[asterisk-dev] Asterisk 16.16.0 Now Available

[Asterisk Development Team]

The Asterisk Development Team would like to announce the release of Asterisk 16.16.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

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:
-----------------------------------
 * 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-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-29240 - chan_pjsip: Incoming PJSIP calls set global
      SIPDOMAIN instead of a channel variable
      (Reported by Ivan
      Poddubny)
 * 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-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-29209 - Debug messages printed by scope trace might
      be missing newlines
      (Reported by Alexander Traud)
 * ASTERISK-29024 - pjsip: Route Header in Cancel request
      incorrectly set
      (Reported by Flole Systems)
 * 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-28798 - [patch] chan_sip: TCP/TLS client without
      server.
      (Reported by Alexander Traud)
 * ASTERISK-29165 - res_pjsip: malformed header Accept-Encoding
      in OPTIONS response
      (Reported by Alexander Greiner-Baer)
 * 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, 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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