The Asterisk Development Team would like to announce
the release of asterisk-20.7.0.

The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/20.7.0
and
https://downloads.asterisk.org/pub/telephony/asterisk

Repository: https://github.com/asterisk/asterisk
Tag: 20.7.0

This release resolves issues reported by the community
and would have not been possible without your participation.

Thank You!

Change Log for Release asterisk-20.7.0

Links:

• Full ChangeLog
• GitHub Diff
• Tarball
• Downloads

Summary:

• res_pjsip_stir_shaken.c: Add checks for missing parameters
• app_dial: Add dial time for progress/ringing.
• app_voicemail: Properly reinitialize config after unit tests.
• app_queue.c : fix "queue add member" usage string
• app_voicemail: Allow preventing mark messages as urgent.
• res_pjsip: Use consistent type for boolean columns.
• attestation_config.c: Use ast_free instead of ast_std_free
• Makefile: Add stir_shaken/cache to directories created on install
• Stir/Shaken Refactor
• alembic: Synchronize alembic heads between supported branches.
• translate.c: implement new direct comp table mode
• README.md: Removed outdated link
• strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
• res_rtp_asterisk.c: Correct coefficient in MOS calculation.
• dsp.c: Fix and improve potentially inaccurate log message.
• pjsip show channelstats: Prevent possible segfault when faxing
• Reduce startup/shutdown verbose logging
• configure: Rerun bootstrap on modern platform.
• Upgrade bundled pjproject to 2.14.
• app_speech_utils.c: Allow partial speech results.
• utils: Make behavior of ast_strsep* match strsep.
• app_chanspy: Add 'D' option for dual-channel audio
• app_if: Fix next priority calculation.
• res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
• BuildSystem: Bump autotools versions on OpenBSD.
• main/utils: Simplify the FreeBSD ast_get_tid() handling
• res_pjsip_session.c: Correctly format SDP connection addresses.
• rtp_engine.c: Correct sample rate typo for L16/44100.
• manager.c: Fix erroneous reloads in UpdateConfig.
• res_calendar_icalendar: Print iCalendar error on parsing failure.
• app_confbridge: Don't emit warnings on valid configurations.
• app_voicemail: add NoOp alembic script to maintain sync
• chan_dahdi: Allow MWI to be manually toggled on channels.
• chan_rtp.c: MulticastRTP missing refcount without codec option
• chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
• func_frame_trace: Add CLI command to dump frame queue.

User Notes:

• app_dial: Add dial time for progress/ringing.

  The timeout argument to Dial now allows
  specifying the maximum amount of time to dial if
  early media is not received.

• app_voicemail: Allow preventing mark messages as urgent.

  The leaveurgent mailbox option can now be used to
  control whether callers may leave messages marked as 'Urgent'.

• Stir/Shaken Refactor

  Asterisk's stir-shaken feature has been refactored to
  correct interoperability, RFC compliance, and performance issues.
  See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more
  information.

• Upgrade bundled pjproject to 2.14.

  Bundled pjproject has been upgraded to 2.14. For more
  information on what all is included in this change, check out the
  pjproject Github page: https://github.com/pjsip/pjproject/releases

• app_speech_utils.c: Allow partial speech results.

  The SpeechBackground dialplan application now supports a 'p'
  option that will return partial results from speech engines that
  provide them when a timeout occurs.

• app_chanspy: Add 'D' option for dual-channel audio

  The ChanSpy application now accepts the 'D' option which
  will interleave the spied audio within the outgoing frames. The
  purpose of this is to allow the audio to be read as a Dual channel
  stream with separate incoming and outgoing audio. Setting both the
  'o' option and the 'D' option and results in the 'D' option being
  ignored.

• chan_dahdi: Allow MWI to be manually toggled on channels.

  The 'dahdi set mwi' now allows MWI on channels
  to be manually toggled if needed for troubleshooting.
  Resolves: #440

Upgrade Notes:

• Stir/Shaken Refactor

  The stir-shaken refactor is a breaking change but since
  it's not working now we don't think it matters. The
  stir_shaken.conf file has changed significantly which means that
  existing ones WILL need to be changed. The stir_shaken.conf.sample
  file in configs/samples/ has quite a bit more information. This is
  also an ABI breaking change since some of the existing objects
  needed to be changed or removed, and new ones added. Additionally,
  if res_stir_shaken is enabled in menuselect, you'll need to either
  have the development package for libjwt v1.15.3 installed or use
  the --with-libjwt-bundled option with ./configure.

Closed Issues:

• #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
• #351: [improvement]: Refactor res_stir_shaken to use libjwt
• #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
• #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
• #492: [improvement]: res_calendar_icalendar: Print icalendar error if available on parsing failure
• #527: [bug]: app_voicemail_odbc no longer working after removal of macrocontext.
• #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed assertion bad magic number 0x0 for object" after ~30 calls
• #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of channel frame queue
• #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
• #560: [bug]: EndIf() causes next priority to be skipped
• #565: [bug]: Application Read() returns immediately
• #569: [improvement]: Add option to interleave input and output frames on spied channel
• #572: [improvement]: Copy partial speech results when Asterisk is ready to move on but the speech backend is not
• #582: [improvement]: Reduce unneeded logging during startup and shutdown
• #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in older gcc versions
• #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not received
• #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault when a fax session is active
• #595: [improvement]: dsp.c: Fix and improve confusing warning message.
• #597: [bug]: wrong MOS calculation
• #601: [new-feature]: translate.c: implement new direct comp table mode (PR #585)
• #619: [new-feature]: app_voicemail: Allow preventing callers from marking messages as urgent
• #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
• #634: [bug]: make install doesn't create the stir_shaken cache directory
• #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
• #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c