asterisk announce - Feb 2019 - Asterisk 16.2.0 Now Available

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

The release of Asterisk 16.2.0 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:

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-28213 - res_pjsip: Threads pile up needlessly when
      AOR is blocked
      (Reported by Ross Beer)
 * 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 Lain��)
 * ASTERISK-28252 - HangupHandler manager events are never
      thrown
      (Reported by Gerald Schnabel)
 * ASTERISK-28249 - res_monitor: Segfault with
      Monitor(wav,file,i)
      (Reported by Valentin Vidi��)
 * 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
      Lain��)
 * 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, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.2.0

Thank you for your continued support of Asterisk!
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