[asterisk-dev] Asterisk 19.2.0-rc1 Now Available

Asterisk Development Team asteriskteam at digium.com
Thu Feb 3 07:34:40 CST 2022


The Asterisk Development Team would like to announce the first
release candidate of Asterisk 19.2.0.
This release candidate is available for immediate download at 
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 19.2.0-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:

New Features made in this release:
-----------------------------------
 * ASTERISK-29808 - cdr: allow disabling CDR by default
     
      (Reported by N A)
 * ASTERISK-29830 - ami: Add AMI event for Wink
      (Reported
      by N A)
 * ASTERISK-29802 - app_sf: Add full tech-agnostic SF support
  
      (Reported by N A)
 * ASTERISK-29759 - app_sendtext: Add ReceiveText application
  
      (Reported by N A)
 * ASTERISK-29706 - func_json: Add JSON parsing function
     
      (Reported by N A)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-29888 - res_pjsip_outbound_authenticator_digest:
      ABRT attempting to clean up auth_sess
      (Reported by George
      Joseph)
 * ASTERISK-29854 - func_frame_drop: fix buffer usage typo
     
      (Reported by N A)
 * ASTERISK-29857 - res_tonedetect: fix logic errors in code
   
      (Reported by N A)
 * ASTERISK-29869 - rtp sequence number can skip after DTMF
      under certain bridges
      (Reported by Torrey Searle)
 * ASTERISK-29817 - gethostbyname_r is misdetected on NetBSD and
      causes a build failure
      (Reported by Micha�� G��rny)
 * ASTERISK-29698 - Segfault if sorcery object_lifetime_maximum
      and qualify_frequency the same value
      (Reported by Alexei
      Gradinari)
 * ASTERISK-29852 - make_version uses GNU-ism that break
      git-svn-id parsing on NetBSD
      (Reported by Micha�� G��rny)
 * ASTERISK-29850 - ast_get_tid() not implemented for NetBSD
   
      (Reported by Micha�� G��rny)
 * ASTERISK-29851 - rdtsc is not enabled (stubbed out) on
      NetBSD
      (Reported by Micha�� G��rny)
 * ASTERISK-29818 - Build failure on NetBSD due to hmac function
      collision
      (Reported by Micha�� G��rny)
 * ASTERISK-29856 - res_rtp_asterisk: Invalid comparison creates
      unreachable code
      (Reported by N A)
 * ASTERISK-29867 - configure fails if libsrtp dev files are not
      installed
      (Reported by Sean Bright)
 * ASTERISK-29813 - res_pjsip_session doesn't support multipart
      message bodies
      (Reported by George Joseph)
 * ASTERISK-29858 - Regression:  Using external pjproject not
      working after "hack" commit
      (Reported by George Joseph)
 * ASTERISK-29859 - VoiceMailMain() fails when encountering
      non-numeric CALLERID(num)
      (Reported by Mark Murawski)
 * ASTERISK-29847 - pbx_variables: ASTSBINDIR is missing
     
      (Reported by N A)
 * ASTERISK-29824 - It's hard to make changes to bundled
      pjproject
      (Reported by George Joseph)
 * ASTERISK-29695 - SAY.CONF wrong logic when converting 24hour
      time to say 12 hour am/pm
      (Reported by Vincent Dubois)
 * ASTERISK-29664 - PJSIP processing token with % incorrectly
  
      (Reported by Dan Cropp)
 * ASTERISK-29827 - Support for Nordic language syntax in
      Queues
      (Reported by Mark Petersen)
 * ASTERISK-29515 - app_queue: QueueSummary and QueueStatus
      events don't exist in documentation
      (Reported by Luke
      Escude)
 * ASTERISK-29746 - tcptls.c: TCP client connect fails due to
      interrupt
      (Reported by Kevin Harwell)
 * ASTERISK-29806 - app_queue: extension state incorrect
     
      (Reported by Steve Davies)
 * ASTERISK-29816 - SAY_DTMF_INTERRUPT channel variable is not
      honored
      (Reported by Sean Bright)
 * ASTERISK-28863 - The ast_rtp_codecs_payloads functions don't
      preserve order
      (Reported by George Joseph)
 * ASTERISK-29320 - res_pjsip_sdp_rtp: Codec preference order of
      remote is not correct on unhold
      (Reported by Ross Beer)
 * ASTERISK-29821 - Deadlock in bridge_channel_internal_join()
      on local channels.
      (Reported by Krzysztof Trempala)
 * ASTERISK-29722 - test_timezone_watch breaks during DST to ST
      transition
      (Reported by Josh Soref)
 * ASTERISK-29804 - bundled_pjproject: sip_inv is missing
      multipart support in some cases
      (Reported by George
      Joseph)
 * ASTERISK-29794 - ast_coredumper does not delete results when
      requested and a specific output dir is set
      (Reported by
      Frederic Van Espen)
 * ASTERISK-29803 - pbx_variables: cp4 variables is used
      uninitialized
      (Reported by N A)
 * ASTERISK-29766 - pbx_variables: MSet truncates sets after 24
      variables
      (Reported by N A)
 * ASTERISK-29772 - chan_sip: ${CHANNEL(ruri)} in Dial/Queue
      b(test,s,1) cause a coredump
      (Reported by Mark Petersen)
 * ASTERISK-29790 - xmldoc: Dump invalid to XML DTD: XSLT
     
      (Reported by Alexander Traud)
 * ASTERISK-29791 - xmldoc: Dump invalid to XML DTD: ACO
      Matchfield
      (Reported by Alexander Traud)
 * ASTERISK-26991 - documentation: Doxygen site is no longer
      being updated
      (Reported by Joshua C. Colp)
 * ASTERISK-20259 - [patch] Update Doxygen Configuration for
      make progdocs
      (Reported by Andrew Latham)
 * ASTERISK-29785 - res_pjsip_sdp_rtp: Warns on every offered
      crypto suite
      (Reported by Alexander Traud)
 * ASTERISK-27406 - Infinite loop when out of ports and rtpstart
      value is odd
      (Reported by Thomas Guebels)
 * ASTERISK-28053 - chan_pjsip: Wrong or missing Q.850 reason in
      CANCEL
      (Reported by Simone Lazzaris)
 * ASTERISK-29761 - res: Fix for Doxygen
      (Reported by
      Alexander Traud)
 * ASTERISK-29763 - main: Fix for Doxygen
      (Reported by
      Alexander Traud)

Improvements made in this release:
-----------------------------------
 * ASTERISK-29832 - Enable pickup on channel after having
      received 183 Progress
      (Reported by Mark Petersen)
 * ASTERISK-29831 - Queue don't play "thank-you" when here is no
      hold time announcements
      (Reported by Mark Petersen)
 * ASTERISK-28890 - res_pjsip_sdp_rtp: Keepalive not supported
      for video streams
      (Reported by Luke Escude)
 * ASTERISK-29855 - frame.h: fix CNG documentation typo
     
      (Reported by N A)
 * ASTERISK-29848 - documentation: Document special system and
      channel variables
      (Reported by N A)
 * ASTERISK-29819 - utils.c: Remove all usages of
      ast_gethostbyname()
      (Reported by Sean Bright)
 * ASTERISK-29815 - dsp: Define magic number as macro
     
      (Reported by N A)
 * ASTERISK-29807 - cli: add module refresh command
     
      (Reported by N A)
 * ASTERISK-29829 - app_mp3: Throw warning if attempting to play
      a nonexistent stream
      (Reported by N A)
 * ASTERISK-24427 - Documentation is missing for a few AMI
      Events - Including CDR and events triggered after the
      QueueStatus action
      (Reported by Dafi Ni)
 * ASTERISK-29795 - DIALEDPEERNUMBER not set on destination
      channel for Queue calls
      (Reported by Mark Petersen)
 * ASTERISK-29801 - app.c: Throw warnings for nonexistent
      options
      (Reported by N A)
 * ASTERISK-29797 - Support for Danish language syntax in VM
   
      (Reported by Mark Petersen)
 * ASTERISK-29800 - strings: Fix misusage in comment examples
  
      (Reported by N A)
 * ASTERISK-29758 - configs: Minor updates to sample configs
   
      (Reported by N A)
 * ASTERISK-29745 - pbx: Add public API for more elegant
      variable substitution with extensions
      (Reported by N A)
 * ASTERISK-29729 - Incompatibility with newer spandsp releases
      (3.0.0+)
      (Reported by Dustin Marquess)

For a full list of changes in this release candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0-rc1

Thank you for your continued support of Asterisk!
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20220203/a05ef552/attachment.html>


More information about the asterisk-dev mailing list