mirror of https://github.com/asterisk/asterisk
Asterisk Development Team
2 years ago
parent
f8632232fd
commit
538802a5af
@ -1 +1 @@
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ChangeLogs/ChangeLog-18.19.0.md
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ChangeLogs/ChangeLog-18.20.0-rc1.md
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@ -0,0 +1,745 @@
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Change Log for Release asterisk-18.20.0-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-18.20.0-rc1.md)
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- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0-rc1)
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- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0-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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- 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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- app_macro: Fix locking around datastore access
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- Revert "app_stack: Print proper exit location for PBXless channels."
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- .github: Use generic releaser
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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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- core/ari/pjsip: Add refer mechanism
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- chan_dahdi: Allow autoreoriginating after hangup.
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- audiohook: Unlock channel in mute if no audiohooks present.
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- sig_analog: Allow three-way flash to time out to silence.
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- res_prometheus: Do not generate broken metrics
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- res_pjsip: Enable TLS v1.3 if present.
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- func_cut: Add example to documentation.
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- extensions.conf.sample: Remove reference to missing context.
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- func_export: Use correct function argument as variable name.
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- app_queue: Add support for applying caller priority change immediately.
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- .github: Fix cherry-pick reminder issues
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- chan_iax2.c: Avoid crash with IAX2 switch support.
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- res_geolocation: Ensure required 'location_info' is present.
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- 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.
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- app_voicemail: add CLI commands for message manipulation
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- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp->themssrc_valid` into the scope of the rtp_instance lock.
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- .github: Minor tweak to Asterisk Releaser
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- .github: Suppress cherry-pick reminder for some situations
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- sig_analog: Allow immediate fake ring to be suppressed.
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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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- ### core/ari/pjsip: Add refer mechanism
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There is a new ARI endpoint `/endpoints/refer` for referring
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an endpoint to some URI or endpoint.
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- ### chan_dahdi: Allow autoreoriginating after hangup.
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The autoreoriginate setting now allows for kewlstart FXS
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channels to automatically reoriginate and provide dial tone to the
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user again after all calls on the line have cleared. This saves users
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from having to manually hang up and pick up the receiver again before
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making another call.
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- ### sig_analog: Allow three-way flash to time out to silence.
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The threewaysilenthold option now allows the three-way
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dial tone to time out to silence, rather than continuing forever.
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- ### res_pjsip: Enable TLS v1.3 if present.
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res_pjsip now allows TLS v1.3 to be enabled if supported by
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the underlying PJSIP library. The bundled version of PJSIP supports
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TLS v1.3.
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- ### app_queue: Add support for applying caller priority change immediately.
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The 'queue priority caller' CLI command and
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'QueueChangePriorityCaller' AMI action now have an 'immediate'
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argument which allows the caller priority change to be reflected
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immediately, causing the position of a caller to move within the
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queue depending on the priorities of the other callers.
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- ### 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.
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The following manager actions have been added
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VoicemailBoxSummary - Generate message list for a given mailbox
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VoicemailRemove - Remove a message from a mailbox folder
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VoicemailMove - Move a message from one folder to another within a mailbox
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VoicemailForward - Copy a message from one folder in one mailbox
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to another folder in another or the same mailbox.
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- ### app_voicemail: add CLI commands for message manipulation
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The following CLI commands have been added to app_voicemail
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voicemail show mailbox <mailbox> <context>
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Show contents of mailbox <mailbox>@<context>
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voicemail remove <mailbox> <context> <from_folder> <messageid>
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Remove message <messageid> from <from_folder> in mailbox <mailbox>@<context>
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voicemail move <mailbox> <context> <from_folder> <messageid> <to_folder>
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Move message <messageid> in mailbox <mailbox>&<context> from <from_folder> to <to_folder>
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voicemail forward <from_mailbox> <from_context> <from_folder> <messageid> <to_mailbox> <to_context> <to_folder>
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Forward message <messageid> in mailbox <mailbox>@<context> <from_folder> to
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mailbox <mailbox>@<context> <to_folder>
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- ### sig_analog: Allow immediate fake ring to be suppressed.
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The immediatering option can now be set to no to suppress
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the fake audible ringback provided when immediate=yes on FXS channels.
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Upgrade Notes:
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----------------------------------------
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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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- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to some resource
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- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when immediate=yes
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- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
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- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
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- #181: [improvement]: app_voicemail - add manager actions to display and manipulate messages
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- #202: [improvement]: app_queue: Add support for immediately applying queue caller priority change
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- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
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- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
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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
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- #233: [bug]: Deadlock with MixMonitorMute AMI action
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- #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
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- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules registered"
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- #263: [bug]: download_externals doesn't always handle versions correctly
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- #265: [bug]: app_macro isn't locking around channel datastore access
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- #267: [bug]: ari: refer with display_name key in request body leads to crash
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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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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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- ### George Joseph (12):
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- .github: Suppress cherry-pick reminder for some situations
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- .github: Minor tweak to Asterisk Releaser
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- .github: Fix cherry-pick reminder issues
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- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
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- rest-api: Run make ari-stubs
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- .github: Use generic releaser
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||||
- download_externals: Fix a few version related issues
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- alembic: Fix quoting of the 100rel column
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- .github: Update workflow-application-token-action to v2
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||||
- ari-stubs: Fix broken documentation anchors
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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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- ### Holger Hans Peter Freyther (1):
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- res_prometheus: Do not generate broken metrics
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||||
|
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- ### Jason D. McCormick (1):
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||||
- install_prereq: Fix dependency install on aarch64.
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||||
|
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- ### Joshua C. Colp (3):
|
||||
- app_queue: Add support for applying caller priority change immediately.
|
||||
- audiohook: Unlock channel in mute if no audiohooks present.
|
||||
- manager: Tolerate stasis messages with no channel snapshot.
|
||||
|
||||
- ### Matthew Fredrickson (2):
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- Revert "app_stack: Print proper exit location for PBXless channels."
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- app_macro: Fix locking around datastore access
|
||||
|
||||
- ### Maximilian Fridrich (2):
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- core/ari/pjsip: Add refer mechanism
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- main/refer.c: Fix double free in refer_data_destructor + potential leak
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||||
|
||||
- ### Mike Bradeen (3):
|
||||
- app_voicemail: add CLI commands for message manipulation
|
||||
- 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: Fix for loop declarations
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|
||||
- ### MikeNaso (1):
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- res_pjsip.c: Set contact_user on incoming call local Contact header
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||||
|
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- ### Naveen Albert (7):
|
||||
- sig_analog: Allow immediate fake ring to be suppressed.
|
||||
- sig_analog: Allow three-way flash to time out to silence.
|
||||
- chan_dahdi: Allow autoreoriginating after hangup.
|
||||
- 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 (6):
|
||||
- res_geolocation: Ensure required 'location_info' is present.
|
||||
- chan_iax2.c: Avoid crash with IAX2 switch support.
|
||||
- func_export: Use correct function argument as variable name.
|
||||
- extensions.conf.sample: Remove reference to missing context.
|
||||
- res_pjsip: Enable TLS v1.3 if present.
|
||||
- extconfig: Allow explicit DB result set ordering to be disabled.
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||||
|
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- ### phoneben (1):
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- func_cut: Add example to documentation.
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- ### zhengsh (2):
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||||
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp->themssrc_valid` into the scope of the rtp_instance lock.
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||||
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
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|
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Detail:
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----------------------------------------
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- ### ari-stubs: Fix more local anchor references
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Author: George Joseph
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Date: 2023-09-05
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Also allow CreateDocs job to be run manually with default branches.
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- ### ari-stubs: Fix more local anchor references
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Author: George Joseph
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Date: 2023-09-05
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||||
|
||||
Also allow CreateDocs job to be run manually with default branches.
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- ### ari-stubs: Fix broken documentation anchors
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Author: George Joseph
|
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Date: 2023-09-05
|
||||
|
||||
All of the links that reference page anchors with capital letters in
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the ids (#Something) have been changed to lower case to match the
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anchors that are generated by mkdocs.
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||||
|
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- ### res_pjsip_session: Send Session Interval too small response
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||||
Author: Bastian Triller
|
||||
Date: 2023-08-28
|
||||
|
||||
Handle session interval lower than endpoint's configured minimum timer
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||||
when sending first answer. Timer setting is checked during this step and
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||||
needs to handled appropriately.
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||||
Before this change, no response was sent at all. After this change a
|
||||
response with 422 Session Interval too small is sent to UAC.
|
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|
||||
|
||||
- ### .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
|
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the caller manually hangs up the channel. We now
|
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abort if digit sending fails.
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|
||||
ASTERISK-29428 #close
|
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|
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Resolves: #281
|
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|
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- ### 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
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||||
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.
|
||||
|
||||
|
||||
- ### app_macro: Fix locking around datastore access
|
||||
Author: Matthew Fredrickson
|
||||
Date: 2023-08-21
|
||||
|
||||
app_macro sometimes would crash due to datastore list corruption on the
|
||||
channel because of lack of locking around find and create process for
|
||||
the macro datastore. This patch locks the channel lock prior to protect
|
||||
against this problem.
|
||||
|
||||
Resolves: #265
|
||||
|
||||
- ### 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
|
||||
|
||||
- ### .github: Use generic releaser
|
||||
Author: George Joseph
|
||||
Date: 2023-08-15
|
||||
|
||||
|
||||
- ### 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.
|
||||
|
||||
|
||||
- ### core/ari/pjsip: Add refer mechanism
|
||||
Author: Maximilian Fridrich
|
||||
Date: 2023-05-10
|
||||
|
||||
This change adds support for refers that are not session based. It
|
||||
includes a refer implementation for the PJSIP technology which results
|
||||
in out-of-dialog REFERs being sent to a PJSIP endpoint. These can be
|
||||
triggered using the new ARI endpoint `/endpoints/refer`.
|
||||
|
||||
Resolves: #71
|
||||
|
||||
UserNote: There is a new ARI endpoint `/endpoints/refer` for referring
|
||||
an endpoint to some URI or endpoint.
|
||||
|
||||
|
||||
- ### chan_dahdi: Allow autoreoriginating after hangup.
|
||||
Author: Naveen Albert
|
||||
Date: 2023-08-04
|
||||
|
||||
Currently, if an FXS channel is still off hook when
|
||||
all calls on the line have hung up, the user is provided
|
||||
reorder tone until going back on hook again.
|
||||
|
||||
In addition to not reflecting what most commercial switches
|
||||
actually do, it's very common for switches to automatically
|
||||
reoriginate for the user so that dial tone is provided without
|
||||
the user having to depress and release the hookswitch manually.
|
||||
This can increase convenience for users.
|
||||
|
||||
This behavior is now supported for kewlstart FXS channels.
|
||||
It's supported only for kewlstart (FXOKS) mainly because the
|
||||
behavior doesn't make any sense for ground start channels,
|
||||
and loop start signalling doesn't provide the necessary DAHDI
|
||||
event that makes this easy to implement. Likely almost everyone
|
||||
is using FXOKS over FXOLS anyways since FXOLS is pretty useless
|
||||
these days.
|
||||
|
||||
ASTERISK-30357 #close
|
||||
|
||||
Resolves: #224
|
||||
|
||||
UserNote: 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.
|
||||
|
||||
|
||||
- ### audiohook: Unlock channel in mute if no audiohooks present.
|
||||
Author: Joshua C. Colp
|
||||
Date: 2023-08-09
|
||||
|
||||
In the case where mute was called on a channel that had no
|
||||
audiohooks the code was not unlocking the channel, resulting
|
||||
in a deadlock.
|
||||
|
||||
Resolves: #233
|
||||
|
||||
- ### sig_analog: Allow three-way flash to time out to silence.
|
||||
Author: Naveen Albert
|
||||
Date: 2023-07-10
|
||||
|
||||
sig_analog allows users to flash and use the three-way dial
|
||||
tone as a primitive hold function, simply by never timing
|
||||
it out.
|
||||
|
||||
Some systems allow this dial tone to time out to silence,
|
||||
so the user is not annoyed by a persistent dial tone.
|
||||
This option allows the dial tone to time out normally to
|
||||
silence.
|
||||
|
||||
ASTERISK-30004 #close
|
||||
Resolves: #205
|
||||
|
||||
UserNote: The threewaysilenthold option now allows the three-way
|
||||
dial tone to time out to silence, rather than continuing forever.
|
||||
|
||||
|
||||
- ### res_prometheus: Do not generate broken metrics
|
||||
Author: Holger Hans Peter Freyther
|
||||
Date: 2023-04-07
|
||||
|
||||
In 8d6fdf9c3adede201f0ef026dab201b3a37b26b6 invisible bridges were
|
||||
skipped but that lead to producing metrics with no name and no help.
|
||||
|
||||
Keep track of the number of metrics configured and then only emit these.
|
||||
Add a basic testcase that verifies that there is no '(NULL)' in the
|
||||
output.
|
||||
|
||||
ASTERISK-30474
|
||||
|
||||
|
||||
- ### res_pjsip: Enable TLS v1.3 if present.
|
||||
Author: Sean Bright
|
||||
Date: 2023-08-02
|
||||
|
||||
Fixes #221
|
||||
|
||||
UserNote: 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.
|
||||
|
||||
|
||||
- ### func_cut: Add example to documentation.
|
||||
Author: phoneben
|
||||
Date: 2023-07-19
|
||||
|
||||
This adds an example to the XML documentation clarifying usage
|
||||
of the CUT function to address a common misusage.
|
||||
|
||||
|
||||
- ### extensions.conf.sample: Remove reference to missing context.
|
||||
Author: Sean Bright
|
||||
Date: 2023-07-16
|
||||
|
||||
c3ff4648 removed the [iaxtel700] context but neglected to remove
|
||||
references to it.
|
||||
|
||||
This commit addresses that and also removes iaxtel and freeworlddialup
|
||||
references from other config files.
|
||||
|
||||
|
||||
- ### func_export: Use correct function argument as variable name.
|
||||
Author: Sean Bright
|
||||
Date: 2023-07-12
|
||||
|
||||
Fixes #208
|
||||
|
||||
|
||||
- ### app_queue: Add support for applying caller priority change immediately.
|
||||
Author: Joshua C. Colp
|
||||
Date: 2023-07-07
|
||||
|
||||
The app_queue module provides both an AMI action and a CLI command
|
||||
to change the priority of a caller in a queue. Up to now this change
|
||||
of priority has only been reflected to new callers into the queue.
|
||||
|
||||
This change adds an "immediate" option to both the AMI action and
|
||||
CLI command which immediately applies the priority change respective
|
||||
to the other callers already in the queue. This can allow, for example,
|
||||
a caller to be placed at the head of the queue immediately if their
|
||||
priority is sufficient.
|
||||
|
||||
Resolves: #202
|
||||
|
||||
UserNote: The 'queue priority caller' CLI command and
|
||||
'QueueChangePriorityCaller' AMI action now have an 'immediate'
|
||||
argument which allows the caller priority change to be reflected
|
||||
immediately, causing the position of a caller to move within the
|
||||
queue depending on the priorities of the other callers.
|
||||
|
||||
|
||||
- ### .github: Fix cherry-pick reminder issues
|
||||
Author: George Joseph
|
||||
Date: 2023-07-17
|
||||
|
||||
|
||||
- ### chan_iax2.c: Avoid crash with IAX2 switch support.
|
||||
Author: Sean Bright
|
||||
Date: 2023-07-07
|
||||
|
||||
A change made in 82cebaa0 did not properly handle the case when a
|
||||
channel was not provided, triggering a crash. ast_check_hangup(...)
|
||||
does not protect against NULL pointers.
|
||||
|
||||
Fixes #180
|
||||
|
||||
|
||||
- ### res_geolocation: Ensure required 'location_info' is present.
|
||||
Author: Sean Bright
|
||||
Date: 2023-07-07
|
||||
|
||||
Fixes #189
|
||||
|
||||
|
||||
- ### 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.
|
||||
Author: Mike Bradeen
|
||||
Date: 2023-06-29
|
||||
|
||||
Resolves: #181
|
||||
|
||||
UserNote: The following manager actions have been added
|
||||
|
||||
VoicemailBoxSummary - Generate message list for a given mailbox
|
||||
|
||||
VoicemailRemove - Remove a message from a mailbox folder
|
||||
|
||||
VoicemailMove - Move a message from one folder to another within a mailbox
|
||||
|
||||
VoicemailForward - Copy a message from one folder in one mailbox
|
||||
to another folder in another or the same mailbox.
|
||||
|
||||
|
||||
- ### app_voicemail: add CLI commands for message manipulation
|
||||
Author: Mike Bradeen
|
||||
Date: 2023-06-20
|
||||
|
||||
Adds CLI commands to allow move/remove/forward individual messages
|
||||
from a particular mailbox folder. The forward command can be used
|
||||
to copy a message within a mailbox or to another mailbox. Also adds
|
||||
a show mailbox, required to retrieve message ID's.
|
||||
|
||||
Resolves: #170
|
||||
|
||||
UserNote: The following CLI commands have been added to app_voicemail
|
||||
|
||||
voicemail show mailbox <mailbox> <context>
|
||||
Show contents of mailbox <mailbox>@<context>
|
||||
|
||||
voicemail remove <mailbox> <context> <from_folder> <messageid>
|
||||
Remove message <messageid> from <from_folder> in mailbox <mailbox>@<context>
|
||||
|
||||
voicemail move <mailbox> <context> <from_folder> <messageid> <to_folder>
|
||||
Move message <messageid> in mailbox <mailbox>&<context> from <from_folder> to <to_folder>
|
||||
|
||||
voicemail forward <from_mailbox> <from_context> <from_folder> <messageid> <to_mailbox> <to_context> <to_folder>
|
||||
Forward message <messageid> in mailbox <mailbox>@<context> <from_folder> to
|
||||
mailbox <mailbox>@<context> <to_folder>
|
||||
|
||||
|
||||
- ### res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp->themssrc_valid` into the scope of the rtp_instance lock.
|
||||
Author: zhengsh
|
||||
Date: 2023-06-30
|
||||
|
||||
From the gdb information, it was found that when calling __ast_free, the size of the
|
||||
allocated space pointed to by the pointer matches the size created when rtp->themssrc_valid
|
||||
is equal to 0. However, in reality, when reading the value of rtp->themssrc_valid in gdb,
|
||||
it is found to be 1.
|
||||
|
||||
Within ast_rtcp_write(), the call to ast_rtp_rtcp_report_alloc() uses rtp->themssrc_valid,
|
||||
which is outside the protection of the rtp_instance lock. However,
|
||||
ast_rtcp_generate_report(), which is called by ast_rtcp_generate_compound_prefix(), uses
|
||||
rtp->themssrc_valid within the protection of the rtp_instance lock.
|
||||
|
||||
This can lead to the possibility that the value of rtp->themssrc_valid used in the call to
|
||||
ast_rtp_rtcp_report_alloc() may be different from the value of rtp->themssrc_valid used
|
||||
within ast_rtcp_generate_report().
|
||||
|
||||
Resolves: asterisk#63
|
||||
|
||||
- ### .github: Minor tweak to Asterisk Releaser
|
||||
Author: George Joseph
|
||||
Date: 2023-07-12
|
||||
|
||||
|
||||
- ### .github: Suppress cherry-pick reminder for some situations
|
||||
Author: George Joseph
|
||||
Date: 2023-07-11
|
||||
|
||||
In PROpenedOrUpdated, the cherry-pick reminder will now be
|
||||
suppressed if there are already valid 'cherry-pick-to' comments
|
||||
in the PR or the PR contained a 'cherry-pick-to: none' comment.
|
||||
|
||||
|
||||
- ### sig_analog: Allow immediate fake ring to be suppressed.
|
||||
Author: Naveen Albert
|
||||
Date: 2023-06-08
|
||||
|
||||
When immediate=yes on an FXS channel, sig_analog will
|
||||
start fake audible ringback that continues until the
|
||||
channel is answered. Even if it answers immediately,
|
||||
the ringback is still audible for a brief moment.
|
||||
This can be disruptive and unwanted behavior.
|
||||
|
||||
This adds an option to disable this behavior, though
|
||||
the default behavior remains unchanged.
|
||||
|
||||
ASTERISK-30003 #close
|
||||
Resolves: #118
|
||||
|
||||
UserNote: The immediatering option can now be set to no to suppress
|
||||
the fake audible ringback provided when immediate=yes on FXS channels.
|
||||
|
||||
|
||||
Loading…
Reference in new issue