The Asterisk Development Team would like to announce the first release candidate of Asterisk 18.2.0. This release candidate is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 18.2.0-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-29219 - res_pjsip_diversion: Crash if Tel URI contains History-Info (Reported by Torrey Searle) Bugs fixed in this release: ----------------------------------- * ASTERISK-29229 - Stasis/messaging: text messages not dispatched to all subscribers when using generic subscription (Reported by Jean Aunis - Prescom) * ASTERISK-29240 - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN instead of a channel variable (Reported by Ivan Poddubny) * ASTERISK-29238 - chan_sip: SDP: Offers without any enabled stream are accepted. (Reported by Alexander Traud) * ASTERISK-29237 - chan_sip: SDP: m=video is parsed even when disabled. (Reported by Alexander Traud) * ASTERISK-29222 - chan_sip: Hold/Resume an sRTP call on a video enabled user-agent. (Reported by Alexander Traud) * ASTERISK-27902 - chan_pjsip isn't updating hangupcause on 4XX responses (Reported by George Joseph) * ASTERISK-28016 - PJSIP sends duplicate 183 Progress responses (Reported by Alex Hermann) * ASTERISK-28185 - chan_pjsip: Subsequent same responses are not stopped (Reported by Julien) * ASTERISK-29230 - pjsip: Asterisk goes crazy and massively spams logfile if registration can't be send (Reported by Michael Maier) * ASTERISK-29231 - pjsip: SIGSEGV in CLI if no trunk is registered (Reported by Michael Maier) * ASTERISK-29217 - LOCK() can grant the same lock to multiple channels spuriously (Reported by Jaco Kroon) * ASTERISK-29201 - Crash occurs when Transfer and execute Hangup before the Transfer result (Reported by Dan Cropp) * ASTERISK-28947 - Segmentation fault in mixmonitor_ds_destroy (Reported by Robert Sutton) * ASTERISK-29168 - Asterisk crashes during call transfer (Reported by Dalius Mockevicius) * ASTERISK-29210 - res_pjsip: Crash when examining transport (Reported by N GM ) * ASTERISK-29191 - tel: URI in Diversion header causes crash (Reported by Mikhail Ivanov) * ASTERISK-28883 - Spyee information ist missing in ChanSpyStop AMI Event (Reported by Hendrik Wedhorn) * ASTERISK-29188 - null media causing the Asterisk crash (Reported by sungtae kim) * ASTERISK-29024 - pjsip: Route Header in Cancel request incorrectly set (Reported by Flole Systems) * ASTERISK-29209 - Debug messages printed by scope trace might be missing newlines (Reported by Alexander Traud) * ASTERISK-29211 - res_musiconhold: Segfault on realtime music on hold without entries (Reported by Nathan Bruning) * ASTERISK-29022 - Crash when manipulating PJSIP invite dlg ref counts (Reported by Sean Bright) * ASTERISK-29173 - Media cache URL requests allow infinite redirects (Reported by Sean Bright) * ASTERISK-29175 - res_pjsip_stir_shaken: Fix module description (Reported by Stanislav Abramenkov) * ASTERISK-29148 - AST_MODULE_INFO no, MODULEINFO depend (Reported by Alexander Traud) * ASTERISK-29165 - res_pjsip: malformed header Accept-Encoding in OPTIONS response (Reported by Alexander Greiner-Baer) * ASTERISK-28798 - [patch] chan_sip: TCP/TLS client without server. (Reported by Alexander Traud) * ASTERISK-29161 - Incorrect setup of recall channels (Reported by Boris P. Korzun) * ASTERISK-29155 - app_queue: Deadlock between queues container and individual queues (Reported by George Joseph) Improvements made in this release: ----------------------------------- * ASTERISK-28549 - Two repeated 183 (Reported by Gant Liu) * ASTERISK-29216 - contrib: systemd asterisk service for centos8 or other newer linux versions (Reported by Mark Petersen) * ASTERISK-29143 - res_http_media_cache: HTTP media cache stored hardcoded in /tmp (Reported by laszlovl) * ASTERISK-29118 - 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 candidate, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0-rc1 Thank you for your continued support of Asterisk!
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