[asterisk-dev] Asterisk 18.2.0 Now Available

Sun Jan 24 02:17:54 CST 2021

What next?

пт, 22 янв. 2021 г. в 3:25, Asterisk Development Team <
asteriskteam at 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!*
>
>
> --
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