mirror of https://github.com/asterisk/asterisk
Asterisk Development Team
7 months ago
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badaf0e383
commit
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ChangeLogs/ChangeLog-22.0.0.md
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ChangeLogs/ChangeLog-22.1.0-rc1.md
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## Change Log for Release asterisk-22.1.0-rc1
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### Links:
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- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0-rc1.md)
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- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0-rc1)
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- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0-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: 38
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- Commit Authors: 9
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- Issues Resolved: 21
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- Security Advisories Resolved: 0
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### User Notes:
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- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
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The new "suppress_moh_on_sendonly" endpoint option
|
||||
can be used to prevent playing MOH back to a caller if the remote
|
||||
end sends "sendonly" or "inactive" (hold) to Asterisk in an SDP.
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|
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- #### app_mixmonitor: Add 'D' option for dual-channel audio.
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The MixMonitor application now has a new 'D' option which
|
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interleaves the recorded audio in the output frames. This allows for
|
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stereo recording output with one channel being the transmitted audio and
|
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the other being the received audio. The 't' and 't' options are
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compatible with this.
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|
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- #### manager.c: Restrict ModuleLoad to the configured modules directory.
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The ModuleLoad AMI action now restricts modules to the
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configured modules directory.
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- #### manager: Enhance event filtering for performance
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You can now perform more granular filtering on events
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in manager.conf using expressions like
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`eventfilter(name(Newchannel),header(Channel),method(starts_with)) = PJSIP/`
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This is much more efficient than
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`eventfilter = Event: Newchannel.*Channel: PJSIP/`
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Full syntax guide is in configs/samples/manager.conf.sample.
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- #### db.c: Remove limit on family/key length
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The `ast_db_*()` APIs have had the 253 byte limit on
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"/family/key" removed and will now accept families and keys with a
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total length of up to SQLITE_MAX_LENGTH (currently 1e9!). This
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affects the `DB*` dialplan applications, dialplan functions,
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manager actions and `databse` CLI commands. Since the
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media_cache also uses the `ast_db_*()` APIs, you can now store
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resources with URIs longer than 253 bytes.
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### Upgrade Notes:
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### Commit Authors:
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- Allan Nathanson: (1)
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- Ben Ford: (3)
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- Chrsmj: (1)
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- George Joseph: (14)
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- Jiangxc: (1)
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- Naveen Albert: (7)
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- Peter Jannesen: (2)
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- Sean Bright: (7)
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- Thomas Guebels: (2)
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## Issue and Commit Detail:
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### Closed Issues:
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- 487: [bug]: Segfault possibly in ast_rtp_stop
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- 821: [bug]: app_dial: The progress timeout doesn't cause Dial to exit
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- 881: [bug]: Long URLs are being rejected by the media cache because of an astdb key length limit
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- 882: [bug]: Value CHANNEL(userfield) is lost by BRIDGE_ENTER
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- 897: [improvement]: Restrict ModuleLoad AMI action to the modules directory
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- 900: [bug]: astfd.c: NULL pointer passed to fclose with nonnull attribute causes compilation failure
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- 902: [bug]: app_voicemail: Pager emails are ill-formatted when custom subject is used
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- 916: [bug]: Compilation errors on FreeBSD
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- 923: [bug]: Transport monitor shutdown callback only works on the first disconnection
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- 924: [bug]: dnsmgr.c: dnsmgr_refresh() should not flag change if IP address order changes
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- 928: [bug]: chan_dahdi: MWI while off-hook when hung up on after recall ring
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- 932: [bug]: When connected to multiple IP addresses the transport monitor is only set on the first one
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- 937: [bug]: Wrong format for sample config file 'geolocation.conf.sample'
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- 938: [bug]: memory leak - CBAnn leaks a small amount format_cap related memory for every confbridge
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- 945: [improvement]: Add stereo recording support for app_mixmonitor
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- 951: [new-feature]: func_evalexten: Add `EVAL_SUB` function
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- 974: [improvement]: change and/or remove some wiki mentions to docs mentions in the sample configs
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- 979: [improvement]: Add ability to suppress MOH when a remote endpoint sends "sendonly" or "inactive"
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- 982: [bug]: The addition of tenantid to the ast_sip_endpoint structure broke ABI compatibility
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- 990: [improvement]: The help for PJSIP_AOR should indicate that you need to call PJSIP_CONTACT to get contact details
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### Commits By Author:
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- #### Allan Nathanson (1):
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- dnsmgr.c: dnsmgr_refresh() incorrectly flags change with DNS round-robin
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- #### Ben Ford (3):
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- manager.c: Restrict ModuleLoad to the configured modules directory.
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- app_mixmonitor: Add 'D' option for dual-channel audio.
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- Add res_pjsip_config_sangoma external module.
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- #### George Joseph (14):
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- db.c: Remove limit on family/key length
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- manager.c: Split XML documentation to manager_doc.xml
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- manager: Enhance event filtering for performance
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- manager.conf.sample: Fix mathcing typo
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- Fix application references to Background
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- res_rtp_asterisk: Fix dtls timer issues causing FRACKs and SEGVs
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- manager.c: Add unit test for Originate app and appdata permissions
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- geolocation.sample.conf: Fix comment marker at end of file
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- core_unreal.c: Fix memory leak in ast_unreal_new_channels()
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- pjproject_bundled: Tweaks to support out-of-tree development
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- res_srtp: Change Unsupported crypto suite msg from verbose to debug
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- res_pjsip: Move tenantid to end of ast_sip_endpoint
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- func_pjsip_aor/contact: Fix documentation for contact ID
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- res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
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- #### Naveen Albert (7):
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- app_voicemail: Fix ill-formatted pager emails with custom subject.
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- astfd.c: Avoid calling fclose with NULL argument.
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- main, res, tests: Fix compilation errors on FreeBSD.
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- chan_dahdi: Never send MWI while off-hook.
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- app_dial: Fix progress timeout.
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- app_dial: Fix progress timeout calculation with no answer timeout.
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- func_evalexten: Add EVAL_SUB function.
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- #### Peter Jannesen (2):
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- cel_custom: Allow absolute filenames.
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- channel: Preserve CHANNEL(userfield) on masquerade.
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- #### Sean Bright (7):
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- res_pjsip_pubsub: Persist subscription 'generator_data' in sorcery
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- cdr_custom: Allow absolute filenames.
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- res_agi.c: Ensure SIGCHLD handler functions are properly balanced.
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- alembic: Drop redundant voicemail_messages index.
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- func_base64.c: Ensure we set aside enough room for base64 encoded data.
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- Revert "res_rtp_asterisk: Count a roll-over of the sequence number even on los..
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- res_pjsip.c: Fix Contact header rendering for IPv6 addresses.
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- #### Thomas Guebels (2):
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- pjsip_transport_events: Avoid monitor destruction
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- pjsip_transport_events: handle multiple addresses for a domain
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- #### chrsmj (1):
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- samples: remove and/or change some wiki mentions
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- #### jiangxc (1):
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- res_agi.c: Prevent possible double free during `SPEECH RECOGNIZE`
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### Commit List:
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|
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- res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
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- res_pjsip.c: Fix Contact header rendering for IPv6 addresses.
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- samples: remove and/or change some wiki mentions
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- func_pjsip_aor/contact: Fix documentation for contact ID
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- res_pjsip: Move tenantid to end of ast_sip_endpoint
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- pjsip_transport_events: handle multiple addresses for a domain
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- func_evalexten: Add EVAL_SUB function.
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- res_srtp: Change Unsupported crypto suite msg from verbose to debug
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- Add res_pjsip_config_sangoma external module.
|
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- app_mixmonitor: Add 'D' option for dual-channel audio.
|
||||
- pjsip_transport_events: Avoid monitor destruction
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- app_dial: Fix progress timeout calculation with no answer timeout.
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- pjproject_bundled: Tweaks to support out-of-tree development
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- core_unreal.c: Fix memory leak in ast_unreal_new_channels()
|
||||
- dnsmgr.c: dnsmgr_refresh() incorrectly flags change with DNS round-robin
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||||
- geolocation.sample.conf: Fix comment marker at end of file
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||||
- func_base64.c: Ensure we set aside enough room for base64 encoded data.
|
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- app_dial: Fix progress timeout.
|
||||
- chan_dahdi: Never send MWI while off-hook.
|
||||
- manager.c: Add unit test for Originate app and appdata permissions
|
||||
- alembic: Drop redundant voicemail_messages index.
|
||||
- res_agi.c: Ensure SIGCHLD handler functions are properly balanced.
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- main, res, tests: Fix compilation errors on FreeBSD.
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||||
- res_rtp_asterisk: Fix dtls timer issues causing FRACKs and SEGVs
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||||
- manager.c: Restrict ModuleLoad to the configured modules directory.
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||||
- res_agi.c: Prevent possible double free during `SPEECH RECOGNIZE`
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- cdr_custom: Allow absolute filenames.
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||||
- astfd.c: Avoid calling fclose with NULL argument.
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||||
- channel: Preserve CHANNEL(userfield) on masquerade.
|
||||
- cel_custom: Allow absolute filenames.
|
||||
- app_voicemail: Fix ill-formatted pager emails with custom subject.
|
||||
- res_pjsip_pubsub: Persist subscription 'generator_data' in sorcery
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- Fix application references to Background
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- manager.conf.sample: Fix mathcing typo
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||||
- manager: Enhance event filtering for performance
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||||
- manager.c: Split XML documentation to manager_doc.xml
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||||
- db.c: Remove limit on family/key length
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||||
|
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### Commit Details:
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#### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
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Author: George Joseph
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Date: 2024-11-05
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Normally, when one party in a call sends Asterisk an SDP with
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a "sendonly" or "inactive" attribute it means "hold" and causes
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Asterisk to start playing MOH back to the other party. This can be
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problematic if it happens at certain times, such as in a 183
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Progress message, because the MOH will replace any early media you
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may be playing to the calling party. If you set this option
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to "yes" on an endpoint and the endpoint receives an SDP
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with "sendonly" or "inactive", Asterisk will NOT play MOH back to
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the other party.
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Resolves: #979
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UserNote: The new "suppress_moh_on_sendonly" endpoint option
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can be used to prevent playing MOH back to a caller if the remote
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end sends "sendonly" or "inactive" (hold) to Asterisk in an SDP.
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#### res_pjsip.c: Fix Contact header rendering for IPv6 addresses.
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Author: Sean Bright
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Date: 2024-11-08
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Fix suggested by @nvsystems.
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Fixes #985
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#### samples: remove and/or change some wiki mentions
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Author: chrsmj
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Date: 2024-11-01
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Cleaned some dead links. Replaced word wiki with
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either docs or link to https://docs.asterisk.org/
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Resolves: #974
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#### func_pjsip_aor/contact: Fix documentation for contact ID
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Author: George Joseph
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Date: 2024-11-09
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Clarified the use of the contact ID returned from PJSIP_AOR.
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Resolves: #990
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#### res_pjsip: Move tenantid to end of ast_sip_endpoint
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Author: George Joseph
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Date: 2024-11-06
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The tenantid field was originally added to the ast_sip_endpoint
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structure at the end of the AST_DECLARE_STRING_FIELDS block. This
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caused everything after it in the structure to move down in memory
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and break ABI compatibility. It's now at the end of the structure
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as an AST_STRING_FIELD_EXTENDED. Given the number of string fields
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in the structure now, the initial string field allocation was
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also increased from 64 to 128 bytes.
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Resolves: #982
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#### pjsip_transport_events: handle multiple addresses for a domain
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Author: Thomas Guebels
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Date: 2024-10-29
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The key used for transport monitors was the remote host name for the
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transport and not the remote address resolved for this domain.
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|
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This was problematic for domains returning multiple addresses as several
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transport monitors were created with the same key.
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Whenever a subsystem wanted to register a callback it would always end
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up attached to the first transport monitor with a matching key.
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|
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The key used for transport monitors is now the remote address and port
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the transport actually connected to.
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Fixes: #932
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|
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#### func_evalexten: Add EVAL_SUB function.
|
||||
Author: Naveen Albert
|
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Date: 2024-10-17
|
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|
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This adds an EVAL_SUB function, which is similar to the existing
|
||||
EVAL_EXTEN function but significantly more powerful, as it allows
|
||||
executing arbitrary dialplan and capturing its return value as
|
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the function's output. While EVAL_EXTEN should be preferred if it
|
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is possible to use it, EVAL_SUB can be used in a wider variety
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of cases and allows arbitrary computation to be performed in
|
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a dialplan function call, leveraging the dialplan.
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Resolves: #951
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|
||||
#### res_srtp: Change Unsupported crypto suite msg from verbose to debug
|
||||
Author: George Joseph
|
||||
Date: 2024-11-01
|
||||
|
||||
There's really no point in spamming logs with a verbose message
|
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for every unsupported crypto suite an older client may send
|
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in an SDP. If none are supported, there will be an error or
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||||
warning.
|
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|
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|
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#### Add res_pjsip_config_sangoma external module.
|
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Author: Ben Ford
|
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Date: 2024-11-01
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||||
|
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Adds res_pjsip_config_sangoma as an external module that can be
|
||||
downloaded via menuselect. It lives under the Resource Modules section.
|
||||
|
||||
|
||||
#### app_mixmonitor: Add 'D' option for dual-channel audio.
|
||||
Author: Ben Ford
|
||||
Date: 2024-10-28
|
||||
|
||||
Adds the 'D' option to app_mixmonitor that interleaves the input and
|
||||
output frames of the channel being recorded in the monitor output frame.
|
||||
This allows for two streams in the recording: the transmitted audio and
|
||||
the received audio. The 't' and 'r' options are compatible with this.
|
||||
|
||||
Fixes: #945
|
||||
|
||||
UserNote: The MixMonitor application now has a new 'D' option which
|
||||
interleaves the recorded audio in the output frames. This allows for
|
||||
stereo recording output with one channel being the transmitted audio and
|
||||
the other being the received audio. The 't' and 't' options are
|
||||
compatible with this.
|
||||
|
||||
|
||||
#### pjsip_transport_events: Avoid monitor destruction
|
||||
Author: Thomas Guebels
|
||||
Date: 2024-10-28
|
||||
|
||||
When a transport is disconnected, several events can arrive following
|
||||
each other. The first event will be PJSIP_TP_STATE_DISCONNECT and it
|
||||
will trigger the destruction of the transport monitor object. The lookup
|
||||
for the transport monitor to destroy is done using the transport key,
|
||||
that contains the transport destination host:port.
|
||||
|
||||
A reconnect attempt by pjsip will be triggered as soon something needs to
|
||||
send a packet using that transport. This can happen directly after a
|
||||
disconnect since ca
|
||||
|
||||
Subsequent events can arrive later like PJSIP_TP_STATE_DESTROY and will
|
||||
also try to trigger the destruction of the transport monitor if not
|
||||
already done. Since the lookup for the transport monitor to destroy is
|
||||
done using the transport key, it can match newly created transports
|
||||
towards the same destination and destroy their monitor object.
|
||||
|
||||
Because of this, it was sometimes not possible to monitor a transport
|
||||
after one or more disconnections.
|
||||
|
||||
This fix adds an additional check on the transport pointer to ensure
|
||||
only a monitor for that specific transport is removed.
|
||||
|
||||
Fixes: #923
|
||||
|
||||
#### app_dial: Fix progress timeout calculation with no answer timeout.
|
||||
Author: Naveen Albert
|
||||
Date: 2024-10-16
|
||||
|
||||
If to_answer is -1, simply comparing to see if the progress timeout
|
||||
is smaller than the answer timeout to prefer it will fail. Add
|
||||
an additional check that chooses the progress timeout if there is
|
||||
no answer timeout (or as before, if the progress timeout is smaller).
|
||||
|
||||
Resolves: #821
|
||||
|
||||
#### pjproject_bundled: Tweaks to support out-of-tree development
|
||||
Author: George Joseph
|
||||
Date: 2024-10-17
|
||||
|
||||
* pjproject is now configured with --disable-libsrtp so it will
|
||||
build correctly when doing "out-of-tree" development. Asterisk
|
||||
doesn't use pjproject for handling media so pjproject doesn't
|
||||
need libsrtp itself.
|
||||
|
||||
* The pjsua app (which we used to use for the testsuite) no longer
|
||||
builds in pjproject's master branch so we just skip it. The
|
||||
testsuite no longer needs it anyway.
|
||||
|
||||
See third-party/pjproject/README-hacking.md for more info on building
|
||||
pjproject "out-of-tree".
|
||||
|
||||
|
||||
#### Revert "res_rtp_asterisk: Count a roll-over of the sequence number even on los..
|
||||
Author: Sean Bright
|
||||
Date: 2024-10-07
|
||||
|
||||
This reverts commit cb5e3445be6c55517c8d05aca601b648341f8ae9.
|
||||
|
||||
The original change from 16 to 15 bit sequence numbers was predicated
|
||||
on the following from the now-defunct libSRTP FAQ on sourceforge.net:
|
||||
|
||||
> *Q6. The use of implicit synchronization via ROC seems
|
||||
> dangerous. Can senders and receivers lose ROC synchronization?*
|
||||
>
|
||||
> **A.** It is possible to lose ROC synchronization between sender and
|
||||
> receiver(s), though it is not likely in practice, and practical
|
||||
> steps can be taken to avoid it. A burst loss of 2^16 packets or more
|
||||
> will always break synchronization. For example, a conversational
|
||||
> voice codec that sends 50 packets per second will have its ROC
|
||||
> increment about every 22 minutes. A network with a burst of packet
|
||||
> loss that long has problems other than ROC synchronization.
|
||||
>
|
||||
> There is a higher sensitivity to loss at the very outset of an SRTP
|
||||
> stream. If the sender's initial sequence number is close to the
|
||||
> maximum value of 2^16-1, and all packets are lost from the initial
|
||||
> packet until the sequence number cycles back to zero, the sender
|
||||
> will increment its ROC, but the receiver will not. The receiver
|
||||
> cannot determine that the initial packets were lost and that
|
||||
> sequence-number rollover has occurred. In this case, the receiver's
|
||||
> ROC would be zero whereas the sender's ROC would be one, while their
|
||||
> sequence numbers would be so close that the ROC-guessing algorithm
|
||||
> could not detect this fact.
|
||||
>
|
||||
> There is a simple solution to this problem: the SRTP sender should
|
||||
> randomly select an initial sequence number that is always less than
|
||||
> 2^15. This ensures correct SRTP operation so long as fewer than 2^15
|
||||
> initial packets are lost in succession, which is within the maximum
|
||||
> tolerance of SRTP packet-index determination (see Appendix A and
|
||||
> page 14, first paragraph of RFC 3711). An SRTP receiver should
|
||||
> carefully implement the index-guessing algorithm. A naive
|
||||
> implementation can unintentionally guess the value of
|
||||
> 0xffffffffffffLL whenever the SEQ in the packet is greater than 2^15
|
||||
> and the locally stored SEQ and ROC are zero. (This can happen when
|
||||
> the implementation fails to treat those zero values as a special
|
||||
> case.)
|
||||
>
|
||||
> When ROC synchronization is lost, the receiver will not be able to
|
||||
> properly process the packets. If anti-replay protection is turned
|
||||
> on, then the desynchronization will appear as a burst of replay
|
||||
> check failures. Otherwise, if authentication is being checked, then
|
||||
> it will appear as a burst of authentication failures. Otherwise, if
|
||||
> encryption is being used, the desynchronization may not be detected
|
||||
> by the SRTP layer, and the packets may be improperly decrypted.
|
||||
|
||||
However, modern libSRTP (as of 1.0.1[1]) now mentions the following in
|
||||
their README.md[2]:
|
||||
|
||||
> The sequence number in the rtp packet is used as the low 16 bits of
|
||||
> the sender's local packet index. Note that RTP will start its
|
||||
> sequence number in a random place, and the SRTP layer just jumps
|
||||
> forward to that number at its first invocation. An earlier version
|
||||
> of this library used initial sequence numbers that are less than
|
||||
> 32,768; this trick is no longer required as the
|
||||
> rdbx_estimate_index(...) function has been made smarter.
|
||||
|
||||
So truncating our initial sequence number to 15 bit is no longer
|
||||
necessary.
|
||||
|
||||
1. https://github.com/cisco/libsrtp/blob/0eb007f0dc611f27cbfe0bf9855ed85182496cec/CHANGES#L271-L289
|
||||
2. https://github.com/cisco/libsrtp/blob/2de20dd9e9c8afbaf02fcf5d4048ce1ec9ddc0ae/README.md#implementation-notes
|
||||
|
||||
|
||||
#### core_unreal.c: Fix memory leak in ast_unreal_new_channels()
|
||||
Author: George Joseph
|
||||
Date: 2024-10-15
|
||||
|
||||
When the channel tech is multistream capable, the reference to
|
||||
chan_topology was passed to the new channel. When the channel tech
|
||||
isn't multistream capable, the reference to chan_topology was never
|
||||
released. "Local" channels are multistream capable so it didn't
|
||||
affect them but the confbridge "CBAnn" and the bridge_media
|
||||
"Recorder" channels are not so they caused a leak every time one
|
||||
of them was created.
|
||||
|
||||
Also added tracing to ast_stream_topology_alloc() and
|
||||
stream_topology_destroy() to assist with debugging.
|
||||
|
||||
Resolves: #938
|
||||
|
||||
#### dnsmgr.c: dnsmgr_refresh() incorrectly flags change with DNS round-robin
|
||||
Author: Allan Nathanson
|
||||
Date: 2024-09-29
|
||||
|
||||
The dnsmgr_refresh() function checks to see if the IP address associated
|
||||
with a name/service has changed. The gotcha is that the ast_get_ip_or_srv()
|
||||
function only returns the first IP address returned by the DNS query. If
|
||||
there are multiple IPs associated with the name and the returned order is
|
||||
not consistent (e.g. with DNS round-robin) then the other IP addresses are
|
||||
not included in the comparison and the entry is flagged as changed even
|
||||
though the IP is still valid.
|
||||
|
||||
Updated the code to check all IP addresses and flag a change only if the
|
||||
original IP is no longer valid.
|
||||
|
||||
Resolves: #924
|
||||
|
||||
#### geolocation.sample.conf: Fix comment marker at end of file
|
||||
Author: George Joseph
|
||||
Date: 2024-10-08
|
||||
|
||||
Resolves: #937
|
||||
|
||||
#### func_base64.c: Ensure we set aside enough room for base64 encoded data.
|
||||
Author: Sean Bright
|
||||
Date: 2024-10-08
|
||||
|
||||
Reported by SingularTricycle on IRC.
|
||||
|
||||
Fixes #940
|
||||
|
||||
|
||||
#### app_dial: Fix progress timeout.
|
||||
Author: Naveen Albert
|
||||
Date: 2024-10-03
|
||||
|
||||
Under some circumstances, the progress timeout feature added in commit
|
||||
320c98eec87c473bfa814f76188a37603ea65ddd does not work as expected,
|
||||
such as if there is no media flowing. Adjust the waitfor call to
|
||||
explicitly use the progress timeout if it would be reached sooner than
|
||||
the answer timeout to ensure we handle the timers properly.
|
||||
|
||||
Resolves: #821
|
||||
|
||||
#### chan_dahdi: Never send MWI while off-hook.
|
||||
Author: Naveen Albert
|
||||
Date: 2024-10-01
|
||||
|
||||
In some circumstances, it is possible for the do_monitor thread to
|
||||
erroneously think that a line is on-hook and send an MWI FSK spill
|
||||
to it when the line is really off-hook and no MWI should be sent.
|
||||
Commit 0a8b3d34673277b70be6b0e8ac50191b1f3c72c6 previously fixed this
|
||||
issue in a more readily encountered scenario, but it has still been
|
||||
possible for MWI to be sent when it shouldn't be. To robustly fix
|
||||
this issue, query DAHDI for the hook status to ensure we don't send
|
||||
MWI on a line that is actually still off hook.
|
||||
|
||||
Resolves: #928
|
||||
|
||||
#### manager.c: Add unit test for Originate app and appdata permissions
|
||||
Author: George Joseph
|
||||
Date: 2024-10-03
|
||||
|
||||
This unit test checks that dialplan apps and app data specified
|
||||
as parameters for the Originate action are allowed with the
|
||||
permissions the user has.
|
||||
|
||||
|
||||
#### alembic: Drop redundant voicemail_messages index.
|
||||
Author: Sean Bright
|
||||
Date: 2024-09-26
|
||||
|
||||
The `voicemail_messages_dir` index is a left prefix of the table's
|
||||
primary key and therefore unnecessary.
|
||||
|
||||
|
||||
#### res_agi.c: Ensure SIGCHLD handler functions are properly balanced.
|
||||
Author: Sean Bright
|
||||
Date: 2024-09-30
|
||||
|
||||
Calls to `ast_replace_sigchld()` and `ast_unreplace_sigchld()` must be
|
||||
balanced to ensure that we can capture the exit status of child
|
||||
processes when we need to. This extends to functions that call
|
||||
`ast_replace_sigchld()` and `ast_unreplace_sigchld()` such as
|
||||
`ast_safe_fork()` and `ast_safe_fork_cleanup()`.
|
||||
|
||||
The primary change here is ensuring that we do not call
|
||||
`ast_safe_fork_cleanup()` in `res_agi.c` if we have not previously
|
||||
called `ast_safe_fork()`.
|
||||
|
||||
Additionally we reinforce some of the documentation and add an
|
||||
assertion to, ideally, catch this sooner were this to happen again.
|
||||
|
||||
Fixes #922
|
||||
|
||||
|
||||
#### main, res, tests: Fix compilation errors on FreeBSD.
|
||||
Author: Naveen Albert
|
||||
Date: 2024-09-29
|
||||
|
||||
asterisk.c, manager.c: Increase buffer sizes to avoid truncation warnings.
|
||||
config.c: Include header file for WIFEXITED/WEXITSTATUS macros.
|
||||
res_timing_kqueue: Use more portable format specifier.
|
||||
test_crypto: Use non-linux limits.h header file.
|
||||
|
||||
Resolves: #916
|
||||
|
||||
#### res_rtp_asterisk: Fix dtls timer issues causing FRACKs and SEGVs
|
||||
Author: George Joseph
|
||||
Date: 2024-09-16
|
||||
|
||||
In dtls_srtp_handle_timeout(), when DTLSv1_get_timeout() returned
|
||||
success but with a timeout of 0, we were stopping the timer and
|
||||
decrementing the refcount on instance but not resetting the
|
||||
timeout_timer to -1. When dtls_srtp_stop_timeout_timer()
|
||||
was later called, it was atempting to stop a stale timer and could
|
||||
decrement the refcount on instance again which would then cause
|
||||
the instance destructor to run early. This would result in either
|
||||
a FRACK or a SEGV when ast_rtp_stop(0 was called.
|
||||
|
||||
According to the OpenSSL docs, we shouldn't have been stopping the
|
||||
timer when DTLSv1_get_timeout() returned success and the new timeout
|
||||
was 0 anyway. We should have been calling DTLSv1_handle_timeout()
|
||||
again immediately so we now reschedule the timer callback for
|
||||
1ms (almost immediately).
|
||||
|
||||
Additionally, instead of scheduling the timer callback at a fixed
|
||||
interval returned by the initial call to DTLSv1_get_timeout()
|
||||
(usually 999 ms), we now reschedule the next callback based on
|
||||
the last call to DTLSv1_get_timeout().
|
||||
|
||||
Resolves: #487
|
||||
|
||||
#### manager.c: Restrict ModuleLoad to the configured modules directory.
|
||||
Author: Ben Ford
|
||||
Date: 2024-09-25
|
||||
|
||||
When using the ModuleLoad AMI action, it was possible to traverse
|
||||
upwards through the directories to files outside of the configured
|
||||
modules directory. We decided it would be best to restrict access to
|
||||
modules exclusively in the configured directory. You will now get an
|
||||
error when the specified module is outside of this limitation.
|
||||
|
||||
Fixes: #897
|
||||
|
||||
UserNote: The ModuleLoad AMI action now restricts modules to the
|
||||
configured modules directory.
|
||||
|
||||
|
||||
#### res_agi.c: Prevent possible double free during `SPEECH RECOGNIZE`
|
||||
Author: jiangxc
|
||||
Date: 2024-07-17
|
||||
|
||||
When using the speech recognition module, crashes can occur
|
||||
sporadically due to a "double free or corruption (out)" error. Now, in
|
||||
the section where the audio stream is being captured in a loop, each
|
||||
time after releasing fr, it is set to NULL to prevent repeated
|
||||
deallocation.
|
||||
|
||||
Fixes #772
|
||||
|
||||
|
||||
#### cdr_custom: Allow absolute filenames.
|
||||
Author: Sean Bright
|
||||
Date: 2024-09-26
|
||||
|
||||
A follow up to #893 that brings the same functionality to
|
||||
cdr_custom. Also update the sample configuration files to note support
|
||||
for absolute paths.
|
||||
|
||||
|
||||
#### astfd.c: Avoid calling fclose with NULL argument.
|
||||
Author: Naveen Albert
|
||||
Date: 2024-09-24
|
||||
|
||||
Don't pass through a NULL argument to fclose, which is undefined
|
||||
behavior, and instead return -1 and set errno appropriately. This
|
||||
also avoids a compiler warning with glibc 2.38 and newer, as glibc
|
||||
commit 71d9e0fe766a3c22a730995b9d024960970670af
|
||||
added the nonnull attribute to this argument.
|
||||
|
||||
Resolves: #900
|
||||
|
||||
#### channel: Preserve CHANNEL(userfield) on masquerade.
|
||||
Author: Peter Jannesen
|
||||
Date: 2024-09-20
|
||||
|
||||
In certain circumstances a channel may undergo an operation
|
||||
referred to as a masquerade. If this occurs the CHANNEL(userfield)
|
||||
value was not preserved causing it to get lost. This change makes
|
||||
it so that this field is now preserved.
|
||||
|
||||
Fixes: #882
|
||||
|
||||
#### cel_custom: Allow absolute filenames.
|
||||
Author: Peter Jannesen
|
||||
Date: 2024-09-20
|
||||
|
||||
If a filename starts with a '/' in cel_custom [mappings] assume it is
|
||||
a absolute file path and not relative filename/path to
|
||||
AST_LOG_DIR/cel_custom/
|
||||
|
||||
|
||||
#### app_voicemail: Fix ill-formatted pager emails with custom subject.
|
||||
Author: Naveen Albert
|
||||
Date: 2024-09-24
|
||||
|
||||
Add missing end-of-headers newline to pager emails with custom
|
||||
subjects, since this was missing from this code path.
|
||||
|
||||
Resolves: #902
|
||||
|
||||
#### res_pjsip_pubsub: Persist subscription 'generator_data' in sorcery
|
||||
Author: Sean Bright
|
||||
Date: 2024-09-23
|
||||
|
||||
Fixes #895
|
||||
|
||||
|
||||
#### Fix application references to Background
|
||||
Author: George Joseph
|
||||
Date: 2024-09-20
|
||||
|
||||
The app is actually named "BackGround" but several references
|
||||
in XML documentation were spelled "Background" with the lower
|
||||
case "g". This was causing documentation links to return
|
||||
"not found" messages.
|
||||
|
||||
|
||||
#### manager.conf.sample: Fix mathcing typo
|
||||
Author: George Joseph
|
||||
Date: 2024-09-24
|
||||
|
||||
|
||||
#### manager: Enhance event filtering for performance
|
||||
Author: George Joseph
|
||||
Date: 2024-07-31
|
||||
|
||||
UserNote: You can now perform more granular filtering on events
|
||||
in manager.conf using expressions like
|
||||
`eventfilter(name(Newchannel),header(Channel),method(starts_with)) = PJSIP/`
|
||||
This is much more efficient than
|
||||
`eventfilter = Event: Newchannel.*Channel: PJSIP/`
|
||||
Full syntax guide is in configs/samples/manager.conf.sample.
|
||||
|
||||
|
||||
#### manager.c: Split XML documentation to manager_doc.xml
|
||||
Author: George Joseph
|
||||
Date: 2024-08-01
|
||||
|
||||
|
||||
#### db.c: Remove limit on family/key length
|
||||
Author: George Joseph
|
||||
Date: 2024-09-11
|
||||
|
||||
Consumers like media_cache have been running into issues with
|
||||
the previous astdb "/family/key" limit of 253 bytes when needing
|
||||
to store things like long URIs. An Amazon S3 URI is a good example
|
||||
of this. Now, instead of using a static 256 byte buffer for
|
||||
"/family/key", we use ast_asprintf() to dynamically create it.
|
||||
|
||||
Both test_db.c and test_media_cache.c were also updated to use
|
||||
keys/URIs over the old 253 character limit.
|
||||
|
||||
Resolves: #881
|
||||
|
||||
UserNote: The `ast_db_*()` APIs have had the 253 byte limit on
|
||||
"/family/key" removed and will now accept families and keys with a
|
||||
total length of up to SQLITE_MAX_LENGTH (currently 1e9!). This
|
||||
affects the `DB*` dialplan applications, dialplan functions,
|
||||
manager actions and `databse` CLI commands. Since the
|
||||
media_cache also uses the `ast_db_*()` APIs, you can now store
|
||||
resources with URIs longer than 253 bytes.
|
||||
|
||||
|
||||
Loading…
Reference in new issue