mirror of https://github.com/asterisk/asterisk
Asterisk Development Team
2 years ago
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ChangeLogs/ChangeLog-21.0.0-rc1.md
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Change Log for Release asterisk-21.0.0-rc1
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========================================
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Links:
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----------------------------------------
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- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0-rc1.md)
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- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0-rc1)
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- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0-rc1.tar.gz)
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- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
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Summary:
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----------------------------------------
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- Update master branch for Asterisk 21
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- translate.c: Prefer better codecs upon translate ties.
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- chan_skinny: Remove deprecated module.
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- app_osplookup: Remove deprecated module.
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- chan_mgcp: Remove deprecated module.
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- chan_alsa: Remove deprecated module.
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- pbx_builtins: Remove deprecated and defunct functionality.
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- chan_sip: Remove deprecated module.
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- app_cdr: Remove deprecated application and option.
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- app_macro: Remove deprecated module.
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- res_monitor: Remove deprecated module.
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- http.c: Minor simplification to HTTP status output.
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- app_osplookup: Remove obsolete sample config.
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- say.c: Fix French time playback. (#42)
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- core: Cleanup gerrit and JIRA references. (#58)
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- utils.h: Deprecate `ast_gethostbyname()`. (#79)
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- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
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- app_sla: Migrate SLA applications out of app_meetme.
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- Update config.yml
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- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
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- .github: Update AsteriskReleaser for security releases
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- users.conf: Deprecate users.conf configuration.
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- Update version for Asterisk 21
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- Remove unneeded CHANGES and UPGRADE files
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- ari-stubs: Fix more local anchor references
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- ari-stubs: Fix more local anchor references
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- ari-stubs: Fix broken documentation anchors
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- res_pjsip_session: Send Session Interval too small response
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- .github: Update workflow-application-token-action to v2
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- app_dial: Fix infinite loop when sending digits.
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- app_voicemail: Fix for loop declarations
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- alembic: Fix quoting of the 100rel column
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- pbx.c: Fix gcc 12 compiler warning.
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- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
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- download_externals: Fix a few version related issues
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- main/refer.c: Fix double free in refer_data_destructor + potential leak
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- sig_analog: Add Called Subscriber Held capability.
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- Revert "app_stack: Print proper exit location for PBXless channels."
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- install_prereq: Fix dependency install on aarch64.
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- res_pjsip.c: Set contact_user on incoming call local Contact header
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- extconfig: Allow explicit DB result set ordering to be disabled.
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- rest-api: Run make ari-stubs
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- res_pjsip_header_funcs: Make prefix argument optional.
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- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
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- manager: Tolerate stasis messages with no channel snapshot.
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- Remove unneeded CHANGES and UPGRADE files
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User Notes:
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----------------------------------------
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- ### sig_analog: Add Called Subscriber Held capability.
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Called Subscriber Held is now supported for analog
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FXS channels, using the calledsubscriberheld option. This allows
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a station user to go on hook when receiving an incoming call
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and resume from another phone on the same line by going on hook,
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without disconnecting the call.
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- ### res_pjsip_header_funcs: Make prefix argument optional.
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The prefix argument to PJSIP_HEADERS is now
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optional. If not specified, all header names will be
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returned.
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- ### http.c: Minor simplification to HTTP status output.
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For bound addresses, the HTTP status page now combines the bound
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address and bound port in a single line. Additionally, the SSL bind
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address has been renamed to TLS.
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Upgrade Notes:
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----------------------------------------
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- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
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ast_gethostbyname() has been deprecated and will be removed
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in Asterisk 23. New code should use `ast_sockaddr_resolve()` and
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`ast_sockaddr_resolve_first_af()`.
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- ### app_sla: Migrate SLA applications out of app_meetme.
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The SLAStation and SLATrunk applications have been moved
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from app_meetme to app_sla. If you are using these applications and have
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autoload=no, you will need to explicitly load this module in modules.conf.
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- ### users.conf: Deprecate users.conf configuration.
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The users.conf config is now deprecated
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and will be removed in a future version of Asterisk.
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- ### app_osplookup: Remove deprecated module.
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This module was deprecated in Asterisk 19
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and is now being removed in accordance with
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the Asterisk Module Deprecation policy.
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- ### res_monitor: Remove deprecated module.
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This module was deprecated in Asterisk 16
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and is now being removed in accordance with
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the Asterisk Module Deprecation policy.
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This also removes the 'w' and 'W' options
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for app_queue.
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MixMonitor should be default and only option
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for all settings that previously used either
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Monitor or MixMonitor.
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- ### chan_sip: Remove deprecated module.
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This module was deprecated in Asterisk 17
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and is now being removed in accordance with
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the Asterisk Module Deprecation policy.
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- ### chan_alsa: Remove deprecated module.
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This module was deprecated in Asterisk 19
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and is now being removed in accordance with
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the Asterisk Module Deprecation policy.
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- ### chan_mgcp: Remove deprecated module.
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This module was deprecated in Asterisk 19
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and is now being removed in accordance with
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the Asterisk Module Deprecation policy.
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- ### chan_skinny: Remove deprecated module.
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This module was deprecated in Asterisk 19
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and is now being removed in accordance with
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the Asterisk Module Deprecation policy.
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- ### app_macro: Remove deprecated module.
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This module was deprecated in Asterisk 16
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and is now being removed in accordance with
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the Asterisk Module Deprecation policy.
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For most modules that interacted with app_macro,
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this change is limited to no longer looking for
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the current context from the macrocontext when set.
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The following modules have additional impacts:
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app_dial - no longer supports M^ connected/redirecting macro
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app_minivm - samples written using macro will no longer work.
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The sample needs to be re-written
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app_queue - can no longer call a macro on the called party's
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channel. Use gosub which is currently supported
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ccss - no callback macro, gosub only
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app_voicemail - no macro support
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channel - remove macrocontext and priority, no connected
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line or redirection macro options
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options - stdexten is deprecated to gosub as the default
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and only options
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pbx - removed macrolock
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pbx_dundi - no longer look for macro
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snmp - removed macro context, exten, and priority
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- ### pbx_builtins: Remove deprecated and defunct functionality.
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The previously deprecated ImportVar and SetAMAFlags
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applications have now been removed.
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- ### translate.c: Prefer better codecs upon translate ties.
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When setting up translation between two codecs the quality was not taken into account,
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resulting in suboptimal translation. The quality is now taken into account,
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which can reduce the number of translation steps required, and improve the resulting quality.
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- ### app_cdr: Remove deprecated application and option.
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The previously deprecated NoCDR application has been removed.
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Additionally, the previously deprecated 'e' option to the ResetCDR
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application has been removed.
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Closed Issues:
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----------------------------------------
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- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on aarch64 Debian platforms
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- #39: [Bug]: Remove .gitreview from repository.
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- #41: [Bug]: say.c Time announcement does not say o'clock for the French language
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- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
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- #78: [improvement]: Deprecate ast_gethostbyname()
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- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
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- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
|
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- #183: [deprecation]: Deprecate users.conf
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- #226: [improvement]: Apply contact_user to incoming calls
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- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
|
||||||
|
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
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- #253: app_gosub patch appear to have broken predial handlers that utilize macros that call gosubs
|
||||||
|
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules registered"
|
||||||
|
- #263: [bug]: download_externals doesn't always handle versions correctly
|
||||||
|
- #267: [bug]: ari: refer with display_name key in request body leads to crash
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||||||
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- #274: [bug]: Syntax Error in SQL Code
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- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99 -Wdeclaration-after-statement'
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||||||
|
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
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- #281: [bug]: app_dial: Infinite loop if called channel hangs up while receiving digits
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|
|
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|
Commits By Author:
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|
----------------------------------------
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|
|
||||||
|
- ### Bastian Triller (1):
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|
- res_pjsip_session: Send Session Interval too small response
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|
|
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|
- ### George Joseph (9):
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||||||
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- Remove unneeded CHANGES and UPGRADE files
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||||||
|
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
|
||||||
|
- rest-api: Run make ari-stubs
|
||||||
|
- download_externals: Fix a few version related issues
|
||||||
|
- alembic: Fix quoting of the 100rel column
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||||||
|
- .github: Update workflow-application-token-action to v2
|
||||||
|
- ari-stubs: Fix broken documentation anchors
|
||||||
|
- ari-stubs: Fix more local anchor references
|
||||||
|
- ari-stubs: Fix more local anchor references
|
||||||
|
|
||||||
|
- ### Jason D. McCormick (1):
|
||||||
|
- install_prereq: Fix dependency install on aarch64.
|
||||||
|
|
||||||
|
- ### Joshua C. Colp (1):
|
||||||
|
- manager: Tolerate stasis messages with no channel snapshot.
|
||||||
|
|
||||||
|
- ### Matthew Fredrickson (1):
|
||||||
|
- Revert "app_stack: Print proper exit location for PBXless channels."
|
||||||
|
|
||||||
|
- ### Maximilian Fridrich (1):
|
||||||
|
- main/refer.c: Fix double free in refer_data_destructor + potential leak
|
||||||
|
|
||||||
|
- ### Mike Bradeen (1):
|
||||||
|
- app_voicemail: Fix for loop declarations
|
||||||
|
|
||||||
|
- ### MikeNaso (1):
|
||||||
|
- res_pjsip.c: Set contact_user on incoming call local Contact header
|
||||||
|
|
||||||
|
- ### Naveen Albert (4):
|
||||||
|
- res_pjsip_header_funcs: Make prefix argument optional.
|
||||||
|
- sig_analog: Add Called Subscriber Held capability.
|
||||||
|
- pbx.c: Fix gcc 12 compiler warning.
|
||||||
|
- app_dial: Fix infinite loop when sending digits.
|
||||||
|
|
||||||
|
- ### Sean Bright (1):
|
||||||
|
- extconfig: Allow explicit DB result set ordering to be disabled.
|
||||||
|
|
||||||
|
- ### zhengsh (1):
|
||||||
|
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
|
||||||
|
|
||||||
|
|
||||||
|
Detail:
|
||||||
|
----------------------------------------
|
||||||
|
|
||||||
|
- ### Update master branch for Asterisk 21
|
||||||
|
Author: George Joseph
|
||||||
|
Date: 2022-07-20
|
||||||
|
|
||||||
|
|
||||||
|
- ### translate.c: Prefer better codecs upon translate ties.
|
||||||
|
Author: Naveen Albert
|
||||||
|
Date: 2021-05-27
|
||||||
|
|
||||||
|
If multiple codecs are available for the same
|
||||||
|
resource and the translation costs between
|
||||||
|
multiple codecs are the same, ties are
|
||||||
|
currently broken arbitrarily, which means a
|
||||||
|
lower quality codec would be used. This forces
|
||||||
|
Asterisk to explicitly use the higher quality
|
||||||
|
codec, ceteris paribus.
|
||||||
|
|
||||||
|
ASTERISK-29455
|
||||||
|
|
||||||
|
|
||||||
|
- ### chan_skinny: Remove deprecated module.
|
||||||
|
Author: Mike Bradeen
|
||||||
|
Date: 2022-11-16
|
||||||
|
|
||||||
|
ASTERISK-30300
|
||||||
|
|
||||||
|
|
||||||
|
- ### app_osplookup: Remove deprecated module.
|
||||||
|
Author: Mike Bradeen
|
||||||
|
Date: 2022-11-18
|
||||||
|
|
||||||
|
ASTERISK-30302
|
||||||
|
|
||||||
|
|
||||||
|
- ### chan_mgcp: Remove deprecated module.
|
||||||
|
Author: Mike Bradeen
|
||||||
|
Date: 2022-11-15
|
||||||
|
|
||||||
|
Also removes res_pktcops to avoid merge conflicts
|
||||||
|
with ASTERISK~30301.
|
||||||
|
|
||||||
|
ASTERISK-30299
|
||||||
|
|
||||||
|
|
||||||
|
- ### chan_alsa: Remove deprecated module.
|
||||||
|
Author: Mike Bradeen
|
||||||
|
Date: 2022-11-14
|
||||||
|
|
||||||
|
ASTERISK-30298
|
||||||
|
|
||||||
|
|
||||||
|
- ### pbx_builtins: Remove deprecated and defunct functionality.
|
||||||
|
Author: Naveen Albert
|
||||||
|
Date: 2022-11-29
|
||||||
|
|
||||||
|
This removes the ImportVar and SetAMAFlags applications
|
||||||
|
which have been deprecated since Asterisk 12, but were
|
||||||
|
never removed previously.
|
||||||
|
|
||||||
|
Additionally, it removes remnants of defunct options
|
||||||
|
that themselves were removed years ago.
|
||||||
|
|
||||||
|
ASTERISK-30335 #close
|
||||||
|
|
||||||
|
|
||||||
|
- ### chan_sip: Remove deprecated module.
|
||||||
|
Author: Mike Bradeen
|
||||||
|
Date: 2022-11-28
|
||||||
|
|
||||||
|
ASTERISK-30297
|
||||||
|
|
||||||
|
|
||||||
|
- ### app_cdr: Remove deprecated application and option.
|
||||||
|
Author: Naveen Albert
|
||||||
|
Date: 2022-12-22
|
||||||
|
|
||||||
|
This removes the deprecated NoCDR application, which
|
||||||
|
was deprecated in Asterisk 12, having long been fully
|
||||||
|
superseded by the CDR_PROP function.
|
||||||
|
|
||||||
|
The deprecated e option to ResetCDR is also removed
|
||||||
|
for the same reason.
|
||||||
|
|
||||||
|
ASTERISK-30371 #close
|
||||||
|
|
||||||
|
|
||||||
|
- ### app_macro: Remove deprecated module.
|
||||||
|
Author: Mike Bradeen
|
||||||
|
Date: 2022-12-12
|
||||||
|
|
||||||
|
For most modules that interacted with app_macro, this change is limited
|
||||||
|
to no longer looking for the current context from the macrocontext when
|
||||||
|
set. Additionally, the following modules are impacted:
|
||||||
|
|
||||||
|
app_dial - no longer supports M^ connected/redirecting macro
|
||||||
|
app_minivm - samples written using macro will no longer work.
|
||||||
|
The sample needs a re-write
|
||||||
|
|
||||||
|
app_queue - can no longer a macro on the called party's channel.
|
||||||
|
Use gosub which is currently supported
|
||||||
|
|
||||||
|
ccss - no callback macro, gosub only
|
||||||
|
|
||||||
|
app_voicemail - no macro support
|
||||||
|
|
||||||
|
channel - remove macrocontext and priority, no connected line or
|
||||||
|
redirection macro options
|
||||||
|
options - stdexten is deprecated to gosub as the default and only
|
||||||
|
pbx - removed macrolock
|
||||||
|
pbx_dundi - no longer look for macro
|
||||||
|
|
||||||
|
snmp - removed macro context, exten, and priority
|
||||||
|
|
||||||
|
ASTERISK-30304
|
||||||
|
|
||||||
|
|
||||||
|
- ### res_monitor: Remove deprecated module.
|
||||||
|
Author: Mike Bradeen
|
||||||
|
Date: 2022-11-18
|
||||||
|
|
||||||
|
ASTERISK-30303
|
||||||
|
|
||||||
|
|
||||||
|
- ### http.c: Minor simplification to HTTP status output.
|
||||||
|
Author: Boris P. Korzun
|
||||||
|
Date: 2023-01-05
|
||||||
|
|
||||||
|
Change the HTTP status page (located at /httpstatus by default) by:
|
||||||
|
|
||||||
|
* Combining the address and port into a single line.
|
||||||
|
* Changing "SSL" to "TLS"
|
||||||
|
|
||||||
|
ASTERISK-30433 #close
|
||||||
|
|
||||||
|
|
||||||
|
- ### app_osplookup: Remove obsolete sample config.
|
||||||
|
Author: Naveen Albert
|
||||||
|
Date: 2023-02-24
|
||||||
|
|
||||||
|
ASTERISK_30302 previously removed app_osplookup,
|
||||||
|
but its sample config was not removed.
|
||||||
|
This removes it since nothing else uses it.
|
||||||
|
|
||||||
|
ASTERISK-30438 #close
|
||||||
|
|
||||||
|
|
||||||
|
- ### say.c: Fix French time playback. (#42)
|
||||||
|
Author: InterLinked1
|
||||||
|
Date: 2023-05-02
|
||||||
|
|
||||||
|
ast_waitstream was not called after ast_streamfile,
|
||||||
|
resulting in "o'clock" being skipped in French.
|
||||||
|
|
||||||
|
Additionally, the minute announcements should be
|
||||||
|
feminine.
|
||||||
|
|
||||||
|
Reported-by: Danny Lloyd
|
||||||
|
|
||||||
|
Resolves: #41
|
||||||
|
ASTERISK-30488
|
||||||
|
- ### core: Cleanup gerrit and JIRA references. (#58)
|
||||||
|
Author: Sean Bright
|
||||||
|
Date: 2023-05-03
|
||||||
|
|
||||||
|
* Remove .gitreview and switch to pulling the main asterisk branch
|
||||||
|
version from configure.ac instead.
|
||||||
|
|
||||||
|
* Replace references to JIRA with GitHub.
|
||||||
|
|
||||||
|
* Other minor cleanup found along the way.
|
||||||
|
|
||||||
|
Resolves: #39
|
||||||
|
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
|
||||||
|
Author: Sean Bright
|
||||||
|
Date: 2023-05-11
|
||||||
|
|
||||||
|
Deprecate `ast_gethostbyname()` in favor of `ast_sockaddr_resolve()` and
|
||||||
|
`ast_sockaddr_resolve_first_af()`. `ast_gethostbyname()` has not been
|
||||||
|
used by any in-tree code since 2021.
|
||||||
|
|
||||||
|
This function will be removed entirely in Asterisk 23.
|
||||||
|
|
||||||
|
Resolves: #78
|
||||||
|
|
||||||
|
UpgradeNote: 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()`.
|
||||||
|
- ### res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
|
||||||
|
Author: InterLinked1
|
||||||
|
Date: 2023-05-18
|
||||||
|
|
||||||
|
The existing res_pjsip_pubsub APIs are somewhat limited in
|
||||||
|
what they can do. This adds a few API extensions that make
|
||||||
|
it possible for PJSIP pubsub modules to implement richer
|
||||||
|
features than is currently possible.
|
||||||
|
|
||||||
|
* Allow pubsub modules to get a handle to pjsip_rx_data on subscription
|
||||||
|
* Allow pubsub modules to run a callback when a subscription is renewed
|
||||||
|
* Allow pubsub modules to run a callback for outgoing NOTIFYs, with
|
||||||
|
a handle to the tdata, so that modules can append their own headers
|
||||||
|
to the NOTIFYs
|
||||||
|
|
||||||
|
This change does not add any features directly, but makes possible
|
||||||
|
several new features that will be added in future changes.
|
||||||
|
|
||||||
|
Resolves: #81
|
||||||
|
ASTERISK-30485 #close
|
||||||
|
|
||||||
|
Master-Only: True
|
||||||
|
- ### app_sla: Migrate SLA applications out of app_meetme.
|
||||||
|
Author: Naveen Albert
|
||||||
|
Date: 2023-05-02
|
||||||
|
|
||||||
|
This removes the dependency of the SLAStation and SLATrunk
|
||||||
|
applications on app_meetme, in anticipation of the imminent
|
||||||
|
removal of the deprecated app_meetme module.
|
||||||
|
|
||||||
|
The user interface for the SLA applications is exactly the
|
||||||
|
same, and in theory, users should not notice a difference.
|
||||||
|
However, the SLA applications now use ConfBridge under the
|
||||||
|
hood, rather than MeetMe, and they are now contained within
|
||||||
|
their own module.
|
||||||
|
|
||||||
|
Resolves: #50
|
||||||
|
ASTERISK-30309
|
||||||
|
|
||||||
|
UpgradeNote: 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.
|
||||||
|
|
||||||
|
- ### Update config.yml
|
||||||
|
Author: Joshua C. Colp
|
||||||
|
Date: 2023-06-15
|
||||||
|
|
||||||
|
|
||||||
|
- ### rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
|
||||||
|
Author: George Joseph
|
||||||
|
Date: 2023-06-27
|
||||||
|
|
||||||
|
|
||||||
|
- ### .github: Update AsteriskReleaser for security releases
|
||||||
|
Author: George Joseph
|
||||||
|
Date: 2023-07-07
|
||||||
|
|
||||||
|
|
||||||
|
- ### users.conf: Deprecate users.conf configuration.
|
||||||
|
Author: Naveen Albert
|
||||||
|
Date: 2023-06-30
|
||||||
|
|
||||||
|
This deprecates the users.conf config file, which
|
||||||
|
is no longer as widely supported but still integrated
|
||||||
|
with a number of different modules.
|
||||||
|
|
||||||
|
Because there is no real mechanism for marking a
|
||||||
|
configuration file as "deprecated", and users.conf
|
||||||
|
is not just used in a single place, this now emits
|
||||||
|
a warning to the user when the PBX loads to notify
|
||||||
|
about the deprecation.
|
||||||
|
|
||||||
|
This configuration mechanism has been widely criticized
|
||||||
|
and discouraged since its inception, and is no longer
|
||||||
|
relevant to the configuration that most users are doing
|
||||||
|
today. Removing it will allow for some simplification
|
||||||
|
and cleanup in the codebase.
|
||||||
|
|
||||||
|
Resolves: #183
|
||||||
|
|
||||||
|
UpgradeNote: The users.conf config is now deprecated
|
||||||
|
and will be removed in a future version of Asterisk.
|
||||||
|
|
||||||
|
- ### Update version for Asterisk 21
|
||||||
|
Author: George Joseph
|
||||||
|
Date: 2023-08-09
|
||||||
|
|
||||||
|
|
||||||
|
- ### Remove unneeded CHANGES and UPGRADE files
|
||||||
|
Author: George Joseph
|
||||||
|
Date: 2023-08-09
|
||||||
|
|
||||||
|
|
||||||
|
- ### ari-stubs: Fix more local anchor references
|
||||||
|
Author: George Joseph
|
||||||
|
Date: 2023-09-05
|
||||||
|
|
||||||
|
Also allow CreateDocs job to be run manually with default branches.
|
||||||
|
|
||||||
|
|
||||||
|
- ### ari-stubs: Fix more local anchor references
|
||||||
|
Author: George Joseph
|
||||||
|
Date: 2023-09-05
|
||||||
|
|
||||||
|
Also allow CreateDocs job to be run manually with default branches.
|
||||||
|
|
||||||
|
|
||||||
|
- ### ari-stubs: Fix broken documentation anchors
|
||||||
|
Author: George Joseph
|
||||||
|
Date: 2023-09-05
|
||||||
|
|
||||||
|
All of the links that reference page anchors with capital letters in
|
||||||
|
the ids (#Something) have been changed to lower case to match the
|
||||||
|
anchors that are generated by mkdocs.
|
||||||
|
|
||||||
|
|
||||||
|
- ### res_pjsip_session: Send Session Interval too small response
|
||||||
|
Author: Bastian Triller
|
||||||
|
Date: 2023-08-28
|
||||||
|
|
||||||
|
Handle session interval lower than endpoint's configured minimum timer
|
||||||
|
when sending first answer. Timer setting is checked during this step and
|
||||||
|
needs to handled appropriately.
|
||||||
|
Before this change, no response was sent at all. After this change a
|
||||||
|
response with 422 Session Interval too small is sent to UAC.
|
||||||
|
|
||||||
|
|
||||||
|
- ### .github: Update workflow-application-token-action to v2
|
||||||
|
Author: George Joseph
|
||||||
|
Date: 2023-08-31
|
||||||
|
|
||||||
|
|
||||||
|
- ### app_dial: Fix infinite loop when sending digits.
|
||||||
|
Author: Naveen Albert
|
||||||
|
Date: 2023-08-28
|
||||||
|
|
||||||
|
If the called party hangs up while digits are being
|
||||||
|
sent, -1 is returned to indicate so, but app_dial
|
||||||
|
was not checking the return value, resulting in
|
||||||
|
the hangup being lost and looping forever until
|
||||||
|
the caller manually hangs up the channel. We now
|
||||||
|
abort if digit sending fails.
|
||||||
|
|
||||||
|
ASTERISK-29428 #close
|
||||||
|
|
||||||
|
Resolves: #281
|
||||||
|
|
||||||
|
- ### app_voicemail: Fix for loop declarations
|
||||||
|
Author: Mike Bradeen
|
||||||
|
Date: 2023-08-29
|
||||||
|
|
||||||
|
Resolve for loop initial declarations added in cli changes.
|
||||||
|
|
||||||
|
Resolves: #275
|
||||||
|
|
||||||
|
- ### alembic: Fix quoting of the 100rel column
|
||||||
|
Author: George Joseph
|
||||||
|
Date: 2023-08-28
|
||||||
|
|
||||||
|
Add quoting around the ps_endpoints 100rel column in the ALTER
|
||||||
|
statements. Although alembic doesn't complain when generating
|
||||||
|
sql statements, postgresql does (rightly so).
|
||||||
|
|
||||||
|
Resolves: #274
|
||||||
|
|
||||||
|
- ### pbx.c: Fix gcc 12 compiler warning.
|
||||||
|
Author: Naveen Albert
|
||||||
|
Date: 2023-08-27
|
||||||
|
|
||||||
|
Resolves: #277
|
||||||
|
|
||||||
|
- ### app_audiosocket: Fixed timeout with -1 to avoid busy loop.
|
||||||
|
Author: zhengsh
|
||||||
|
Date: 2023-08-24
|
||||||
|
|
||||||
|
Resolves: asterisk#234
|
||||||
|
|
||||||
|
- ### download_externals: Fix a few version related issues
|
||||||
|
Author: George Joseph
|
||||||
|
Date: 2023-08-18
|
||||||
|
|
||||||
|
* Fixed issue with the script not parsing the new tag format for
|
||||||
|
certified releases. The format changed from certified/18.9-cert5
|
||||||
|
to certified-18.9-cert5.
|
||||||
|
|
||||||
|
* Fixed issue where the asterisk version wasn't being considered
|
||||||
|
when looking for cached versions.
|
||||||
|
|
||||||
|
Resolves: #263
|
||||||
|
|
||||||
|
- ### main/refer.c: Fix double free in refer_data_destructor + potential leak
|
||||||
|
Author: Maximilian Fridrich
|
||||||
|
Date: 2023-08-21
|
||||||
|
|
||||||
|
Resolves: #267
|
||||||
|
|
||||||
|
- ### sig_analog: Add Called Subscriber Held capability.
|
||||||
|
Author: Naveen Albert
|
||||||
|
Date: 2023-08-09
|
||||||
|
|
||||||
|
This adds support for Called Subscriber Held for FXS
|
||||||
|
lines, which allows users to go on hook when receiving
|
||||||
|
a call and resume the call later from another phone on
|
||||||
|
the same line, without disconnecting the call. This is
|
||||||
|
a convenience mechanism that most real PSTN telephone
|
||||||
|
switches support.
|
||||||
|
|
||||||
|
ASTERISK-30372 #close
|
||||||
|
|
||||||
|
Resolves: #240
|
||||||
|
|
||||||
|
UserNote: 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.
|
||||||
|
|
||||||
|
|
||||||
|
- ### Revert "app_stack: Print proper exit location for PBXless channels."
|
||||||
|
Author: Matthew Fredrickson
|
||||||
|
Date: 2023-08-10
|
||||||
|
|
||||||
|
This reverts commit 617dad4cba1513dddce87b8e95a61415fb587cf1.
|
||||||
|
|
||||||
|
apps/app_stack.c: Revert buggy gosub patch
|
||||||
|
|
||||||
|
This seems to break the case when a predial macro calls a gosub.
|
||||||
|
When the gosub calls return, the Return function outputs:
|
||||||
|
|
||||||
|
app_stack.c:423 return_exec: Return without Gosub: stack is empty
|
||||||
|
|
||||||
|
This returns -1 to the calling macro, which returns to app_dial
|
||||||
|
and causes the call to hangup instead of proceeding with the macro
|
||||||
|
that invoked the gosub.
|
||||||
|
|
||||||
|
Resolves: #253
|
||||||
|
|
||||||
|
- ### install_prereq: Fix dependency install on aarch64.
|
||||||
|
Author: Jason D. McCormick
|
||||||
|
Date: 2023-04-28
|
||||||
|
|
||||||
|
Fixes dependency solutions in install_prereq for Debian aarch64
|
||||||
|
platforms. install_prereq was attempting to forcibly install 32-bit
|
||||||
|
armhf packages due to the aptitude search for dependencies.
|
||||||
|
|
||||||
|
Resolves: #37
|
||||||
|
|
||||||
|
- ### res_pjsip.c: Set contact_user on incoming call local Contact header
|
||||||
|
Author: MikeNaso
|
||||||
|
Date: 2023-08-08
|
||||||
|
|
||||||
|
If the contact_user is configured on the endpoint it will now be set on the local Contact header URI for incoming calls. The contact_user has already been set on the local Contact header URI for outgoing calls.
|
||||||
|
|
||||||
|
Resolves: #226
|
||||||
|
|
||||||
|
- ### extconfig: Allow explicit DB result set ordering to be disabled.
|
||||||
|
Author: Sean Bright
|
||||||
|
Date: 2023-07-12
|
||||||
|
|
||||||
|
Added a new boolean configuration flag -
|
||||||
|
`order_multi_row_results_by_initial_column` - to both res_pgsql.conf
|
||||||
|
and res_config_odbc.conf that allows the administrator to disable the
|
||||||
|
explicit `ORDER BY` that was previously being added to all generated
|
||||||
|
SQL statements that returned multiple rows.
|
||||||
|
|
||||||
|
Fixes: #179
|
||||||
|
|
||||||
|
- ### rest-api: Run make ari-stubs
|
||||||
|
Author: George Joseph
|
||||||
|
Date: 2023-08-09
|
||||||
|
|
||||||
|
An earlier cherry-pick that involved rest-api somehow didn't include
|
||||||
|
a comment change in res/ari/resource_endpoints.h. This commit
|
||||||
|
corrects that. No changes other than the comment.
|
||||||
|
|
||||||
|
|
||||||
|
- ### res_pjsip_header_funcs: Make prefix argument optional.
|
||||||
|
Author: Naveen Albert
|
||||||
|
Date: 2023-08-09
|
||||||
|
|
||||||
|
The documentation for PJSIP_HEADERS claims that
|
||||||
|
prefix is optional, but in the code it is actually not.
|
||||||
|
However, there is no inherent reason for this, as users
|
||||||
|
may want to retrieve all header names, not just those
|
||||||
|
beginning with a certain prefix.
|
||||||
|
|
||||||
|
This makes the prefix optional for this function,
|
||||||
|
simply fetching all header names if not specified.
|
||||||
|
As a result, the documentation is now correct.
|
||||||
|
|
||||||
|
Resolves: #230
|
||||||
|
|
||||||
|
UserNote: The prefix argument to PJSIP_HEADERS is now
|
||||||
|
optional. If not specified, all header names will be
|
||||||
|
returned.
|
||||||
|
|
||||||
|
|
||||||
|
- ### pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
|
||||||
|
Author: George Joseph
|
||||||
|
Date: 2023-08-11
|
||||||
|
|
||||||
|
The default is 32 with 8 being used by pjproject itself. Recent
|
||||||
|
commits have put us over the limit resulting in assertions in
|
||||||
|
pjproject. Since this value is used in invites, dialogs,
|
||||||
|
transports and subscriptions as well as the global pjproject
|
||||||
|
endpoint, we don't want to increase it too much.
|
||||||
|
|
||||||
|
Resolves: #255
|
||||||
|
|
||||||
|
- ### manager: Tolerate stasis messages with no channel snapshot.
|
||||||
|
Author: Joshua C. Colp
|
||||||
|
Date: 2023-08-09
|
||||||
|
|
||||||
|
In some cases I have yet to determine some stasis messages may
|
||||||
|
be created without a channel snapshot. This change adds some
|
||||||
|
tolerance to this scenario, preventing a crash from occurring.
|
||||||
|
|
||||||
|
|
||||||
|
- ### Remove unneeded CHANGES and UPGRADE files
|
||||||
|
Author: George Joseph
|
||||||
|
Date: 2023-08-09
|
||||||
|
|
||||||
|
|
||||||
@ -0,0 +1,41 @@
|
|||||||
|
CREATE TABLE alembic_version (
|
||||||
|
version_num VARCHAR(32) NOT NULL,
|
||||||
|
CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
|
||||||
|
);
|
||||||
|
|
||||||
|
-- Running upgrade -> 210693f3123d
|
||||||
|
|
||||||
|
CREATE TABLE cdr (
|
||||||
|
accountcode VARCHAR(20),
|
||||||
|
src VARCHAR(80),
|
||||||
|
dst VARCHAR(80),
|
||||||
|
dcontext VARCHAR(80),
|
||||||
|
clid VARCHAR(80),
|
||||||
|
channel VARCHAR(80),
|
||||||
|
dstchannel VARCHAR(80),
|
||||||
|
lastapp VARCHAR(80),
|
||||||
|
lastdata VARCHAR(80),
|
||||||
|
start DATETIME,
|
||||||
|
answer DATETIME,
|
||||||
|
end DATETIME,
|
||||||
|
duration INTEGER,
|
||||||
|
billsec INTEGER,
|
||||||
|
disposition VARCHAR(45),
|
||||||
|
amaflags VARCHAR(45),
|
||||||
|
userfield VARCHAR(256),
|
||||||
|
uniqueid VARCHAR(150),
|
||||||
|
linkedid VARCHAR(150),
|
||||||
|
peeraccount VARCHAR(20),
|
||||||
|
sequence INTEGER
|
||||||
|
);
|
||||||
|
|
||||||
|
INSERT INTO alembic_version (version_num) VALUES ('210693f3123d');
|
||||||
|
|
||||||
|
-- Running upgrade 210693f3123d -> 54cde9847798
|
||||||
|
|
||||||
|
ALTER TABLE cdr MODIFY accountcode VARCHAR(80) NULL;
|
||||||
|
|
||||||
|
ALTER TABLE cdr MODIFY peeraccount VARCHAR(80) NULL;
|
||||||
|
|
||||||
|
UPDATE alembic_version SET version_num='54cde9847798' WHERE alembic_version.version_num = '210693f3123d';
|
||||||
|
|
||||||
File diff suppressed because it is too large
Load Diff
@ -0,0 +1,29 @@
|
|||||||
|
BEGIN;
|
||||||
|
|
||||||
|
CREATE TABLE alembic_version (
|
||||||
|
version_num VARCHAR(32) NOT NULL,
|
||||||
|
CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
|
||||||
|
);
|
||||||
|
|
||||||
|
-- Running upgrade -> 4105ee839f58
|
||||||
|
|
||||||
|
CREATE TABLE queue_log (
|
||||||
|
id BIGSERIAL NOT NULL,
|
||||||
|
time TIMESTAMP WITHOUT TIME ZONE,
|
||||||
|
callid VARCHAR(80),
|
||||||
|
queuename VARCHAR(256),
|
||||||
|
agent VARCHAR(80),
|
||||||
|
event VARCHAR(32),
|
||||||
|
data1 VARCHAR(100),
|
||||||
|
data2 VARCHAR(100),
|
||||||
|
data3 VARCHAR(100),
|
||||||
|
data4 VARCHAR(100),
|
||||||
|
data5 VARCHAR(100),
|
||||||
|
PRIMARY KEY (id),
|
||||||
|
UNIQUE (id)
|
||||||
|
);
|
||||||
|
|
||||||
|
INSERT INTO alembic_version (version_num) VALUES ('4105ee839f58');
|
||||||
|
|
||||||
|
COMMIT;
|
||||||
|
|
||||||
@ -0,0 +1,35 @@
|
|||||||
|
CREATE TABLE alembic_version (
|
||||||
|
version_num VARCHAR(32) NOT NULL,
|
||||||
|
CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
|
||||||
|
);
|
||||||
|
|
||||||
|
-- Running upgrade -> a2e9769475e
|
||||||
|
|
||||||
|
CREATE TABLE voicemail_messages (
|
||||||
|
dir VARCHAR(255) NOT NULL,
|
||||||
|
msgnum INTEGER NOT NULL,
|
||||||
|
context VARCHAR(80),
|
||||||
|
macrocontext VARCHAR(80),
|
||||||
|
callerid VARCHAR(80),
|
||||||
|
origtime INTEGER,
|
||||||
|
duration INTEGER,
|
||||||
|
recording BLOB,
|
||||||
|
flag VARCHAR(30),
|
||||||
|
category VARCHAR(30),
|
||||||
|
mailboxuser VARCHAR(30),
|
||||||
|
mailboxcontext VARCHAR(30),
|
||||||
|
msg_id VARCHAR(40)
|
||||||
|
);
|
||||||
|
|
||||||
|
ALTER TABLE voicemail_messages ADD CONSTRAINT voicemail_messages_dir_msgnum PRIMARY KEY (dir, msgnum);
|
||||||
|
|
||||||
|
CREATE INDEX voicemail_messages_dir ON voicemail_messages (dir);
|
||||||
|
|
||||||
|
INSERT INTO alembic_version (version_num) VALUES ('a2e9769475e');
|
||||||
|
|
||||||
|
-- Running upgrade a2e9769475e -> 39428242f7f5
|
||||||
|
|
||||||
|
ALTER TABLE voicemail_messages MODIFY recording BLOB(4294967295) NULL;
|
||||||
|
|
||||||
|
UPDATE alembic_version SET version_num='39428242f7f5' WHERE alembic_version.version_num = 'a2e9769475e';
|
||||||
|
|
||||||
@ -0,0 +1,45 @@
|
|||||||
|
BEGIN;
|
||||||
|
|
||||||
|
CREATE TABLE alembic_version (
|
||||||
|
version_num VARCHAR(32) NOT NULL,
|
||||||
|
CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
|
||||||
|
);
|
||||||
|
|
||||||
|
-- Running upgrade -> 210693f3123d
|
||||||
|
|
||||||
|
CREATE TABLE cdr (
|
||||||
|
accountcode VARCHAR(20),
|
||||||
|
src VARCHAR(80),
|
||||||
|
dst VARCHAR(80),
|
||||||
|
dcontext VARCHAR(80),
|
||||||
|
clid VARCHAR(80),
|
||||||
|
channel VARCHAR(80),
|
||||||
|
dstchannel VARCHAR(80),
|
||||||
|
lastapp VARCHAR(80),
|
||||||
|
lastdata VARCHAR(80),
|
||||||
|
start TIMESTAMP WITHOUT TIME ZONE,
|
||||||
|
answer TIMESTAMP WITHOUT TIME ZONE,
|
||||||
|
"end" TIMESTAMP WITHOUT TIME ZONE,
|
||||||
|
duration INTEGER,
|
||||||
|
billsec INTEGER,
|
||||||
|
disposition VARCHAR(45),
|
||||||
|
amaflags VARCHAR(45),
|
||||||
|
userfield VARCHAR(256),
|
||||||
|
uniqueid VARCHAR(150),
|
||||||
|
linkedid VARCHAR(150),
|
||||||
|
peeraccount VARCHAR(20),
|
||||||
|
sequence INTEGER
|
||||||
|
);
|
||||||
|
|
||||||
|
INSERT INTO alembic_version (version_num) VALUES ('210693f3123d');
|
||||||
|
|
||||||
|
-- Running upgrade 210693f3123d -> 54cde9847798
|
||||||
|
|
||||||
|
ALTER TABLE cdr ALTER COLUMN accountcode TYPE VARCHAR(80);
|
||||||
|
|
||||||
|
ALTER TABLE cdr ALTER COLUMN peeraccount TYPE VARCHAR(80);
|
||||||
|
|
||||||
|
UPDATE alembic_version SET version_num='54cde9847798' WHERE alembic_version.version_num = '210693f3123d';
|
||||||
|
|
||||||
|
COMMIT;
|
||||||
|
|
||||||
File diff suppressed because it is too large
Load Diff
@ -0,0 +1,29 @@
|
|||||||
|
BEGIN;
|
||||||
|
|
||||||
|
CREATE TABLE alembic_version (
|
||||||
|
version_num VARCHAR(32) NOT NULL,
|
||||||
|
CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
|
||||||
|
);
|
||||||
|
|
||||||
|
-- Running upgrade -> 4105ee839f58
|
||||||
|
|
||||||
|
CREATE TABLE queue_log (
|
||||||
|
id BIGSERIAL NOT NULL,
|
||||||
|
time TIMESTAMP WITHOUT TIME ZONE,
|
||||||
|
callid VARCHAR(80),
|
||||||
|
queuename VARCHAR(256),
|
||||||
|
agent VARCHAR(80),
|
||||||
|
event VARCHAR(32),
|
||||||
|
data1 VARCHAR(100),
|
||||||
|
data2 VARCHAR(100),
|
||||||
|
data3 VARCHAR(100),
|
||||||
|
data4 VARCHAR(100),
|
||||||
|
data5 VARCHAR(100),
|
||||||
|
PRIMARY KEY (id),
|
||||||
|
UNIQUE (id)
|
||||||
|
);
|
||||||
|
|
||||||
|
INSERT INTO alembic_version (version_num) VALUES ('4105ee839f58');
|
||||||
|
|
||||||
|
COMMIT;
|
||||||
|
|
||||||
@ -0,0 +1,39 @@
|
|||||||
|
BEGIN;
|
||||||
|
|
||||||
|
CREATE TABLE alembic_version (
|
||||||
|
version_num VARCHAR(32) NOT NULL,
|
||||||
|
CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
|
||||||
|
);
|
||||||
|
|
||||||
|
-- Running upgrade -> a2e9769475e
|
||||||
|
|
||||||
|
CREATE TABLE voicemail_messages (
|
||||||
|
dir VARCHAR(255) NOT NULL,
|
||||||
|
msgnum INTEGER NOT NULL,
|
||||||
|
context VARCHAR(80),
|
||||||
|
macrocontext VARCHAR(80),
|
||||||
|
callerid VARCHAR(80),
|
||||||
|
origtime INTEGER,
|
||||||
|
duration INTEGER,
|
||||||
|
recording BYTEA,
|
||||||
|
flag VARCHAR(30),
|
||||||
|
category VARCHAR(30),
|
||||||
|
mailboxuser VARCHAR(30),
|
||||||
|
mailboxcontext VARCHAR(30),
|
||||||
|
msg_id VARCHAR(40)
|
||||||
|
);
|
||||||
|
|
||||||
|
ALTER TABLE voicemail_messages ADD CONSTRAINT voicemail_messages_dir_msgnum PRIMARY KEY (dir, msgnum);
|
||||||
|
|
||||||
|
CREATE INDEX voicemail_messages_dir ON voicemail_messages (dir);
|
||||||
|
|
||||||
|
INSERT INTO alembic_version (version_num) VALUES ('a2e9769475e');
|
||||||
|
|
||||||
|
-- Running upgrade a2e9769475e -> 39428242f7f5
|
||||||
|
|
||||||
|
ALTER TABLE voicemail_messages ALTER COLUMN recording TYPE BYTEA;
|
||||||
|
|
||||||
|
UPDATE alembic_version SET version_num='39428242f7f5' WHERE alembic_version.version_num = 'a2e9769475e';
|
||||||
|
|
||||||
|
COMMIT;
|
||||||
|
|
||||||
Loading…
Reference in new issue