The release of Asterisk 16.16.0 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:
-----------------------------------
| res_pjsip_diversion: Crash if Tel URI contains History-Info | (Reported by Torrey Searle) |
Bugs fixed in this release:
-----------------------------------
| Stasis/messaging: text messages not dispatched to all subscribers when using generic subscription | (Reported by Jean Aunis - Prescom) | |
| chan_sip: SDP: Offers without any enabled stream are accepted. | (Reported by Alexander Traud) | |
| chan_sip: SDP: m=video is parsed even when disabled. | (Reported by Alexander Traud) | |
| chan_sip: Hold/Resume an sRTP call on a video enabled user-agent. | (Reported by Alexander Traud) | |
| chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN instead of a channel variable | (Reported by Ivan Poddubny) | |
| chan_pjsip isn't updating hangupcause on 4XX responses | (Reported by George Joseph) | |
| PJSIP sends duplicate 183 Progress responses | (Reported by Alex Hermann) | |
| chan_pjsip: Subsequent same responses are not stopped | (Reported by Julien) | |
| pjsip: Asterisk goes crazy and massively spams logfile if registration can't be send | (Reported by Michael Maier) | |
| pjsip: SIGSEGV in CLI if no trunk is registered | (Reported by Michael Maier) | |
| LOCK() can grant the same lock to multiple channels spuriously | (Reported by Jaco Kroon) | |
| Crash occurs when Transfer and execute Hangup before the Transfer result | (Reported by Dan Cropp) | |
| Segmentation fault in mixmonitor_ds_destroy | (Reported by Robert Sutton) | |
| tel: URI in Diversion header causes crash | (Reported by Mikhail Ivanov) | |
| Spyee information ist missing in ChanSpyStop AMI Event | (Reported by Hendrik Wedhorn) | |
| null media causing the Asterisk crash | (Reported by sungtae kim) | |
| Debug messages printed by scope trace might be missing newlines | (Reported by Alexander Traud) | |
| pjsip: Route Header in Cancel request incorrectly set | (Reported by Flole Systems) | |
| res_musiconhold: Segfault on realtime music on hold without entries | (Reported by Nathan Bruning) | |
| Crash when manipulating PJSIP invite dlg ref counts | (Reported by Sean Bright) | |
| Media cache URL requests allow infinite redirects | (Reported by Sean Bright) | |
| res_pjsip_stir_shaken: Fix module description | (Reported by Stanislav Abramenkov) | |
| AST_MODULE_INFO no, MODULEINFO depend | (Reported by Alexander Traud) | |
| [patch] chan_sip: TCP/TLS client without server. | (Reported by Alexander Traud) | |
| res_pjsip: malformed header Accept-Encoding in OPTIONS response | (Reported by Alexander Greiner-Baer) | |
| Incorrect setup of recall channels | (Reported by Boris P. Korzun) | |
| app_queue: Deadlock between queues container and individual queues | (Reported by George Joseph) |
Improvements made in this release:
-----------------------------------
| Two repeated 183 | (Reported by Gant Liu) | |
| contrib: systemd asterisk service for centos8 or other newer linux versions | (Reported by Mark Petersen) | |
| res_http_media_cache: HTTP media cache stored hardcoded in /tmp | (Reported by laszlovl) | |
| VoiceMail() should have an option to play greetings as Early Media | (Reported by Juan Carlos Castro y Castro) |
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
Thank you for your continued support of Asterisk!
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