[asterisk-dev] Asterisk 13.25.0 Now Available

Asterisk Development Team asteriskteam at digium.com
Fri Feb 15 15:17:39 CST 2019


The Asterisk Development Team would like to announce the release of Asterisk 13.25.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.25.0 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-28288 - Resources (udptl fd) leaking for T.38 calls

      (Reported by Paulo Vicentini)
 * ASTERISK-28213 - res_pjsip: Threads pile up needlessly when
      AOR is blocked
      (Reported by Ross Beer)
 * ASTERISK-28271 - Opensuse Leap 15 --with-jannson-bundled will
      not compile
      (Reported by David Wilcox)
 * ASTERISK-28104 - AstriCon Feedback:  Automatically create a 1
      line dialplan context for stasis apps
      (Reported by George
      Joseph)
 * ASTERISK-28238 - PJSIP realtime. getcontext not working with
      DUNDI
      (Reported by Ray)
 * ASTERISK-28173 - Deadlock in chan_sip handling subscribe
      request during res_parking reload
      (Reported by Giuseppe
      Sucameli)
 * 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-28156 - Race condition involving session->media
      (res_pjsip_session) leads to crash.
      (Reported by Paulo
      Vicentini)
 * 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-28231 - res_http_websocket: Not responding to
      Connection Close Frame (opcode 8)
      (Reported by Jeremy
      Lain��)
 * 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-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-28225 - app_voicemail: Channel variable
      VM_MESSAGEFILE not updated correctly if message marked "urgent"

      (Reported by boatright)
 * 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-28162 - [patch] need to reset DTMF last sequence
      number and timestamp on RTP renegotiation
      (Reported by
      Alexei Gradinari)
 * 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-28117 - stasis: Add statistics for usage when in
      developer mode
      (Reported by Joshua C. Colp)
 * ASTERISK-28201 - [patch] confbridge: no announce to the
      marked users when they join an empty conference
      (Reported
      by Alexei Gradinari)
 * ASTERISK-28194 - chan_sip: Leak using contact ACL
     
      (Reported by Giuseppe Sucameli)
 * ASTERISK-28186 - stasis: Filter messages at publishing based
      on to_* presence
      (Reported by Joshua C. Colp)
 * 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)

Improvements made in this release:
-----------------------------------
 * ASTERISK-28246 - Support skipping on the g726 format
     
      (Reported by Eyal Hasson)

For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.25.0

Thank you for your continued support of Asterisk!
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