[asterisk-announce] Asterisk 16.3.0 Now Available

[Asterisk Development Team]

The Asterisk Development Team would like to announce the release of Asterisk 16.3.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.3.0 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

Security bugs fixed in this release:
-----------------------------------
 * ASTERISK-28260 - Asterisk segfault when rtp negotiation is
      wrong or fails
      (Reported by Sotiris Ganouris)

New Features made in this release:
-----------------------------------
 * ASTERISK-28267 - res_stasis: Add ability to switch
      applications
      (Reported by Benjamin Keith Ford)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-27541 - app_queue: Queue paused reason was (big
      number) secs ago when reason is set
      (Reported by C��sar
      Benjam��n Garc��a Mart��nez)
 * ASTERISK-20986 - QUEUE_MEMBER 's description is inaccurate
  
      (Reported by Olivier Krief)
 * ASTERISK-28350 - manager: Stasis backed up due to locking
   
      (Reported by Joshua C. Colp)
 * ASTERISK-25792 - chan_sip: qualifygap bounds checking
     
      (Reported by Paul Sandys)
 * ASTERISK-28341 - res_config_odbc eliminates empty custom (���@���
      prefix) variables 
      (Reported by Alexei Gradinari)
 * ASTERISK-28333 - StasisEnd event makes wrong timestamp value

      (Reported by sungtae kim)
 * ASTERISK-28306 - res_pjsip_mwi: MWI NOTIFY occasionally takes
      minutes to be sent
      (Reported by Jared Hull)
 * ASTERISK-28332 - Variable ALTCONF ignored when service is
      used in Debian
      (Reported by Cirillo Ferreira)
 * ASTERISK-28314 - ARI: API changed but "apiVersion" in
      rest-api\resources.json did not
      (Reported by Stefan Repke)
 * ASTERISK-28335 - stasis: Make topic and maybe subscription
      names unique and more useful
      (Reported by Joshua C. Colp)
 * ASTERISK-28321 - res_rtp_asterisk: Fixing possible divide by
      zero for rtcp stat calculation
      (Reported by sungtae kim)
 * ASTERISK-28322 - chan_pjsip: Add option to allow ignoring of
      183 without SDP
      (Reported by Torrey Searle)
 * ASTERISK-28328 - MeetMe global non-admin mute is muting
      admins that subsequently join
      (Reported by Philip Mott)
 * ASTERISK-27964 - app_queue: ring_entry accesses nativeformats
      without channel lock or reference
      (Reported by Francisco
      Seratti)
 * ASTERISK-28168 - app_queue: Adding a blank entry into sql
      queue_members crashes asterisk.
      (Reported by Michael)
 * ASTERISK-28323 - pjsip: sip.conf to pjsip.conf conversion
      script fails
      (Reported by Guido Weckwerth)
 * ASTERISK-28272 - The basic-pbx config samples don't produce a
      running asterisk
      (Reported by George Joseph)
 * ASTERISK-28312 - res_pjsip_diversion: Corrupted SIP Diversion
      field after handling a 302 redirect
      (Reported by Alex
      Odrov)
 * ASTERISK-24173 - File menuselect/menuselect_gtk.c has no
      license header
      (Reported by Jeremy Lain��)
 * ASTERISK-28309 - res_pjsip: Wrong Contact and Via fields with
      multiple UDP interfaces
      (Reported by Nikolay shakin)
 * ASTERISK-27992 - PJSIP: Adding `sends_registrations = yes` to
      pjsip_wizard.conf  causes crash
      (Reported by Jonathan
      Harris)
 * ASTERISK-28166 - app_voicemail: Asterisk unresponsive after
      changing voicemail password with ODBC
      (Reported by
      Michael)
 * ASTERISK-28213 - res_pjsip: Threads pile up needlessly when
      AOR is blocked
      (Reported by Ross Beer)
 * ASTERISK-28301 - Allow voicemail boxes to be subscribed to
      with a presence event package
      (Reported by George Joseph)
 * ASTERISK-28303 - res_rtp_asterisk: Interaction between
      smoother and DTMF can cause out of order timestamps
     
      (Reported by Torrey Searle)
 * ASTERISK-28302 - ARI: "Error destroying mutex" when listing
      all ARI applications
      (Reported by Stefan Repke)
 * ASTERISK-28300 - AST_PBX_MAX_STACK is too low for some
      applications
      (Reported by George Joseph)
 * ASTERISK-28106 - Astricon Feedback: Unable to filter ARI
      events when GETting causes overload of events
      (Reported by
      George Joseph)
 * ASTERISK-28284 - switching between native_bridge and
      simple_bridge can cause one way audio
      (Reported by Torrey
      Searle)
 * ASTERISK-28251 - CI: Fix CI so it reverifies commit message
      changes
      (Reported by George Joseph)
 * ASTERISK-28277 - database: Add some basic logging
     
      (Reported by Joshua C. Colp)
 * ASTERISK-28181 - ari: Originating overwrites channel start
      time
      (Reported by sungtae kim)

Improvements made in this release:
-----------------------------------
 * ASTERISK-28326 - ari: Added timestamp for some ari events.
  
      (Reported by sungtae kim)
 * ASTERISK-28317 - Add logical group at DAHDIChannel event and
      create "dahdi_group" at CHANNEL function
      (Reported by
      Cirillo Ferreira)
 * ASTERISK-28279 - Added creation timestamp for bridge
     
      (Reported by sungtae kim)
 * ASTERISK-27483 - Allow wrapuptime to be set for each queue
      member
      (Reported by Rodrigo Ramirez Norambuena)
 * ASTERISK-28055 - app_queue: Per-member wrapup time missing
      from AddQueueMember application
      (Reported by Niksa Baldun)
 * ASTERISK-28292 - Changed to show all channel stats including
      wrong media
      (Reported by sungtae kim)
 * ASTERISK-28253 - res_pjsip_session: Adding rtcp stats result
      into the session
      (Reported by sungtae kim)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.3.0

Thank you for your continued support of Asterisk!
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