mirror of https://github.com/asterisk/asterisk
Asterisk Development Team
9 months ago
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ChangeLogs/ChangeLog-22.0.0-rc1.md
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## Change Log for Release asterisk-22.0.0-rc1
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### Links:
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- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0-rc1.md)
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- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0-rc1)
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- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0-rc1.tar.gz)
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- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
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### Summary:
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- Commits: 19
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- Commit Authors: 8
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- Issues Resolved: 10
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- Security Advisories Resolved: 1
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- [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9): A malformed Contact or Record-Route URI in an incoming SIP request can cause Asterisk to crash when res_resolver_unbound is used
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### User Notes:
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- #### feat: ARI "ChannelToneDetected" event
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Setting the TONE_DETECT dialplan function on a channel
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in ARI will now cause a ChannelToneDetected ARI event to be raised
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when the specified tone is detected.
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### Upgrade Notes:
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- #### app_record: Add RECORD_TIME output variable.
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The RECORD_TIME variable now contains
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the duration of Record() recordings in milliseconds.
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### Commit Authors:
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- Alexei Gradinari: (2)
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- George Joseph: (7)
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- Gibbz00: (1)
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- Joshua C. Colp: (2)
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- Mike Bradeen: (2)
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- Naveen Albert: (1)
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- Sean Bright: (2)
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- Tinet-Mucw: (2)
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## Issue and Commit Detail:
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### Closed Issues:
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- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming SIP request can cause Asterisk to crash when res_resolver_unbound is used
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- 548: [improvement]: Get Record() audio duration/length
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- 763: [bug]: autoservice thread stuck in an endless sleep
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- 811: [new-feature]: ARI channel tone detect events.
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- 845: [bug]: Buffer overflow in handling of security mechanisms in res_pjsip
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- 847: [bug]: Asterisk not using negotiated fall-back 8K digits
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- 854: [bug]: wrong properties in stir_shaken.conf.sample
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- 856: [bug]: res_pjsip_sdp_rtp leaks astobj2 ast_format
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- 861: [bug]: ChanSpy unable to read audiohook read direction frame when no packet lost on both side of the call
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- 876: [bug]: ChanSpy unable to write whisper_audiohook when set flag OPTION_READONLY
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- 879: [bug]: res_stir_shaken/verification.c: Getting verification errors when global_disable=yes
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### Commits By Author:
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- #### Alexei Gradinari (2):
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- res_pjsip_sdp_rtp fix leaking astobj2 ast_format
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- autoservice: Do not sleep if autoservice_stop is called within autoservice thr..
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- #### George Joseph (5):
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- stir_shaken.conf.sample: Fix bad references to private_key_path
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- security_agreements.c: Refactor the to_str functions and fix a few other bugs
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- app_voicemail: Use ast_asprintf to create mailbox SQL query
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- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
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- res_stir_shaken: Check for disabled before param validation
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- #### Mike Bradeen (1):
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- res_pjsip_sdp_rtp: Use negotiated DTMF Payload types on bitrate mismatch
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- #### Sean Bright (2):
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- alembic: Make 'revises' header comment match reality.
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- res_pjsip_logger.c: Fix 'OPTIONS' tab completion.
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- #### Tinet-mucw (2):
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- app_chanspy.c: resolving the issue with audiohook direction read
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- app_chanspy.c: resolving the issue writing frame to whisper audiohook.
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### Commit List:
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- Prepare master for Asterisk 22
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- Update issue guidelines link for bug reports.
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- .github: NightlyAdmin now calls external CloseStaleIssuesAndPRs
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- app_record: Add RECORD_TIME output variable.
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- Revert "app_record: Add RECORD_TIME output variable."
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- feat: ARI "ChannelToneDetected" event
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- Update version for Asterisk 22
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- res_stir_shaken: Check for disabled before param validation
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- app_chanspy.c: resolving the issue writing frame to whisper audiohook.
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- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
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- app_voicemail: Use ast_asprintf to create mailbox SQL query
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- res_pjsip_sdp_rtp: Use negotiated DTMF Payload types on bitrate mismatch
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- app_chanspy.c: resolving the issue with audiohook direction read
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- security_agreements.c: Refactor the to_str functions and fix a few other bugs
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- res_pjsip_sdp_rtp fix leaking astobj2 ast_format
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- stir_shaken.conf.sample: Fix bad references to private_key_path
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- res_pjsip_logger.c: Fix 'OPTIONS' tab completion.
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- alembic: Make 'revises' header comment match reality.
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### Commit Details:
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#### Prepare master for Asterisk 22
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Author: George Joseph
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Date: 2023-08-09
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#### Update issue guidelines link for bug reports.
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Author: Joshua C. Colp
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Date: 2023-10-27
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#### .github: NightlyAdmin now calls external CloseStaleIssuesAndPRs
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Author: George Joseph
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Date: 2024-03-20
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#### app_record: Add RECORD_TIME output variable.
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Author: Naveen Albert
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Date: 2024-01-22
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This adds the RECORD_TIME variable to Record(),
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which is set to the recording duration before
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the application returns.
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Resolves: #548
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UpgradeNote: The RECORD_TIME variable now contains
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the duration of Record() recordings in milliseconds.
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#### Revert "app_record: Add RECORD_TIME output variable."
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Author: Joshua C. Colp
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Date: 2024-04-30
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This reverts commit 6e8dccdbbf896bcc99046ae249db360698ede0b2.
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#### feat: ARI "ChannelToneDetected" event
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Author: gibbz00
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Date: 2024-07-18
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A stasis event is now produced when using the TONE_DETECT dialplan
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function. This event is published over ARI using the ChannelToneDetected
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event. This change does not make it available over AMI.
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Fixes: #811
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UserNote: Setting the TONE_DETECT dialplan function on a channel
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in ARI will now cause a ChannelToneDetected ARI event to be raised
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when the specified tone is detected.
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#### Update version for Asterisk 22
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Author: Mike Bradeen
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Date: 2024-08-14
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#### res_stir_shaken: Check for disabled before param validation
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Author: George Joseph
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Date: 2024-09-11
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For both attestation and verification, we now check whether they've
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been disabled either globally or by the profile before validating
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things like callerid, orig_tn, dest_tn, etc. This prevents useless
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error messages.
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Resolves: #879
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#### app_chanspy.c: resolving the issue writing frame to whisper audiohook.
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Author: Tinet-mucw
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Date: 2024-09-10
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ChanSpy(${channel}, qEoSw): because flags set o, ast_audiohook_set_frame_feed_direction(audiohook, AST_AUDIOHOOK_DIRECTION_READ); this will effect whisper audiohook and spy audiohook, this makes writing frame to whisper audiohook impossible. So add function start_whispering to starting whisper audiohook.
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Resolves: #876
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#### autoservice: Do not sleep if autoservice_stop is called within autoservice thr..
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Author: Alexei Gradinari
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Date: 2024-09-04
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It's possible that ast_autoservice_stop is called within the autoservice thread.
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In this case the autoservice thread is stuck in an endless sleep.
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To avoid endless sleep ast_autoservice_stop must check that it's not called
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within the autoservice thread.
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Fixes: #763
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#### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
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Author: George Joseph
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Date: 2024-08-12
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The ub_result pointer passed to unbound_resolver_callback by
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libunbound can be NULL if the query was for something malformed
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like `.1` or `[.1]`. If it is, we now set a 'ns_r_formerr' result
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and return instead of crashing with a SEGV. This causes pjproject
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to simply cancel the transaction with a "No answer record in the DNS
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response" error. The existing "off nominal" unit test was also
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updated to check this condition.
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Although not necessary for this fix, we also made
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ast_dns_resolver_completed() tolerant of a NULL result.
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Resolves: GHSA-v428-g3cw-7hv9
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#### app_voicemail: Use ast_asprintf to create mailbox SQL query
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Author: George Joseph
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Date: 2024-09-03
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...instead of trying to calculate the length of the buffer needed
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manually.
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#### res_pjsip_sdp_rtp: Use negotiated DTMF Payload types on bitrate mismatch
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Author: Mike Bradeen
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Date: 2024-08-21
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When Asterisk sends an offer to Bob that includes 48K and 8K codecs with
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matching 4733 offers, Bob may want to use the 48K audio codec but can not
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accept 48K digits and so negotiates for a mixed set.
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Asterisk will now check Bob's offer to make sure Bob has indicated this is
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acceptible and if not, will use Bob's preference.
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Fixes: #847
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#### app_chanspy.c: resolving the issue with audiohook direction read
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Author: Tinet-mucw
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Date: 2024-08-30
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ChanSpy(${channel}, qEoS): When chanspy spy the direction read, reading frame is often failed when reading direction read audiohook. because chanspy only read audiohook direction read; write_factory_ms will greater than 100ms soon, then ast_slinfactory_flush will being called, then direction read will fail.
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Resolves: #861
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#### security_agreements.c: Refactor the to_str functions and fix a few other bugs
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Author: George Joseph
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Date: 2024-08-17
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* A static array of security mechanism type names was created.
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* ast_sip_str_to_security_mechanism_type() was refactored to do
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a lookup in the new array instead of using fixed "if/else if"
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statments.
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* security_mechanism_to_str() and ast_sip_security_mechanisms_to_str()
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were refactored to use ast_str instead of a fixed length buffer
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to store the result.
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* ast_sip_security_mechanism_type_to_str was removed in favor of
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just referencing the new type name array. Despite starting with
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"ast_sip_", it was a static function so removing it doesn't affect
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ABI.
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* Speaking of "ast_sip_", several other static functions that
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started with "ast_sip_" were renamed to avoid confusion about
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their public availability.
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* A few VECTOR free loops were replaced with AST_VECTOR_RESET().
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* Fixed a meomry leak in pjsip_configuration.c endpoint_destructor
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caused by not calling ast_sip_security_mechanisms_vector_destroy().
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* Fixed a memory leak in res_pjsip_outbound_registration.c
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add_security_headers() caused by not specifying OBJ_NODATA in
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an ao2_callback.
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* Fixed a few ao2_callback return code misuses.
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Resolves: #845
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#### res_pjsip_sdp_rtp fix leaking astobj2 ast_format
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Author: Alexei Gradinari
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Date: 2024-08-23
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PR #700 added a preferred_format for the struct ast_rtp_codecs,
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but when set the preferred_format it leaks an astobj2 ast_format.
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In the next code
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ast_rtp_codecs_set_preferred_format(&codecs, ast_format_cap_get_format(joint, 0));
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both functions ast_rtp_codecs_set_preferred_format
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and ast_format_cap_get_format increases the ao2 reference count.
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Fixes: #856
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#### stir_shaken.conf.sample: Fix bad references to private_key_path
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Author: George Joseph
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Date: 2024-08-22
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They should be private_key_file.
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Resolves: #854
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#### res_pjsip_logger.c: Fix 'OPTIONS' tab completion.
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Author: Sean Bright
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Date: 2024-08-19
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Fixes #843
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#### alembic: Make 'revises' header comment match reality.
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Author: Sean Bright
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Date: 2024-08-17
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@ -0,0 +1,41 @@
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CREATE TABLE alembic_version (
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version_num VARCHAR(32) NOT NULL,
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CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
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);
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-- Running upgrade -> 210693f3123d
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CREATE TABLE cdr (
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accountcode VARCHAR(20),
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src VARCHAR(80),
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dst VARCHAR(80),
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dcontext VARCHAR(80),
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clid VARCHAR(80),
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channel VARCHAR(80),
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dstchannel VARCHAR(80),
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lastapp VARCHAR(80),
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lastdata VARCHAR(80),
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start DATETIME,
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answer DATETIME,
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end DATETIME,
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duration INTEGER,
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billsec INTEGER,
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disposition VARCHAR(45),
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amaflags VARCHAR(45),
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userfield VARCHAR(256),
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uniqueid VARCHAR(150),
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linkedid VARCHAR(150),
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peeraccount VARCHAR(20),
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sequence INTEGER
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);
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INSERT INTO alembic_version (version_num) VALUES ('210693f3123d');
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-- Running upgrade 210693f3123d -> 54cde9847798
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ALTER TABLE cdr MODIFY accountcode VARCHAR(80) NULL;
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ALTER TABLE cdr MODIFY peeraccount VARCHAR(80) NULL;
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UPDATE alembic_version SET version_num='54cde9847798' WHERE alembic_version.version_num = '210693f3123d';
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File diff suppressed because it is too large
Load Diff
@ -0,0 +1,29 @@
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BEGIN;
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CREATE TABLE alembic_version (
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version_num VARCHAR(32) NOT NULL,
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CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
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);
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-- Running upgrade -> 4105ee839f58
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CREATE TABLE queue_log (
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id BIGSERIAL NOT NULL,
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time TIMESTAMP WITHOUT TIME ZONE,
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callid VARCHAR(80),
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queuename VARCHAR(256),
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agent VARCHAR(80),
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event VARCHAR(32),
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data1 VARCHAR(100),
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data2 VARCHAR(100),
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data3 VARCHAR(100),
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data4 VARCHAR(100),
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data5 VARCHAR(100),
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PRIMARY KEY (id),
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UNIQUE (id)
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);
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INSERT INTO alembic_version (version_num) VALUES ('4105ee839f58') RETURNING alembic_version.version_num;
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COMMIT;
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@ -0,0 +1,41 @@
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CREATE TABLE alembic_version (
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version_num VARCHAR(32) NOT NULL,
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CONSTRAINT alembic_version_pkc PRIMARY KEY (version_num)
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);
|
||||
|
||||
-- Running upgrade -> a2e9769475e
|
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CREATE TABLE voicemail_messages (
|
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dir VARCHAR(255) NOT NULL,
|
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msgnum INTEGER NOT NULL,
|
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context VARCHAR(80),
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macrocontext VARCHAR(80),
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callerid VARCHAR(80),
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origtime INTEGER,
|
||||
duration INTEGER,
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recording BLOB,
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flag VARCHAR(30),
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category VARCHAR(30),
|
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mailboxuser VARCHAR(30),
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mailboxcontext VARCHAR(30),
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msg_id VARCHAR(40)
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);
|
||||
|
||||
ALTER TABLE voicemail_messages ADD CONSTRAINT voicemail_messages_dir_msgnum PRIMARY KEY (dir, msgnum);
|
||||
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||||
CREATE INDEX voicemail_messages_dir ON voicemail_messages (dir);
|
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|
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INSERT INTO alembic_version (version_num) VALUES ('a2e9769475e');
|
||||
|
||||
-- Running upgrade a2e9769475e -> 39428242f7f5
|
||||
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||||
ALTER TABLE voicemail_messages MODIFY recording BLOB(4294967295) NULL;
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||||
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UPDATE alembic_version SET version_num='39428242f7f5' WHERE alembic_version.version_num = 'a2e9769475e';
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||||
-- Running upgrade 39428242f7f5 -> 1c55c341360f
|
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ALTER TABLE voicemail_messages DROP COLUMN macrocontext;
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UPDATE alembic_version SET version_num='1c55c341360f' WHERE alembic_version.version_num = '39428242f7f5';
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||||
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||||
@ -0,0 +1,45 @@
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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') RETURNING alembic_version.version_num;
|
||||
|
||||
-- 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') RETURNING alembic_version.version_num;
|
||||
|
||||
COMMIT;
|
||||
|
||||
@ -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 -> 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') RETURNING alembic_version.version_num;
|
||||
|
||||
-- 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';
|
||||
|
||||
-- Running upgrade 39428242f7f5 -> 1c55c341360f
|
||||
|
||||
ALTER TABLE voicemail_messages DROP COLUMN macrocontext;
|
||||
|
||||
UPDATE alembic_version SET version_num='1c55c341360f' WHERE alembic_version.version_num = '39428242f7f5';
|
||||
|
||||
COMMIT;
|
||||
|
||||
Loading…
Reference in new issue