[asterisk-dev] Asterisk 16.3.0-rc1 Now Available

[Asterisk Development Team]

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

The release of Asterisk 16.3.0-rc1 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 candidate:

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 candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.3.0-rc1

Thank you for your continued support of Asterisk!
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