kenmirrors/asterisk
1
0
Fork 0
mirror of https://github.com/asterisk/asterisk.git synced 2025-09-02 11:06:31 +00:00
Code Issues Packages Projects Releases Wiki Activity
Files
62547b25e15fdf44f2614076b0a1267e0a2c4dfc
asterisk/ChangeLogs/ChangeLog-21.0.0.md
Asterisk Development Team 12da95e53f Update for 21.0.0
2023-10-18 17:14:20 +00:00

25 KiB
Raw Blame History

Change Log for Release asterisk-21.0.0

Links:

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 removes the 'w' and 'W' options for app_queue. MixMonitor should be default and only option for all settings that previously used either Monitor or MixMonitor.

  • 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.

  • app_cdr: Remove deprecated application and option.

    The previously deprecated NoCDR application has been removed. Additionally, the previously deprecated 'e' option to the ResetCDR application has been removed.

  • app_macro: Remove deprecated module.

    This module was deprecated in Asterisk 16 and is now being removed in accordance with the Asterisk Module Deprecation policy. 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. The following modules have additional impacts: app_dial - no longer supports M^ connected/redirecting macro app_minivm - samples written using macro will no longer work. The sample needs to be re-written app_queue - can no longer call 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 options pbx - removed macrolock pbx_dundi - no longer look for macro snmp - removed macro context, exten, and priority

  • translate.c: Prefer better codecs upon translate ties.

    When setting up translation between two codecs the quality was not taken into account, resulting in suboptimal translation. The quality is now taken into account, which can reduce the number of translation steps required, and improve the resulting quality.

  • chan_sip: Remove deprecated module.

    This module was deprecated in Asterisk 17 and is now being removed in accordance with the Asterisk Module Deprecation policy.

  • chan_alsa: Remove deprecated module.

    This module was deprecated in Asterisk 19 and is now being removed in accordance with the Asterisk Module Deprecation policy.

  • pbx_builtins: Remove deprecated and defunct functionality.

    The previously deprecated ImportVar and SetAMAFlags applications have now been removed.

  • chan_mgcp: Remove deprecated module.

    This module was deprecated in Asterisk 19 and is now being removed in accordance with the Asterisk Module Deprecation policy.

  • chan_skinny: Remove deprecated module.

    This module was deprecated in Asterisk 19 and is now being removed in accordance with the Asterisk Module Deprecation policy.

Closed Issues:

  • #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on aarch64 Debian platforms
  • #39: [Bug]: Remove .gitreview from repository.
  • #41: [Bug]: say.c Time announcement does not say o'clock for the French language
  • #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
  • #78: [improvement]: Deprecate ast_gethostbyname()
  • #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
  • #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
  • #183: [deprecation]: Deprecate users.conf
  • #226: [improvement]: Apply contact_user to incoming calls
  • #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
  • #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
  • #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
  • #274: [bug]: Syntax Error in SQL Code
  • #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99 -Wdeclaration-after-statement'
  • #277: [bug]: pbx.c: Compiler error with gcc 12.2
  • #281: [bug]: app_dial: Infinite loop if called channel hangs up while receiving digits
  • #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in build_resource_tree

Commits By Author:

  • Asterisk Development Team (1):

    • Update for 21.0.0-rc1

  • Bastian Triller (1):

    • res_pjsip_session: Send Session Interval too small response

  • George Joseph (9):

    • Remove unneeded CHANGES and UPGRADE files
    • 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
    • .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.

  • 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

  • res_pjsip_pubsub: Add body_type to test_handler for unit tests

    Author: George Joseph
    Date: 2023-09-15

    The ast_sip_subscription_handler "test_handler" used for the unit tests didn't set "body_type" so the NULL value was causing a SEGV in build_subscription_tree(). It's now set to "".

    Resolves: #335

  • 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 617dad4cba.

    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