[asterisk-announce] Asterisk 19.2.0 Now Available

Thu Feb 10 07:00:00 CST 2022

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

The release of Asterisk 19.2.0 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:

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, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0

Thank you for your continued support of Asterisk!
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