[asterisk-dev] Asterisk 18.2.0-rc1 Now Available

[Asterisk Development Team]

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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