: chan_pjsip - -The PJSIP_SEND_SESSION_REFRESH dialplan function now issues a warning, and -returns unsuccessful if it's used on a channel prior to answering. diff --git a/doc/CHANGES-staging/pjsip_transport_partial_reload.txt b/doc/CHANGES-staging/pjsip_transport_partial_reload.txt deleted file mode 100644 index 1d1b0b6266..0000000000 --- a/doc/CHANGES-staging/pjsip_transport_partial_reload.txt +++ /dev/null @@ -1,4 +0,0 @@ -Subject: res_pjsip - -PJSIP transports can now be partially reloaded safely. This allows the -local_net and external_* options to be updated without restarting Asterisk. diff --git a/doc/CHANGES-staging/res_pjproject.txt b/doc/CHANGES-staging/res_pjproject.txt deleted file mode 100644 index 132c9506b8..0000000000 --- a/doc/CHANGES-staging/res_pjproject.txt +++ /dev/null @@ -1,8 +0,0 @@ -Subject: res_pjproject - -In pjproject.conf you can now map pjproject log levels -to the Asterisk TRACE log level. The default mappings -have therefore changed so that only pjproject levels -3 and 4 are mapped to DEBUG and 5 and 6 are now mapped -to TRACE. Previously 3, 4, 5, and 6 were all mapped to -DEBUG. diff --git a/doc/CHANGES-staging/res_pjsip.txt b/doc/CHANGES-staging/res_pjsip.txt deleted file mode 100644 index ffbf13a9c2..0000000000 --- a/doc/CHANGES-staging/res_pjsip.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: res_pjsip - -PJSIP support of registrations of endpoints in multidomain -scenarios, where the endpoint contains the domain info -in pjsip_endpoint.conf diff --git a/doc/CHANGES-staging/res_pjsip_dialog_info_body_generator.txt b/doc/CHANGES-staging/res_pjsip_dialog_info_body_generator.txt deleted file mode 100644 index 0dd0a5762d..0000000000 --- a/doc/CHANGES-staging/res_pjsip_dialog_info_body_generator.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: res_pjsip_dialog_info_body_generator - -PJSIP now supports RFC 4235 Section 4.1.6 dialog-info+xml local and -remote elements by iterating through ringing channels and inserting -that info into NOTIFY packet sent to the endpoint. diff --git a/doc/CHANGES-staging/res_pjsip_dtmf.txt b/doc/CHANGES-staging/res_pjsip_dtmf.txt deleted file mode 100644 index 4dc2088c6f..0000000000 --- a/doc/CHANGES-staging/res_pjsip_dtmf.txt +++ /dev/null @@ -1,5 +0,0 @@ -res_pjsip_dtmf_info: Hook flash - -Adds recognition for application/ -hook-flash as a hook flash event. - diff --git a/doc/CHANGES-staging/res_pjsip_messaging.txt b/doc/CHANGES-staging/res_pjsip_messaging.txt deleted file mode 100644 index 46874dc55d..0000000000 --- a/doc/CHANGES-staging/res_pjsip_messaging.txt +++ /dev/null @@ -1,7 +0,0 @@ -Subject: res_pjsip_messaging - -Implemented the new "to" parameter of the MessageSend() -dialplan application. This allows a user to specify -a complete SIP "To" header separate from the Request URI. -We now also accept a destination in the same format -as Dial()... PJSIP/number@endpoint diff --git a/doc/CHANGES-staging/res_pjsip_registrar.txt b/doc/CHANGES-staging/res_pjsip_registrar.txt deleted file mode 100644 index a80f69ff08..0000000000 --- a/doc/CHANGES-staging/res_pjsip_registrar.txt +++ /dev/null @@ -1,7 +0,0 @@ -Subject: res_pjsip_registrar - -Adds new PJSIP AOR option remove_unavailable to either -remove unavailable contacts when a REGISTER exceeds -max_contacts when remove_existing is disabled, or -prioritize unavailable contacts over other existing -contacts when remove_existing is enabled. diff --git a/doc/CHANGES-staging/res_pjsip_t38_bind_fixes.txt b/doc/CHANGES-staging/res_pjsip_t38_bind_fixes.txt deleted file mode 100644 index d7bc8a1e9f..0000000000 --- a/doc/CHANGES-staging/res_pjsip_t38_bind_fixes.txt +++ /dev/null @@ -1,9 +0,0 @@ -Subject: res_pjsip_t38 - -In res_pjsip_sdp_rtp, the bind_rtp_to_media_address option and the -fallback use of the transport's bind address solve problems sending -media on systems that cannot send ipv4 packets on ipv6 sockets, and -certain other situations. This change extends both of these behaviors -to UDPTL sessions as well in res_pjsip_t38, to fix fax-specific -problems on these systems, introducing a new option -endpoint/t38_bind_udptl_to_media_address. diff --git a/doc/CHANGES-staging/res_rtp_asterisk_stun_software_attribute.txt b/doc/CHANGES-staging/res_rtp_asterisk_stun_software_attribute.txt deleted file mode 100644 index 93905f6d0a..0000000000 --- a/doc/CHANGES-staging/res_rtp_asterisk_stun_software_attribute.txt +++ /dev/null @@ -1,8 +0,0 @@ -Subject: res_rtp_asterisk - -By default Asterisk reports the PJSIP version in all -STUN packets it sends. - -This behaviour may not be desired in a production -environment and can now be disabled by setting the -stun_software_attribute option to 'no' in rtp.conf. diff --git a/doc/CHANGES-staging/res_rtp_asterisk_stunaddr_recurring_resolution.txt b/doc/CHANGES-staging/res_rtp_asterisk_stunaddr_recurring_resolution.txt deleted file mode 100644 index c78f4f51d4..0000000000 --- a/doc/CHANGES-staging/res_rtp_asterisk_stunaddr_recurring_resolution.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: res_rtp_asterisk - -When the address of the STUN server (stunaddr) is a name resolved via DNS, the -stunaddr will be recurringly resolved when the DNS answer Time-To-Live (TTL) -expires. This allows the STUN server to change its IP address without having to -reload the res_rtp_asterisk module. diff --git a/doc/CHANGES-staging/res_stasis_playback.txt b/doc/CHANGES-staging/res_stasis_playback.txt deleted file mode 100644 index cd5fa1102a..0000000000 --- a/doc/CHANGES-staging/res_stasis_playback.txt +++ /dev/null @@ -1,9 +0,0 @@ -Subject: PlaybackFinished has a new error state - -The PlaybackFinished event now has a new state "failed" -that is used when the sound file was not played due to an error. -Before the state on PlaybackFinished was always "done". - -In case of multiple sound files to be played, -the PlaybackFinished is sent only once in the end of the list, -even in case of error. diff --git a/doc/CHANGES-staging/res_statsd.txt b/doc/CHANGES-staging/res_statsd.txt deleted file mode 100644 index 317c65d00b..0000000000 --- a/doc/CHANGES-staging/res_statsd.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: Handle non-standard Meter metric type safely - -A meter_support flag has been introduced that defaults to true to maintain current behaviour. -If disabled, a counter metric type will be used instead wherever a meter metric type was used, -the counter will have a "_meter" suffix appended to the metric name. \ No newline at end of file diff --git a/doc/CHANGES-staging/res_tonedetect.txt b/doc/CHANGES-staging/res_tonedetect.txt deleted file mode 100644 index ddda8e899e..0000000000 --- a/doc/CHANGES-staging/res_tonedetect.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: res_tonedetect - -Arbitrary tone detection is now available through a -WaitForTone application (blocking) and a TONE_DETECT -function (non-blocking). diff --git a/doc/CHANGES-staging/say.txt b/doc/CHANGES-staging/say.txt deleted file mode 100644 index 115ceea15f..0000000000 --- a/doc/CHANGES-staging/say.txt +++ /dev/null @@ -1,7 +0,0 @@ -Subject: say.c - -Adds SAYFILES function to retrieve the file names that would -be played by corresponding Say applications, such as -SayDigits, SayAlpha, etc. - -Additionally adds SayMoney and SayOrdinal applications. diff --git a/doc/CHANGES-staging/srtp_replay_protection.txt b/doc/CHANGES-staging/srtp_replay_protection.txt deleted file mode 100644 index 945ddb5704..0000000000 --- a/doc/CHANGES-staging/srtp_replay_protection.txt +++ /dev/null @@ -1,9 +0,0 @@ -Subject: res_srtp - -SRTP replay protection has been added to res_srtp and -a new configuration option "srtpreplayprotection" has -been added to the rtp.conf config file. For security -reasons, the default setting is "yes". Buggy clients -may not handle this correctly which could result in -no, or one way, audio and Asterisk error messages like -"replay check failed". diff --git a/doc/CHANGES-staging/voicemail_beep.txt b/doc/CHANGES-staging/voicemail_beep.txt deleted file mode 100644 index d98b40356f..0000000000 --- a/doc/CHANGES-staging/voicemail_beep.txt +++ /dev/null @@ -1,3 +0,0 @@ -Subject: app_voicemail - -You can now customize the "beep" tone or omit it entirely. diff --git a/doc/CHANGES-staging/voicemail_early_media.txt b/doc/CHANGES-staging/voicemail_early_media.txt deleted file mode 100644 index 6dd79befae..0000000000 --- a/doc/CHANGES-staging/voicemail_early_media.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: app_voicemail - -The VoiceMail application can now be configured to send greetings and -instructions via early media and only answering the channel when it is -time for the caller to record their message. This behavior can be -activated by passing the new 'e' option to VoiceMail. diff --git a/doc/UPGRADE-staging/app_dahdiras_removal.txt b/doc/UPGRADE-staging/app_dahdiras_removal.txt deleted file mode 100644 index 7baa7ff97f..0000000000 --- a/doc/UPGRADE-staging/app_dahdiras_removal.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: app_dahdiras -Master-Only: True - -This module was deprecated in Asterisk 16 -and is now being removed in accordance with -the Asterisk Module Deprecation policy. diff --git a/doc/UPGRADE-staging/app_fax_removal.txt b/doc/UPGRADE-staging/app_fax_removal.txt deleted file mode 100644 index ec57aecfe3..0000000000 --- a/doc/UPGRADE-staging/app_fax_removal.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: app_fax -Master-Only: True - -This module was deprecated in Asterisk 16 -and is now being removed in accordance with -the Asterisk Module Deprecation policy. diff --git a/doc/UPGRADE-staging/app_ices_removal.txt b/doc/UPGRADE-staging/app_ices_removal.txt deleted file mode 100644 index 79ab27ba59..0000000000 --- a/doc/UPGRADE-staging/app_ices_removal.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: app_ices -Master-Only: True - -This module was deprecated in Asterisk 16 -and is now being removed in accordance with -the Asterisk Module Deprecation policy. diff --git a/doc/UPGRADE-staging/app_image_removal.txt b/doc/UPGRADE-staging/app_image_removal.txt deleted file mode 100644 index f1c7a535f9..0000000000 --- a/doc/UPGRADE-staging/app_image_removal.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: app_image -Master-Only: True - -This module was deprecated in Asterisk 16 -and is now being removed in accordance with -the Asterisk Module Deprecation policy. diff --git a/doc/UPGRADE-staging/app_meetme_deprecation.txt b/doc/UPGRADE-staging/app_meetme_deprecation.txt deleted file mode 100644 index 23f7d4d581..0000000000 --- a/doc/UPGRADE-staging/app_meetme_deprecation.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: app_meetme -Master-Only: True - -This module is now deprecated and will no -longer be built by default. It is scheduled -to be removed as of Asterisk 21. diff --git a/doc/UPGRADE-staging/app_mysql_removal.txt b/doc/UPGRADE-staging/app_mysql_removal.txt deleted file mode 100644 index 7af75faac1..0000000000 --- a/doc/UPGRADE-staging/app_mysql_removal.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: app_mysql -Master-Only: True - -This module was deprecated in Asterisk 1.8 -and is now being removed in accordance with -the Asterisk Module Deprecation policy. diff --git a/doc/UPGRADE-staging/app_nbscat_removal.txt b/doc/UPGRADE-staging/app_nbscat_removal.txt deleted file mode 100644 index a1d15d88dc..0000000000 --- a/doc/UPGRADE-staging/app_nbscat_removal.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: app_nbscat -Master-Only: True - -This module was deprecated in Asterisk 16 -and is now being removed in accordance with -the Asterisk Module Deprecation policy. diff --git a/doc/UPGRADE-staging/app_osplookup_deprecation.txt b/doc/UPGRADE-staging/app_osplookup_deprecation.txt deleted file mode 100644 index 27e3bb6d26..0000000000 --- a/doc/UPGRADE-staging/app_osplookup_deprecation.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: app_osplookup -Master-Only: True - -This module is now deprecated and will no -longer be built by default. It is scheduled -to be removed as of Asterisk 21. diff --git a/doc/UPGRADE-staging/app_url_removal.txt b/doc/UPGRADE-staging/app_url_removal.txt deleted file mode 100644 index bf04380111..0000000000 --- a/doc/UPGRADE-staging/app_url_removal.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: app_url -Master-Only: True - -This module was deprecated in Asterisk 16 -and is now being removed in accordance with -the Asterisk Module Deprecation policy. diff --git a/doc/UPGRADE-staging/asterisk_logrotate.txt b/doc/UPGRADE-staging/asterisk_logrotate.txt deleted file mode 100644 index 2191e51f79..0000000000 --- a/doc/UPGRADE-staging/asterisk_logrotate.txt +++ /dev/null @@ -1,9 +0,0 @@ -Subject: Log Rotate -Master-Only: True - -The sample logger files have been changed to have .log as their file -extension. This was done so that when attached to issues on the issue -tracker, they are able to be opened in the browser for convenience. -Because of this, the asterisk.logrotate script has been updated to look -for .log extensions instead of no extension for files such as full -and messages. diff --git a/doc/UPGRADE-staging/cdr_mysql_removal.txt b/doc/UPGRADE-staging/cdr_mysql_removal.txt deleted file mode 100644 index a90690eef7..0000000000 --- a/doc/UPGRADE-staging/cdr_mysql_removal.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: cdr_mysql -Master-Only: True - -This module was deprecated in Asterisk 1.8 -and is now being removed in accordance with -the Asterisk Module Deprecation policy. diff --git a/doc/UPGRADE-staging/cdr_syslog_removal.txt b/doc/UPGRADE-staging/cdr_syslog_removal.txt deleted file mode 100644 index 17f88ddac2..0000000000 --- a/doc/UPGRADE-staging/cdr_syslog_removal.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: cdr_syslog -Master-Only: True - -This module was deprecated in Asterisk 16 -and is now being removed in accordance with -the Asterisk Module Deprecation policy. diff --git a/doc/UPGRADE-staging/chan_alsa_deprecation.txt b/doc/UPGRADE-staging/chan_alsa_deprecation.txt deleted file mode 100644 index 04edcf0073..0000000000 --- a/doc/UPGRADE-staging/chan_alsa_deprecation.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: chan_alsa -Master-Only: True - -This module is now deprecated and will no -longer be built by default. It is scheduled -to be removed as of Asterisk 21. diff --git a/doc/UPGRADE-staging/chan_iax2_rsa.txt b/doc/UPGRADE-staging/chan_iax2_rsa.txt deleted file mode 100644 index d5a9770862..0000000000 --- a/doc/UPGRADE-staging/chan_iax2_rsa.txt +++ /dev/null @@ -1,15 +0,0 @@ -Subject: chan_iax2 - -Encryption is now supported for RSA authentication. - -Currently, these auth configurations will cause a crash: -auth = md5,rsa -auth = plaintext,md5,rsa - -With a patched peer, the following will cause a crash: -auth = rsa -auth = md5,rsa -auth = plaintext,md5,rsa - -If both the peer and user are patches, no crash occurs. -Existing good configurations should continue to work. diff --git a/doc/UPGRADE-staging/chan_mgcp_deprecation.txt b/doc/UPGRADE-staging/chan_mgcp_deprecation.txt deleted file mode 100644 index 1d0d592978..0000000000 --- a/doc/UPGRADE-staging/chan_mgcp_deprecation.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: chan_mgcp -Master-Only: True - -This module is now deprecated and will no -longer be built by default. It is scheduled -to be removed as of Asterisk 21. diff --git a/doc/UPGRADE-staging/chan_misdn_removal.txt b/doc/UPGRADE-staging/chan_misdn_removal.txt deleted file mode 100644 index bb597dc233..0000000000 --- a/doc/UPGRADE-staging/chan_misdn_removal.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: chan_misdn -Master-Only: True - -This module was deprecated in Asterisk 16 -and is now being removed in accordance with -the Asterisk Module Deprecation policy. diff --git a/doc/UPGRADE-staging/chan_nbs_removal.txt b/doc/UPGRADE-staging/chan_nbs_removal.txt deleted file mode 100644 index 46ae75839f..0000000000 --- a/doc/UPGRADE-staging/chan_nbs_removal.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: chan_nbs -Master-Only: True - -This module was deprecated in Asterisk 16 -and is now being removed in accordance with -the Asterisk Module Deprecation policy. diff --git a/doc/UPGRADE-staging/chan_oss_removal.txt b/doc/UPGRADE-staging/chan_oss_removal.txt deleted file mode 100644 index 062f64b81f..0000000000 --- a/doc/UPGRADE-staging/chan_oss_removal.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: chan_oss -Master-Only: True - -This module was deprecated in Asterisk 16 -and is now being removed in accordance with -the Asterisk Module Deprecation policy. diff --git a/doc/UPGRADE-staging/chan_phone_removal.txt b/doc/UPGRADE-staging/chan_phone_removal.txt deleted file mode 100644 index 76135beff5..0000000000 --- a/doc/UPGRADE-staging/chan_phone_removal.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: chan_phone -Master-Only: True - -This module was deprecated in Asterisk 16 -and is now being removed in accordance with -the Asterisk Module Deprecation policy. diff --git a/doc/UPGRADE-staging/chan_skinny_deprecation.txt b/doc/UPGRADE-staging/chan_skinny_deprecation.txt deleted file mode 100644 index 0f840fb958..0000000000 --- a/doc/UPGRADE-staging/chan_skinny_deprecation.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: chan_skinny -Master-Only: True - -This module is now deprecated and will no -longer be built by default. It is scheduled -to be removed as of Asterisk 21. diff --git a/doc/UPGRADE-staging/chan_vpb_removal.txt b/doc/UPGRADE-staging/chan_vpb_removal.txt deleted file mode 100644 index 17feb89644..0000000000 --- a/doc/UPGRADE-staging/chan_vpb_removal.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: chan_vpb -Master-Only: True - -This module was deprecated in Asterisk 16 -and is now being removed in accordance with -the Asterisk Module Deprecation policy. diff --git a/doc/UPGRADE-staging/conf2ael_removal.txt b/doc/UPGRADE-staging/conf2ael_removal.txt deleted file mode 100644 index 73c300c43c..0000000000 --- a/doc/UPGRADE-staging/conf2ael_removal.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: conf2ael -Master-Only: True - -This application was deprecated in Asterisk 16 -and is now being removed in accordance with -the Asterisk Module Deprecation policy. diff --git a/doc/UPGRADE-staging/do-not-build-chan-sip-by-default.txt b/doc/UPGRADE-staging/do-not-build-chan-sip-by-default.txt deleted file mode 100644 index 31790e448d..0000000000 --- a/doc/UPGRADE-staging/do-not-build-chan-sip-by-default.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: chan_sip -Master-Only: True - -chan_sip is no longer built by default. To build it, make sure to -enable it when running 'make menuselect' diff --git a/doc/UPGRADE-staging/http-media-cache-lookup-order.txt b/doc/UPGRADE-staging/http-media-cache-lookup-order.txt deleted file mode 100644 index 83c31dcbcb..0000000000 --- a/doc/UPGRADE-staging/http-media-cache-lookup-order.txt +++ /dev/null @@ -1,9 +0,0 @@ -Subject: res_http_media_cache - -When fetching a file for playback from a URL, Asterisk will now first -use the value of the Content-Type header in the HTTP response to -determine the format of the audio data, and only if it is unable to do -that will it attempt to parse the URL and extract the extension from -the path portion. Previously Asterisk would first look at the end of -the URL, which may have included query string parameters or a URL -fragment, which was error prone. diff --git a/doc/UPGRADE-staging/menuselect-could-fail.txt b/doc/UPGRADE-staging/menuselect-could-fail.txt deleted file mode 100644 index e3e20ed833..0000000000 --- a/doc/UPGRADE-staging/menuselect-could-fail.txt +++ /dev/null @@ -1,5 +0,0 @@ -Subject: menuselect - -menuselect --enable, --disable, --enable-category and --disable-category will -now fail with a non-zero exit code instead of silently failing if an invalid -option or category is specified. diff --git a/doc/UPGRADE-staging/muted_removal.txt b/doc/UPGRADE-staging/muted_removal.txt deleted file mode 100644 index 5cca25cef9..0000000000 --- a/doc/UPGRADE-staging/muted_removal.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: muted -Master-Only: True - -This application was deprecated in Asterisk 16 -and is now being removed in accordance with -the Asterisk Module Deprecation policy. diff --git a/doc/UPGRADE-staging/res_config_sqlite_removal.txt b/doc/UPGRADE-staging/res_config_sqlite_removal.txt deleted file mode 100644 index 13c259d789..0000000000 --- a/doc/UPGRADE-staging/res_config_sqlite_removal.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: res_config_sqlite -Master-Only: True - -This module was deprecated in Asterisk 16 -and is now being removed in accordance with -the Asterisk Module Deprecation policy. diff --git a/doc/UPGRADE-staging/res_monitor_disabled.txt b/doc/UPGRADE-staging/res_monitor_disabled.txt deleted file mode 100644 index 12cc372f54..0000000000 --- a/doc/UPGRADE-staging/res_monitor_disabled.txt +++ /dev/null @@ -1,8 +0,0 @@ -Subject: res_monitor -Master-Only: True - -This module is no longer built by default in -accordance with the Module Deprecation Policy. -If you require this functionality you will need -to enable it for building in menuselect. Note -that in the future res_monitor will be removed. diff --git a/doc/UPGRADE-staging/res_pktccops_deprecation.txt b/doc/UPGRADE-staging/res_pktccops_deprecation.txt deleted file mode 100644 index 38acea1a03..0000000000 --- a/doc/UPGRADE-staging/res_pktccops_deprecation.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: res_pktccops -Master-Only: True - -This module is now deprecated and will no -longer be built by default. It is scheduled -to be removed as of Asterisk 21. diff --git a/doc/UPGRADE-staging/srtp_replay_protection.txt b/doc/UPGRADE-staging/srtp_replay_protection.txt deleted file mode 100644 index 945ddb5704..0000000000 --- a/doc/UPGRADE-staging/srtp_replay_protection.txt +++ /dev/null @@ -1,9 +0,0 @@ -Subject: res_srtp - -SRTP replay protection has been added to res_srtp and -a new configuration option "srtpreplayprotection" has -been added to the rtp.conf config file. For security -reasons, the default setting is "yes". Buggy clients -may not handle this correctly which could result in -no, or one way, audio and Asterisk error messages like -"replay check failed". diff --git a/doc/UPGRADE-staging/stir-shaken-public-key-url.txt b/doc/UPGRADE-staging/stir-shaken-public-key-url.txt deleted file mode 100644 index 094bccfe72..0000000000 --- a/doc/UPGRADE-staging/stir-shaken-public-key-url.txt +++ /dev/null @@ -1,6 +0,0 @@ -Subject: STIR/SHAKEN - -The configuration option public_key_url in stir_shaken.conf -has been renamed to public_cert_url to better fit what it -contains. Only the name has changed - functionality is the -same. diff --git a/doc/UPGRADE-staging/stir_shaken_origid.txt b/doc/UPGRADE-staging/stir_shaken_origid.txt deleted file mode 100644 index f0b897757f..0000000000 --- a/doc/UPGRADE-staging/stir_shaken_origid.txt +++ /dev/null @@ -1,8 +0,0 @@ -Subject: STIR/SHAKEN - -STIR/SHAKEN originally needed an origid to be specified in -stir_shaken.conf under the certificate config object in -order to work. Now, one is automatically created by -generating a UUID, as recommended by RFC8588. Any origid -you have in your stir_shaken.conf will need to be removed -for the module to read in certificates.