What next? пт, 22 янв. 2021 г. в 3:25, Asterisk Development Team < asteriskt...@digium.com>:
> The Asterisk Development Team would like to announce the release of > Asterisk 18.2.0. > This release is available for immediate download at > https://downloads.asterisk.org/pub/telephony/asterisk > > The release of Asterisk 18.2.0 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-29219 > <https://issues.asterisk.org/jira/browse/ASTERISK-29219>] - > > res_pjsip_diversion: Crash if Tel URI contains History-Info > (Reported by Torrey Searle) > > *Bugs fixed in this release:* > ----------------------------------- > > - [ASTERISK-29229 > <https://issues.asterisk.org/jira/browse/ASTERISK-29229>] - > > Stasis/messaging: text messages not dispatched to all subscribers when > using generic subscription > (Reported by Jean Aunis - Prescom) > > - [ASTERISK-29240 > <https://issues.asterisk.org/jira/browse/ASTERISK-29240>] - > > chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN instead of a channel > variable > (Reported by Ivan Poddubny) > > - [ASTERISK-29238 > <https://issues.asterisk.org/jira/browse/ASTERISK-29238>] - > > chan_sip: SDP: Offers without any enabled stream are accepted. > (Reported by Alexander Traud) > > - [ASTERISK-29237 > <https://issues.asterisk.org/jira/browse/ASTERISK-29237>] - > > chan_sip: SDP: m=video is parsed even when disabled. > (Reported by Alexander Traud) > > - [ASTERISK-29222 > <https://issues.asterisk.org/jira/browse/ASTERISK-29222>] - > > chan_sip: Hold/Resume an sRTP call on a video enabled user-agent. > (Reported by Alexander Traud) > > - [ASTERISK-27902 > <https://issues.asterisk.org/jira/browse/ASTERISK-27902>] - > > chan_pjsip isn't updating hangupcause on 4XX responses > (Reported by George Joseph) > > - [ASTERISK-28016 > <https://issues.asterisk.org/jira/browse/ASTERISK-28016>] - > > PJSIP sends duplicate 183 Progress responses > (Reported by Alex Hermann) > > - [ASTERISK-28185 > <https://issues.asterisk.org/jira/browse/ASTERISK-28185>] - > > chan_pjsip: Subsequent same responses are not stopped > (Reported by Julien) > > - [ASTERISK-29230 > <https://issues.asterisk.org/jira/browse/ASTERISK-29230>] - > > pjsip: Asterisk goes crazy and massively spams logfile if registration > can't be send > (Reported by Michael Maier) > > - [ASTERISK-29231 > <https://issues.asterisk.org/jira/browse/ASTERISK-29231>] - > > pjsip: SIGSEGV in CLI if no trunk is registered > (Reported by Michael Maier) > > - [ASTERISK-29217 > <https://issues.asterisk.org/jira/browse/ASTERISK-29217>] - > > LOCK() can grant the same lock to multiple channels spuriously > (Reported by Jaco Kroon) > > - [ASTERISK-29201 > <https://issues.asterisk.org/jira/browse/ASTERISK-29201>] - > > Crash occurs when Transfer and execute Hangup before the Transfer result > (Reported by Dan Cropp) > > - [ASTERISK-28947 > <https://issues.asterisk.org/jira/browse/ASTERISK-28947>] - > > Segmentation fault in mixmonitor_ds_destroy > (Reported by Robert Sutton) > > - [ASTERISK-29168 > <https://issues.asterisk.org/jira/browse/ASTERISK-29168>] - > > Asterisk crashes during call transfer > (Reported by Dalius Mockevicius) > > - [ASTERISK-29210 > <https://issues.asterisk.org/jira/browse/ASTERISK-29210>] - > > res_pjsip: Crash when examining transport > (Reported by N GM ) > > - [ASTERISK-29191 > <https://issues.asterisk.org/jira/browse/ASTERISK-29191>] - > > tel: URI in Diversion header causes crash > (Reported by Mikhail Ivanov) > > - [ASTERISK-28883 > <https://issues.asterisk.org/jira/browse/ASTERISK-28883>] - > > Spyee information ist missing in ChanSpyStop AMI Event > (Reported by Hendrik Wedhorn) > > - [ASTERISK-29188 > <https://issues.asterisk.org/jira/browse/ASTERISK-29188>] - > > null media causing the Asterisk crash > (Reported by sungtae kim) > > - [ASTERISK-29024 > <https://issues.asterisk.org/jira/browse/ASTERISK-29024>] - > > pjsip: Route Header in Cancel request incorrectly set > (Reported by Flole Systems) > > - [ASTERISK-29209 > <https://issues.asterisk.org/jira/browse/ASTERISK-29209>] - > > Debug messages printed by scope trace might be missing newlines > (Reported by Alexander Traud) > > - [ASTERISK-29211 > <https://issues.asterisk.org/jira/browse/ASTERISK-29211>] - > > res_musiconhold: Segfault on realtime music on hold without entries > (Reported by Nathan Bruning) > > - [ASTERISK-29022 > <https://issues.asterisk.org/jira/browse/ASTERISK-29022>] - > > Crash when manipulating PJSIP invite dlg ref counts > (Reported by Sean Bright) > > - [ASTERISK-29173 > <https://issues.asterisk.org/jira/browse/ASTERISK-29173>] - > > Media cache URL requests allow infinite redirects > (Reported by Sean Bright) > > - [ASTERISK-29175 > <https://issues.asterisk.org/jira/browse/ASTERISK-29175>] - > > res_pjsip_stir_shaken: Fix module description > (Reported by Stanislav Abramenkov) > > - [ASTERISK-29148 > <https://issues.asterisk.org/jira/browse/ASTERISK-29148>] - > > AST_MODULE_INFO no, MODULEINFO depend > (Reported by Alexander Traud) > > - [ASTERISK-29165 > <https://issues.asterisk.org/jira/browse/ASTERISK-29165>] - > > res_pjsip: malformed header Accept-Encoding in OPTIONS response > (Reported by Alexander Greiner-Baer) > > - [ASTERISK-28798 > <https://issues.asterisk.org/jira/browse/ASTERISK-28798>] - > > [patch] chan_sip: TCP/TLS client without server. > (Reported by Alexander Traud) > > - [ASTERISK-29161 > <https://issues.asterisk.org/jira/browse/ASTERISK-29161>] - > > Incorrect setup of recall channels > (Reported by Boris P. Korzun) > > - [ASTERISK-29155 > <https://issues.asterisk.org/jira/browse/ASTERISK-29155>] - > > app_queue: Deadlock between queues container and individual queues > (Reported by George Joseph) > > *Improvements made in this release:* > ----------------------------------- > > - [ASTERISK-28549 > <https://issues.asterisk.org/jira/browse/ASTERISK-28549>] - > > Two repeated 183 > (Reported by Gant Liu) > > - [ASTERISK-29216 > <https://issues.asterisk.org/jira/browse/ASTERISK-29216>] - > > contrib: systemd asterisk service for centos8 or other newer linux versions > (Reported by Mark Petersen) > > - [ASTERISK-29143 > <https://issues.asterisk.org/jira/browse/ASTERISK-29143>] - > > res_http_media_cache: HTTP media cache stored hardcoded in /tmp > (Reported by laszlovl) > > - [ASTERISK-29118 > <https://issues.asterisk.org/jira/browse/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, please see the ChangeLog: > https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0 > > *Thank you for your continued support of Asterisk!* > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev
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