[asterisk-dev] asterisk release 21.0.0
The Asterisk Development Team would like to announce the release of asterisk-21.0.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/21.0.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You! Change Log for Release asterisk-21.0.0 Links: - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md) - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz) - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) Summary: - Update master branch for Asterisk 21 - translate.c: Prefer better codecs upon translate ties. - chan_skinny: Remove deprecated module. - app_osplookup: Remove deprecated module. - chan_mgcp: Remove deprecated module. - chan_alsa: Remove deprecated module. - pbx_builtins: Remove deprecated and defunct functionality. - chan_sip: Remove deprecated module. - app_cdr: Remove deprecated application and option. - app_macro: Remove deprecated module. - res_monitor: Remove deprecated module. - http.c: Minor simplification to HTTP status output. - app_osplookup: Remove obsolete sample config. - say.c: Fix French time playback. (#42) - core: Cleanup gerrit and JIRA references. (#58) - utils.h: Deprecate `ast_gethostbyname()`. (#79) - res_pjsip_pubsub: Add new pubsub module capabilities. (#82) - app_sla: Migrate SLA applications out of app_meetme. - rest-api: Ran make ari stubs to fix resource_endpoints inconsistency - .github: Update AsteriskReleaser for security releases - users.conf: Deprecate users.conf configuration. - Update version for Asterisk 21 - Remove unneeded CHANGES and UPGRADE files - res_pjsip_pubsub: Add body_type to test_handler for unit tests - ari-stubs: Fix more local anchor references - ari-stubs: Fix more local anchor references - ari-stubs: Fix broken documentation anchors - res_pjsip_session: Send Session Interval too small response - .github: Update workflow-application-token-action to v2 - app_dial: Fix infinite loop when sending digits. - app_voicemail: Fix for loop declarations - alembic: Fix quoting of the 100rel column - pbx.c: Fix gcc 12 compiler warning. - app_audiosocket: Fixed timeout with -1 to avoid busy loop. - download_externals: Fix a few version related issues - main/refer.c: Fix double free in refer_data_destructor + potential leak - sig_analog: Add Called Subscriber Held capability. - Revert "app_stack: Print proper exit location for PBXless channels." - install_prereq: Fix dependency install on aarch64. - res_pjsip.c: Set contact_user on incoming call local Contact header - extconfig: Allow explicit DB result set ordering to be disabled. - rest-api: Run make ari-stubs - res_pjsip_header_funcs: Make prefix argument optional. - pjproject_bundled: Increase PJSIP_MAX_MODULE to 38 - manager: Tolerate stasis messages with no channel snapshot. - Remove unneeded CHANGES and UPGRADE files User Notes: - ### sig_analog: Add Called Subscriber Held capability. Called Subscriber Held is now supported for analog FXS channels, using the calledsubscriberheld option. This allows a station user to go on hook when receiving an incoming call and resume from another phone on the same line by going on hook, without disconnecting the call. - ### res_pjsip_header_funcs: Make prefix argument optional. The prefix argument to PJSIP_HEADERS is now optional. If not specified, all header names will be returned. - ### http.c: Minor simplification to HTTP status output. For bound addresses, the HTTP status page now combines the bound address and bound port in a single line. Additionally, the SSL bind address has been renamed to TLS. Upgrade Notes: - ### utils.h: Deprecate `ast_gethostbyname()`. (#79) ast_gethostbyname() has been deprecated and will be removed in Asterisk 23. New code should use `ast_sockaddr_resolve()` and `ast_sockaddr_resolve_first_af()`. - ### app_sla: Migrate SLA applications out of app_meetme. The SLAStation and SLATrunk applications have been moved from app_meetme to app_sla. If you are using these applications and have autoload=no, you will need to explicitly load this module in modules.conf. - ### users.conf: Deprecate users.conf configuration. The users.conf config is now deprecated and will be removed in a future version of Asterisk. - ### res_monitor: Remove deprecated module. This module was deprecated in Asterisk 16 and is now being removed in accordance with the Asterisk Module Deprecation policy. This also
[asterisk-dev] asterisk release 21.0.0-rc1
The Asterisk Development Team would like to announce release candidate 1 of asterisk-21.0.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/21.0.0-rc1 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You! Change Log for Release asterisk-21.0.0-rc1 Links: - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0-rc1.md) - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0-rc1) - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0-rc1.tar.gz) - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) Summary: - Update master branch for Asterisk 21 - translate.c: Prefer better codecs upon translate ties. - chan_skinny: Remove deprecated module. - app_osplookup: Remove deprecated module. - chan_mgcp: Remove deprecated module. - chan_alsa: Remove deprecated module. - pbx_builtins: Remove deprecated and defunct functionality. - chan_sip: Remove deprecated module. - app_cdr: Remove deprecated application and option. - app_macro: Remove deprecated module. - res_monitor: Remove deprecated module. - http.c: Minor simplification to HTTP status output. - app_osplookup: Remove obsolete sample config. - say.c: Fix French time playback. (#42) - core: Cleanup gerrit and JIRA references. (#58) - utils.h: Deprecate `ast_gethostbyname()`. (#79) - res_pjsip_pubsub: Add new pubsub module capabilities. (#82) - app_sla: Migrate SLA applications out of app_meetme. - Update config.yml - rest-api: Ran make ari stubs to fix resource_endpoints inconsistency - .github: Update AsteriskReleaser for security releases - users.conf: Deprecate users.conf configuration. - Update version for Asterisk 21 - Remove unneeded CHANGES and UPGRADE files - ari-stubs: Fix more local anchor references - ari-stubs: Fix more local anchor references - ari-stubs: Fix broken documentation anchors - res_pjsip_session: Send Session Interval too small response - .github: Update workflow-application-token-action to v2 - app_dial: Fix infinite loop when sending digits. - app_voicemail: Fix for loop declarations - alembic: Fix quoting of the 100rel column - pbx.c: Fix gcc 12 compiler warning. - app_audiosocket: Fixed timeout with -1 to avoid busy loop. - download_externals: Fix a few version related issues - main/refer.c: Fix double free in refer_data_destructor + potential leak - sig_analog: Add Called Subscriber Held capability. - Revert "app_stack: Print proper exit location for PBXless channels." - install_prereq: Fix dependency install on aarch64. - res_pjsip.c: Set contact_user on incoming call local Contact header - extconfig: Allow explicit DB result set ordering to be disabled. - rest-api: Run make ari-stubs - res_pjsip_header_funcs: Make prefix argument optional. - pjproject_bundled: Increase PJSIP_MAX_MODULE to 38 - manager: Tolerate stasis messages with no channel snapshot. - Remove unneeded CHANGES and UPGRADE files User Notes: - ### sig_analog: Add Called Subscriber Held capability. Called Subscriber Held is now supported for analog FXS channels, using the calledsubscriberheld option. This allows a station user to go on hook when receiving an incoming call and resume from another phone on the same line by going on hook, without disconnecting the call. - ### res_pjsip_header_funcs: Make prefix argument optional. The prefix argument to PJSIP_HEADERS is now optional. If not specified, all header names will be returned. - ### http.c: Minor simplification to HTTP status output. For bound addresses, the HTTP status page now combines the bound address and bound port in a single line. Additionally, the SSL bind address has been renamed to TLS. Upgrade Notes: - ### utils.h: Deprecate `ast_gethostbyname()`. (#79) ast_gethostbyname() has been deprecated and will be removed in Asterisk 23. New code should use `ast_sockaddr_resolve()` and `ast_sockaddr_resolve_first_af()`. - ### app_sla: Migrate SLA applications out of app_meetme. The SLAStation and SLATrunk applications have been moved from app_meetme to app_sla. If you are using these applications and have autoload=no, you will need to explicitly load this module in modules.conf. - ### users.conf: Deprecate users.conf configuration. The users.conf config is now deprecated and will be removed in a future version of Asterisk. - ### app_osplookup: Remove deprecated module. This module was deprecated in Asterisk 19 and is now being removed in accordance with the Asterisk Module Deprecation policy. - ### res_monitor: Remove
