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 segfault when rtp negotiation is wrong or fails | (Reported by Sotiris Ganouris) |
New Features made in this release:
-----------------------------------
| res_stasis: Add ability to switch applications | (Reported by Benjamin Keith Ford) |
Bugs fixed in this release:
-----------------------------------
| app_queue: Queue paused reason was (big number) secs ago when reason is set | (Reported by César BenjamÃn GarcÃa MartÃnez) | |
| QUEUE_MEMBER 's description is inaccurate | (Reported by Olivier Krief) | |
| manager: Stasis backed up due to locking | (Reported by Joshua C. Colp) | |
| chan_sip: qualifygap bounds checking | (Reported by Paul Sandys) | |
| res_config_odbc eliminates empty custom (â??@â?? prefix) variables | (Reported by Alexei Gradinari) | |
| StasisEnd event makes wrong timestamp value | (Reported by sungtae kim) | |
| res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be sent | (Reported by Jared Hull) | |
| Variable ALTCONF ignored when service is used in Debian | (Reported by Cirillo Ferreira) | |
| ARI: API changed but "apiVersion" in rest-api\resources.json did not | (Reported by Stefan Repke) | |
| stasis: Make topic and maybe subscription names unique and more useful | (Reported by Joshua C. Colp) | |
| res_rtp_asterisk: Fixing possible divide by zero for rtcp stat calculation | (Reported by sungtae kim) | |
| chan_pjsip: Add option to allow ignoring of 183 without SDP | (Reported by Torrey Searle) | |
| MeetMe global non-admin mute is muting admins that subsequently join | (Reported by Philip Mott) | |
| app_queue: ring_entry accesses nativeformats without channel lock or reference | (Reported by Francisco Seratti) | |
| app_queue: Adding a blank entry into sql queue_members crashes asterisk. | (Reported by Michael) | |
| pjsip: sip.conf to pjsip.conf conversion script fails | (Reported by Guido Weckwerth) | |
| The basic-pbx config samples don't produce a running asterisk | (Reported by George Joseph) | |
| res_pjsip_diversion: Corrupted SIP Diversion field after handling a 302 redirect | (Reported by Alex Odrov) | |
| File menuselect/menuselect_gtk.c has no license header | (Reported by Jeremy Lainé) | |
| res_pjsip: Wrong Contact and Via fields with multiple UDP interfaces | (Reported by Nikolay shakin) | |
| PJSIP: Adding `sends_registrations = yes` to pjsip_wizard.conf causes crash | (Reported by Jonathan Harris) | |
| app_voicemail: Asterisk unresponsive after changing voicemail password with ODBC | (Reported by Michael) | |
| res_pjsip: Threads pile up needlessly when AOR is blocked | (Reported by Ross Beer) | |
| Allow voicemail boxes to be subscribed to with a presence event package | (Reported by George Joseph) | |
| res_rtp_asterisk: Interaction between smoother and DTMF can cause out of order timestamps | (Reported by Torrey Searle) | |
| ARI: "Error destroying mutex" when listing all ARI applications | (Reported by Stefan Repke) | |
| AST_PBX_MAX_STACK is too low for some applications | (Reported by George Joseph) | |
| Astricon Feedback: Unable to filter ARI events when GETting causes overload of events | (Reported by George Joseph) | |
| switching between native_bridge and simple_bridge can cause one way audio | (Reported by Torrey Searle) | |
| CI: Fix CI so it reverifies commit message changes | (Reported by George Joseph) | |
| database: Add some basic logging | (Reported by Joshua C. Colp) | |
| ari: Originating overwrites channel start time | (Reported by sungtae kim) |
Improvements made in this release:
-----------------------------------
| ari: Added timestamp for some ari events. | (Reported by sungtae kim) | |
| Add logical group at DAHDIChannel event and create "dahdi_group" at CHANNEL function | (Reported by Cirillo Ferreira) | |
| Added creation timestamp for bridge | (Reported by sungtae kim) | |
| Allow wrapuptime to be set for each queue member | (Reported by Rodrigo Ramirez Norambuena) | |
| app_queue: Per-member wrapup time missing from AddQueueMember application | (Reported by Niksa Baldun) | |
| Changed to show all channel stats including wrong media | (Reported by sungtae kim) | |
| 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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