[asterisk-users] 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-users] asterisk release 20.5.0
The Asterisk Development Team would like to announce the release of asterisk-20.5.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/20.5.0 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-20.5.0 Links: - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md) - [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz) - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) Summary: - 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. - app_macro: Fix locking around datastore access - Revert "app_stack: Print proper exit location for PBXless channels." - .github: Use generic releaser - 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. - core/ari/pjsip: Add refer mechanism - chan_dahdi: Allow autoreoriginating after hangup. - audiohook: Unlock channel in mute if no audiohooks present. - sig_analog: Allow three-way flash to time out to silence. - res_prometheus: Do not generate broken metrics - res_pjsip: Enable TLS v1.3 if present. - func_cut: Add example to documentation. - extensions.conf.sample: Remove reference to missing context. - func_export: Use correct function argument as variable name. - app_queue: Add support for applying caller priority change immediately. - .github: Fix cherry-pick reminder issues - chan_iax2.c: Avoid crash with IAX2 switch support. - res_geolocation: Ensure required 'location_info' is present. - Adds manager actions to allow move/remove/forward individual messages in a particular mailbox folder. The forward command can be used to copy a message within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry, required to retrieve message ID's. - app_voicemail: add CLI commands for message manipulation - res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp->themssrc_valid` into the scope of the rtp_instance lock. - .github: Minor tweak to Asterisk Releaser - .github: Suppress cherry-pick reminder for some situations - sig_analog: Allow immediate fake ring to be suppressed. 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. - ### core/ari/pjsip: Add refer mechanism There is a new ARI endpoint `/endpoints/refer` for referring an endpoint to some URI or endpoint. - ### chan_dahdi: Allow autoreoriginating after hangup. The autoreoriginate setting now allows for kewlstart FXS channels to automatically reoriginate and provide dial tone to the user again after all calls on the line have cleared. This saves users from having to manually hang up and pick up the receiver again before making another call. - ### sig_analog: Allow three-way flash to time out to silence. The threewaysilenthold option now allows the three-way dial tone to time out to silence, rather than continuing forever. - ### res_pjsip: Enable TLS v1.3 if present. res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying PJSIP library. The bundled version of PJSIP supports TLS v1.3. - ### app_queue: Add support for applying caller priority change
[asterisk-users] asterisk release 18.20.0
The Asterisk Development Team would like to announce the release of asterisk-18.20.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/18.20.0 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-18.20.0 Links: - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md) - [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0) - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz) - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) Summary: - 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. - app_macro: Fix locking around datastore access - Revert "app_stack: Print proper exit location for PBXless channels." - .github: Use generic releaser - 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. - core/ari/pjsip: Add refer mechanism - chan_dahdi: Allow autoreoriginating after hangup. - audiohook: Unlock channel in mute if no audiohooks present. - sig_analog: Allow three-way flash to time out to silence. - res_prometheus: Do not generate broken metrics - res_pjsip: Enable TLS v1.3 if present. - func_cut: Add example to documentation. - extensions.conf.sample: Remove reference to missing context. - func_export: Use correct function argument as variable name. - app_queue: Add support for applying caller priority change immediately. - .github: Fix cherry-pick reminder issues - chan_iax2.c: Avoid crash with IAX2 switch support. - res_geolocation: Ensure required 'location_info' is present. - Adds manager actions to allow move/remove/forward individual messages in a particular mailbox folder. The forward command can be used to copy a message within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry, required to retrieve message ID's. - app_voicemail: add CLI commands for message manipulation - res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp->themssrc_valid` into the scope of the rtp_instance lock. - .github: Minor tweak to Asterisk Releaser - .github: Suppress cherry-pick reminder for some situations - sig_analog: Allow immediate fake ring to be suppressed. 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. - ### core/ari/pjsip: Add refer mechanism There is a new ARI endpoint `/endpoints/refer` for referring an endpoint to some URI or endpoint. - ### chan_dahdi: Allow autoreoriginating after hangup. The autoreoriginate setting now allows for kewlstart FXS channels to automatically reoriginate and provide dial tone to the user again after all calls on the line have cleared. This saves users from having to manually hang up and pick up the receiver again before making another call. - ### sig_analog: Allow three-way flash to time out to silence. The threewaysilenthold option now allows the three-way dial tone to time out to silence, rather than continuing forever. - ### res_pjsip: Enable TLS v1.3 if present. res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying PJSIP library. The bundled version of PJSIP supports TLS v1.3. - ### app_queue: Add support for applying caller priority change
