Asterisk 16.2.0-rc1 Now Available


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The Asterisk Development Team would like to announce the first
release candidate of Asterisk 16.2.0.
This release candidate is available for immediate download at 
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.2.0-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:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-28173 - Deadlock in chan_sip handling subscribe
      request during res_parking reload
      (Reported by Giuseppe
      Sucameli)
 * ASTERISK-28104 - AstriCon Feedback:  Automatically create a 1
      line dialplan context for stasis apps
      (Reported by George
      Joseph)
 * ASTERISK-28271 - Opensuse Leap 15 --with-jannson-bundled will
      not compile
      (Reported by David Wilcox)
 * ASTERISK-28238 - PJSIP realtime. getcontext not working with
      DUNDI
      (Reported by Ray)
 * ASTERISK-28263 - codec_opus: errors setting max_playback_rate
      and bitrate to "sdp"
      (Reported by Gianluca Merlo)
 * ASTERISK-28250 - build: Cross-compilation fails for target
      arm-linux-gnueabihf
      (Reported by Jean Aunis - Prescom)
 * ASTERISK-28257 - res_http_websocket: PING / PONG opcodes
      break data reception
      (Reported by Jeremy Lainé)
 * ASTERISK-28252 - HangupHandler manager events are never
      thrown
      (Reported by Gerald Schnabel)
 * ASTERISK-28213 - res_pjsip: Threads pile up needlessly when
      AOR is blocked
      (Reported by Ross Beer)
 * ASTERISK-28249 - res_monitor: Segfault with
      Monitor(wav,file,i)
      (Reported by Valentin Vidić)
 * ASTERISK-28244 - stasis: Filter messages at publishing to
      AMI/ARI
      (Reported by Joshua C. Colp)
 * ASTERISK-28231 - res_http_websocket: Not responding to
      Connection Close Frame (opcode 8)
      (Reported by Jeremy
      Lainé)
 * ASTERISK-28197 - stasis: ast_endpoint struct holds the
      channel_ids of channels past destruction in certain cases
     
      (Reported by Mohit Dhiman)
 * ASTERISK-28232 - core: RAII using clang use-after-scope
      issue
      (Reported by Diederik de Groot)
 * ASTERISK-28230 - res_rtp_asterisk: abs-send-time extension
      added with Asterisk 15.5.0 breaks GXV3140 video telephony
     
      (Reported by David Kuehling)
 * ASTERISK-28162 - [patch] need to reset DTMF last sequence
      number and timestamp on RTP renegotiation
      (Reported by
      Alexei Gradinari)
 * ASTERISK-28225 - app_voicemail: Channel variable
      VM_MESSAGEFILE not updated correctly if message marked "urgent"

      (Reported by boatright)
 * ASTERISK-28218 - app_queue: Asterisk crashes when using Queue
      with a pre-dial handler (option b)
      (Reported by Mark)
 * ASTERISK-28212 - stasis: Statistics broke ABI under developer
      mode
      (Reported by Joshua C. Colp)
 * ASTERISK-28222 - Regression: MWI polling no longer works
    
      (Reported by abelbeck)
 * ASTERISK-28221 - Bug in ast_coredumper
      (Reported by
      Andrew Nagy)
 * ASTERISK-28215 - app_voicemail: Leaving voicemail sometimes
      doesn't trigger NOTIFYs
      (Reported by George Joseph)
 * ASTERISK-27959 - [patch] Asterisk 15.4.1 h264 fmtp
      negotiation problem
      (Reported by David Kuehling)
 * ASTERISK-28201 - [patch] confbridge: no announce to the
      marked users when they join an empty conference
      (Reported
      by Alexei Gradinari)
 * ASTERISK-28117 - stasis: Add statistics for usage when in
      developer mode
      (Reported by Joshua C. Colp)
 * ASTERISK-28186 - stasis: Filter messages at publishing based
      on to_* presence
      (Reported by Joshua C. Colp)
 * ASTERISK-28194 - chan_sip: Leak using contact ACL
     
      (Reported by Giuseppe Sucameli)
 * ASTERISK-27095 - chan_pjsip: When connected_line_method is
      set to invite, we're not trying UPDATE
      (Reported by George
      Joseph)
 * ASTERISK-28182 - chan_pjsip: When connected_line_method is
      set to invite, asterisk is not trying UPDATE
      (Reported by
      nappsoft)
 * ASTERISK-28157 - Asterisk crashes when the res_pjsip_*
      modules unload
      (Reported by sungtae kim)

Improvements made in this release:
-----------------------------------
 * ASTERISK-28246 - Support skipping on the g726 format
     
      (Reported by Eyal Hasson)
 * ASTERISK-28196 - bridge_softmix: Does not support WebRTC
      source with multi video tracks.
      (Reported by Xiemin Chen)
 * ASTERISK-28198 - res_ari: Add new hangup causes for ARI
      Channel DELETE command
      (Reported by Sebastian Damm)

For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.2.0-rc1

Thank you for your continued support of Asterisk!

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