Asterisk 16.3.0 Now Available

Asterisk Asterisk News
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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 César
    Benjamín García Martínez)

  • 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 Lainé)

  • 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!