Asterisk Development Team
2019-Apr-04 15:59 UTC
[asterisk-users] Asterisk 16.3.0 Now Available
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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