[asterisk-announce] Asterisk 20.0.0 Now Available

Asterisk Development Team asteriskteam at digium.com
Wed Oct 19 11:26:02 CDT 2022


The Asterisk Development Team would like to announce the release of Asterisk 20.0.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 20.0.0 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:

Deprecations made in this release:
-----------------------------------
 * ASTERISK-29601 - moduleinfo: Add replacement module
      information
      (Reported by N A)
 * ASTERISK-29602 - res_monitor: Disable building by default.
  
      (Reported by Joshua C. Colp)
 * ASTERISK-29600 - muted: Remove deprecated application
     
      (Reported by Joshua C. Colp)
 * ASTERISK-29599 - conf2ael: Remove deprecated application
    
      (Reported by Joshua C. Colp)
 * ASTERISK-29598 - res_config_sqlite: Remove deprecated module

      (Reported by Joshua C. Colp)
 * ASTERISK-29597 - chan_vpb: Remove deprecated module
     
      (Reported by Joshua C. Colp)
 * ASTERISK-29596 - chan_misdn: Remove deprecated module
     
      (Reported by Joshua C. Colp)
 * ASTERISK-29595 - chan_nbs: Remove deprecated module
     
      (Reported by Joshua C. Colp)
 * ASTERISK-29594 - chan_phone: Remove deprecated module
     
      (Reported by Joshua C. Colp)
 * ASTERISK-29593 - chan_oss: Remove deprecated module
     
      (Reported by Joshua C. Colp)
 * ASTERISK-29592 - cdr_syslog: Remove deprecated module
     
      (Reported by Joshua C. Colp)
 * ASTERISK-29591 - app_dahdiras: Remove deprecated module
     
      (Reported by Joshua C. Colp)
 * ASTERISK-29590 - app_nbscat: Remove deprecated module
     
      (Reported by Joshua C. Colp)
 * ASTERISK-29589 - app_image: Remove deprecated module
     
      (Reported by Joshua C. Colp)
 * ASTERISK-29588 - app_url: Remove deprecated module
     
      (Reported by Joshua C. Colp)
 * ASTERISK-29587 - app_fax: Remove deprecated module
     
      (Reported by Joshua C. Colp)
 * ASTERISK-29586 - app_ices: Remove deprecated module
     
      (Reported by Joshua C. Colp)
 * ASTERISK-29585 - app_mysql: Remove deprecated module
     
      (Reported by Joshua C. Colp)
 * ASTERISK-29584 - cdr_mysql: Remove deprecated module
     
      (Reported by Joshua C. Colp)
 * ASTERISK-29548 - app_meetme: Deprecated in 19, to be removed
      in 21
      (Reported by Joshua C. Colp)
 * ASTERISK-29549 - app_osploop: Deprecated in 19, to be removed
      in 21
      (Reported by Joshua C. Colp)
 * ASTERISK-29550 - chan_alsa: Deprecated in 19, to be removed
      in 21
      (Reported by Joshua C. Colp)
 * ASTERISK-29551 - chan_mgcp: Deprecated in 19, to be removed
      in 21
      (Reported by Joshua C. Colp)
 * ASTERISK-29552 - chan_skinny: Deprecated in 19, to be removed
      in 21
      (Reported by Joshua C. Colp)
 * ASTERISK-29553 - res_pktccops: Deprecated in 19, to be
      removed in 21
      (Reported by Joshua C. Colp)
 * ASTERISK-29558 - app_macro: Deprecated in 16, to be removed
      in 21
      (Reported by Joshua C. Colp)
 * ASTERISK-29567 - chan_sip: Deprecated in 17, to be removed in
      21
      (Reported by Joshua C. Colp)
 * ASTERISK-29572 - res_monitor: Deprecated in 16, to be removed
      in 21
      (Reported by Joshua C. Colp)

Security bugs fixed in this release:
-----------------------------------
 * ASTERISK-29476 - res_stir_shaken: Blind SSRF vulnerabilities

      (Reported by Clint Ruoho)
 * ASTERISK-29872 - res_stir_shaken: Resource exhaustion with
      large files
      (Reported by Benjamin Keith Ford)
 * ASTERISK-29838 - ${SQL_ESC()} not correctly escaping a
      terminating \
      (Reported by Leandro Dardini)
 * ASTERISK-29415 - Crash in PJSIP TLS transport 
     
      (Reported by Andrew Yager)
 * ASTERISK-29381 - chan_pjsip: Remote denial of service by an
      authenticated user
      (Reported by Ivan Poddubny)

New Features made in this release:
-----------------------------------
 * ASTERISK-30037 - Add test support to calling external
      processes
      (Reported by Philip Prindeville)
 * ASTERISK-30161 - locks: add AMI event for deadlock
     
      (Reported by N A)
 * ASTERISK-30211 - app_confbridge: Add end_marked_any option
  
      (Reported by N A)
 * ASTERISK-30186 - res_pjsip: Add support for reloading TLS
      certificate and key information
      (Reported by Joshua C.
      Colp)
 * ASTERISK-29899 - features: Add advanced transfer initiation
      options
      (Reported by N A)
 * ASTERISK-30136 - db: Add AMI action to retrieve all keys
      beginning with a prefix
      (Reported by N A)
 * ASTERISK-30000 - chan_dahdi: Add POLARITY function
     
      (Reported by N A)
 * ASTERISK-30062 - cli: Add CLI command to execute a dialplan
      app
      (Reported by N A)
 * ASTERISK-29999 - pjsip: Get information from 200 OK INVITE
      reply headers
      (Reported by Jos�� Lopes)
 * ASTERISK-30061 - pbx: Add pbx helper application
     
      (Reported by N A)
 * ASTERISK-30063 - app_voicemail: Add option to prevent
      deletion of messages
      (Reported by N A)
 * ASTERISK-30087 - res_parking: Add music on hold override
      option
      (Reported by N A)
 * ASTERISK-29965 - res_pjsip_outbound_registration: Make max
      registration delay configurable
      (Reported by N A)
 * ASTERISK-30036 - app_confbridge: Add CONFBRIDGE_CHANNELS
      function
      (Reported by N A)
 * ASTERISK-29931 - Option to allow a user to not hear the join
      sound on enter but everyone else can
      (Reported by Michael
      Cargile)
 * ASTERISK-29968 - func_db: Add a function to return
      cardinality of keys at prefix
      (Reported by N A)
 * ASTERISK-29486 - Hint-like extension value lookup function
      without device state
      (Reported by N A)
 * ASTERISK-29941 - chan_pjsip: Add ability to send flash
      events
      (Reported by N A)
 * ASTERISK-29820 - cli: Add command to evaluate a function
    
      (Reported by N A)
 * ASTERISK-29876 - app_queue: Add music on hold option
     
      (Reported by N A)
 * ASTERISK-29840 - func_channel: Add LASTCONTEXT and LASTEXTEN
      fields
      (Reported by N A)
 * ASTERISK-29853 - ami: Allow events to be globally disabled
  
      (Reported by N A)
 * 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)
 * ASTERISK-29720 - res_tonedetect: Add call progress tone
      detection
      (Reported by N A)
 * ASTERISK-18069 - [patch] app_queue Add Login Time and Last
      Paused Times to Queue Members
      (Reported by Jamuel Starkey)
 * ASTERISK-29656 - Add CHANNEL_EXISTS function
      (Reported
      by N A)
 * ASTERISK-29496 - Add SendMF application
      (Reported by N
      A)
 * ASTERISK-29627 - Add STRBETWEEN function
      (Reported by N
      A)
 * ASTERISK-29628 - Add file and directory functions
     
      (Reported by N A)
 * ASTERISK-29531 - Add SAYFILES function
      (Reported by N
      A)
 * ASTERISK-29546 - Add tone detection module
      (Reported by
      N A)
 * ASTERISK-18454 - Option for Read to be able to accept #
     
      (Reported by Sta Retji)
 * ASTERISK-29542 - Add audio scrambler
      (Reported by N A)
 * ASTERISK-29478 - Function to drop frames in the TX or RX
      directions
      (Reported by N A)
 * ASTERISK-29389 - Add PJSIP_HEADERS() and ability to read
      header by pattern
      (Reported by Igor Goncharovsky)
 * ASTERISK-29477 - Function to asynchronously store digits
      dialed
      (Reported by N A)
 * ASTERISK-11 - AGI channel_status failure
      (Reported by
      bbawkon)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-30235 - res_crypto and tests:  Memory issues and and
      uninitialized variable error
      (Reported by George Joseph)
 * ASTERISK-30234 - res_geolocation: ...may be used
      uninitialized error in geoloc_config.c
      (Reported by George
      Joseph)
 * ASTERISK-30215 - Inbound SIP INVITE with Geo Location causing
      a Segmentation Fault
      (Reported by Dan Cropp)
 * ASTERISK-30135 - [res_musiconhold] Allows the moh only for
      the answered call
      (Reported by sungtae kim)
 * ASTERISK-26894 - pjsip should support tel uri scheme
     
      (Reported by Gergely D��ms��di)
 * ASTERISK-30210 - func_frame_trace: Channel masquerade
      triggers assertion
      (Reported by N A)
 * ASTERISK-30190 - res_geolocation:  GEOLOC_PROFILE isn't
      returning correct values on incoming channel
      (Reported by
      George Joseph)
 * ASTERISK-29185 - chan_pjsip: Endpoint: allow = all is
      broken.
      (Reported by Alexander Traud)
 * ASTERISK-30192 - res_tonedetect: fix typo for frametype
     
      (Reported by N A)
 * ASTERISK-29453 - alembic: incoming_call_offer_pref and
      outgoing_call_offer_pref missing in "ps_endpoints" table
     
      (Reported by Daniel Th��men)
 * ASTERISK-26826 - testsuite: Add support for Python 3
     
      (Reported by Joshua C. Colp)
 * ASTERISK-30167 - res_geolocation:  Refactor for issues found
      by users
      (Reported by George Joseph)
 * ASTERISK-28422 - Memory Leak in Confbridge menu
     
      (Reported by Ted G)
 * ASTERISK-29917 - ami: FilterList action doesn't exist
     
      (Reported by N A)
 * ASTERISK-30020 - ConfbridgeListRooms Event Not Documented
   
      (Reported by Michael Cargile)
 * ASTERISK-30018 - app_meetme: MeetmeList AMI event not
      documented
      (Reported by Michael Cargile)
 * ASTERISK-30151 - Documentation doesn't include info about
      "field", a 3rd required parameter.
      (Reported by Chris
      Young)
 * ASTERISK-29966 - pbx_variables: ast_str_strlen can be wrong
 
      (Reported by N A)
 * ASTERISK-29905 - OSX: bininstall launchd issue on
      cross-platfrom build
      (Reported by Sergey V. Lobanov)
 * ASTERISK-30137 - manager: Global disabled event filtered is
      incomplete
      (Reported by N A)
 * ASTERISK-30109 - res_pjsip: no contact-status AMI event on
      register of prune-on-boot contact that uses the same URI as
      before Asterisk restart
      (Reported by Michael Neuhauser)
 * ASTERISK-29991 - chan_dahdi, callerid: Caller ID does not
      honor presentation
      (Reported by N A)
 * ASTERISK-30126 - Spelling mistake in
      configs/samples/queues.conf.sample
      (Reported by Sam Banks)
 * ASTERISK-30029 - build: Git security vulnerability fix is sad
      with our accessing git as root during "make install"
     
      (Reported by Joshua C. Colp)
 * ASTERISK-29907 - res_pjsip, app_confbridge: Video call
      through ConfBridge with normal endpoints causes infinite
      loop/crash
      (Reported by N A)
 * ASTERISK-30138 - Compile failure in
      res_geolocation/geoloc_eprofile.c when optimization is enabled
 
      (Reported by George Joseph)
 * ASTERISK-30096 -  cel_odbc: Column type 9 (field
      'cdr:cel:eventtime') is unsupported at this time
      (Reported
      by Morvai Szabolcs)
 * ASTERISK-30083 - chan_iax2: Optional dependency on
      openssl/res_crypto is now mandatory
      (Reported by Dmitry
      Melekhov)
 * ASTERISK-30099 - test_aeap_transport: transport_connect_fail
      sporadically causes failure
      (Reported by Kevin Harwell)
 * ASTERISK-30123 - features: Update automixmon documentation to
      reflect reality
      (Reported by Trevor Peirce)
 * ASTERISK-30117 - pbx_lua: Remove compiler warnings
     
      (Reported by Boris P. Korzun)
 * ASTERISK-30101 - res_prometheus: Optional load
      res_pjsip_outbound_registration.so
      (Reported by Boris P.
      Korzun)
 * ASTERISK-29989 - app_dial, chan_dahdi: DIALSTATUS is
      inconsistent for busy
      (Reported by N A)
 * ASTERISK-30001 - db: Removing nonexistent entries shows
      "Database entry removed"
      (Reported by N A)
 * ASTERISK-29822 - cli: Typing \? freezes the CLI permanently
      with remote console
      (Reported by N A)
 * ASTERISK-30115 - app_dial: Allow hook flashes to propogate on
      outbound dials
      (Reported by N A)
 * ASTERISK-30106 - res_calendar_icalendar: Microsoft online ICS
      calendars no longer work
      (Reported by N A)
 * ASTERISK-30075 - say: Abort if channel hangs up during
      playback
      (Reported by N A)
 * ASTERISK-30072 - res_pjsip: allow TLS verification of
      wildcard cert-bearing servers
      (Reported by Kevin Harwell)
 * ASTERISK-30097 - console: Recent documentation changes for
      connecting to remote console are inconsistent
      (Reported by
      Matthias Hensler)
 * ASTERISK-30043 - Wrong party is disconnected when
      hook-flashing on 3-way bridge
      (Reported by Josh Alberts)
 * ASTERISK-29603 - res_pjsip: UPDATE/re-INVITE not sent when
      "timers=always" is specified in pjsip.conf
      (Reported by
      Ray Crumrine)
 * ASTERISK-30092 - DateTime application: wrong inflection for
      one o'clock in German
      (Reported by Christof Efkemann)
 * ASTERISK-30064 - pbx: iax2 switch causes crash due to
      deadlock and assertion
      (Reported by N A)
 * ASTERISK-30039 - cli: Targeted debug on startup deadlocks and
      creates unstable system
      (Reported by N A)
 * ASTERISK-29981 - res_calendar: Asterisk crashes when
      starting, and will not run
      (Reported by N A)
 * ASTERISK-30051 - res_pjsip: No video after un-hold with
      moh_passthrough=yes
      (Reported by Maximilian Fridrich)
 * ASTERISK-24601 - [patch]Missing RFC4235 tags and attributes
      in PJSIP NOTIFY event: dialog  XML body
      (Reported by Marco
      Paland)
 * ASTERISK-30059 - menuselect: libxml include fails under
      Gentoo
      (Reported by waltermoeller)
 * ASTERISK-30060 - loader: format warnings in dev mode
     
      (Reported by N A)
 * ASTERISK-30065 - pjsip: Open Websocket connection is not
      reused for outgoing requests
      (Reported by LA)
 * ASTERISK-30042 - res_pjsip_transport_websocket: Registration
      over websocket returns a rewritten contact
      (Reported by
      Thomas Guebels)
 * ASTERISK-29993 - chan_dahdi: Operator control option borks
      both lines involved on callee disconnect
      (Reported by N A)
 * ASTERISK-30044 - GCC 12 issues
      (Reported by George
      Joseph)
 * ASTERISK-29655 - res_pjsip_session: No video to caller if no
      camera available
      (Reported by Michael Auracher)
 * ASTERISK-29638 - res_pjsip_session: No video after early
      media
      (Reported by Michael Auracher)
 * ASTERISK-28518 - chan_dahdi: Caller ID FSK Erroneously Sent
      when Picking Up Dahdi Call On Hold
      (Reported by Josh
      Alberts)
 * ASTERISK-29990 - chan_dahdi: adding ring cadences is not
      idempotent on dahdi restart
      (Reported by N A)
 * ASTERISK-30007 - chan_iax2: Prevent crashes due to attempted
      encryption with missing secrets
      (Reported by N A)
 * ASTERISK-29728 - menuselect: Disabled by default modules that
      are enabled are always recompiled
      (Reported by N A)
 * ASTERISK-30002 - app_meetme: Don't erroneously set global
      variables when channel is NULL
      (Reported by N A)
 * ASTERISK-22246 - Asterisk's "T" flag is ignored when used
      with "r" or "R" flags. (documentation bug)
      (Reported by
      Rusty Newton)
 * ASTERISK-26582 - Asterisk seems to ignore the "n" parameter
      for "disable console colorization"
      (Reported by Sebastian
      Gutierrez)
 * ASTERISK-29994 - chan_dahdi: Round robin array size is too
      small for max number of groups
      (Reported by N A)
 * ASTERISK-29843 - Session timers get removed on UPDATE
     
      (Reported by Mark Petersen)
 * ASTERISK-29943 - file.c: seeking to negative file offset is
      not prevented
      (Reported by N A)
 * ASTERISK-29842 - Do not change 180 Ringing to 183 Progress
      even if early_media already enabled
      (Reported by Mark
      Petersen)
 * ASTERISK-29955 - chan_sip: SIP route header is missing on
      UPDATE
      (Reported by Mark Petersen)
 * ASTERISK-29948 - iostream: Infinite TCP timeout writing data

      (Reported by N A)
 * ASTERISK-29253 - Incorrect bridging on transfer
     
      (Reported by Yury Kirsanov)
 * ASTERISK-30024 - Failed to sign STIR/SHAKEN payload with
      functionality not enabled
      (Reported by Claude Diderich)
 * ASTERISK-30006 - res_pjsip: UDP transport does not work when
      async_operations is greater than 1
      (Reported by Ross Beer)
 * ASTERISK-30021 - ast_variable_list_replace_variable uses
      variable with new keyword
      (Reported by Jasper
      Hafkenscheid)
 * ASTERISK-30023 - cdr_adaptive_odbc: does not support DATETIME
      database columns
      (Reported by Gregory Massel)
 * ASTERISK-30015 - pjsip / WebRTC: Chrome creating large number
      of SDP attributes
      (Reported by Josh Hogan)
 * ASTERISK-29411 - Crash in pjsip_msg_find_hdr_by_name
     
      (Reported by LA)
 * ASTERISK-29535 - Segmentation fault in libasteriskpj.so.2
   
      (Reported by Daniel Bonazzi)
 * ASTERISK-26719 - pbx: Only up to 127 includes in a dialplan
      context (AST_PBX_MAX_STACK - 1)
      (Reported by Tzafrir
      Cohen)
 * ASTERISK-29988 - REGRESSION: The build process is requiring
      xmllint or xmlstarlet ro be installed when it shouldn't
     
      (Reported by George Joseph)
 * ASTERISK-29986 - build: Asterisk 18.11.0 doesn't compile when
      wget isn't available
      (Reported by Stefan Ruijsenaars)
 * ASTERISK-29895 - chan_iax2: Fix misaligned spacing in iax2
      show netstats printout
      (Reported by N A)
 * ASTERISK-29048 - chan_iax2: "iax2 show registry" shows host
      for perceived
      (Reported by David Herselman)
 * ASTERISK-26689 - res_pjsip_sdp_rtp: 183 Session in Progress.
      Disconnecting channel for lack of RTP activity
      (Reported
      by Dmitriy Serov)
 * ASTERISK-29929 - res_pjsip_sdp_rtp: Disconnecting channel for
      lack of RTP activity in one way sessions
      (Reported by
      Boris P. Korzun)
 * ASTERISK-29674 - Adjust for 64bit time_t
      (Reported by
      Andre Heider)
 * ASTERISK-29961 - RLS: domain part of 'uri' list attribute
      mismatch with SUBSCRIBE request
      (Reported by Alexei
      Gradinari)
 * ASTERISK-29960 - ari: Retrieving stored recording can returns
      wrong file
      (Reported by Arix)
 * ASTERISK-29950 - SayNumber can handle '01' to '07', but not
      '08' or '09'
      (Reported by Jim Van Meggelen)
 * ASTERISK-29928 - logging messages truncated when using MUSL
      runtime
      (Reported by Philip Prindeville)
 * ASTERISK-29939 - agi: Fix xmldoc bug with set music
     
      (Reported by N A)
 * ASTERISK-28891 - documentation: AGICommand_set+music
      documentation arguments displayed incorreclty
      (Reported by
      Jonathan Harris)
 * ASTERISK-29924 - res_config_pgsql: omit "unsupported column
      type 'text'" error
      (Reported by Boris P. Korzun)
 * ASTERISK-29923 - docs, LICENSE: pbx.digium.com no longer
      exists
      (Reported by N A)
 * ASTERISK-29904 - RLS: Batched Notifications stop working
    
      (Reported by Alexei Gradinari)
 * ASTERISK-29365 - taskprocessor: Can cause assert at shutdown

      (Reported by Joshua C. Colp)
 * ASTERISK-29873 - [patch] Queue Realtime load
      (Reported
      by Alexei Gradinari)
 * ASTERISK-18416 - [patch] Realtime queue agents unavailable
      via AMI before a call event.
      (Reported by kwk)
 * ASTERISK-27597 - AMI Queuestatus not working (with realtime
      queue)
      (Reported by cagdas kopuz)
 * ASTERISK-29871 - res_prometheus: Failure to load causes
      FRACKs
      (Reported by Mark Petersen)
 * ASTERISK-29886 - Asterisk AMI sends not-valid XML
     
      (Reported by Napadailo Yaroslav)
 * ASTERISK-29888 - res_pjsip_outbound_authenticator_digest:
      ABRT attempting to clean up auth_sess
      (Reported by George
      Joseph)
 * ASTERISK-29857 - res_tonedetect: fix logic errors in code
   
      (Reported by N A)
 * ASTERISK-29854 - func_frame_drop: fix buffer usage typo
     
      (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-28219 - res_ari: Channel create and dial may cause
      "BUG! Must supply a channel name.." error
      (Reported by
      Anil Gupta)
 * 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)
 * ASTERISK-29779 - progdocs: Hidden code sections with syntax
      errors.
      (Reported by Alexander Traud)
 * ASTERISK-29732 - progdocs: Fix grouping for latest Doxygen
  
      (Reported by Alexander Traud)
 * ASTERISK-29771 - Crash occurs when 2 realtime sippeers mysql
      connections are configured and we have a schema warning
     
      (Reported by Mario Ban)
 * ASTERISK-29776 - stir/shaken: Requires GNU designator
     
      (Reported by Alexander Traud)
 * ASTERISK-29773 - progdocs: doxyref.h outdated
      (Reported
      by Alexander Traud)
 * ASTERISK-29765 - xmldoc: Fix for Doxygen
      (Reported by
      Alexander Traud)
 * ASTERISK-29730 - Segfault in __ao2_ref if refdebug = yes
    
      (Reported by Alexei Gradinari)
 * ASTERISK-29762 - channels: Fix for Doxygen
      (Reported by
      Alexander Traud)
 * ASTERISK-29748 - bridging: Infinite loop when both Local
      channel halves in same bridge
      (Reported by Joshua C. Colp)
 * ASTERISK-29754 - odbc: Fix for Doxygen
      (Reported by
      Alexander Traud)
 * ASTERISK-29753 - parking: Fix for Doxygen
      (Reported by
      Alexander Traud)
 * ASTERISK-29756 - res_ari: Fix for Doxygen
      (Reported by
      Alexander Traud)
 * ASTERISK-29755 - frame: Fix for Doxygen
      (Reported by
      Alexander Traud)
 * ASTERISK-29750 - stasis: Fix for Doxygen
      (Reported by
      Alexander Traud)
 * ASTERISK-29752 - app: Fix for Doxygen
      (Reported by
      Alexander Traud)
 * ASTERISK-29749 - res_xmpp: Fix for Doxygen
      (Reported by
      Alexander Traud)
 * ASTERISK-29751 - channel: Fix for Doxygen
      (Reported by
      Alexander Traud)
 * ASTERISK-29737 - chan_iax2: Fix for Doxygen
      (Reported
      by Alexander Traud)
 * ASTERISK-29747 - res_pjsip: Fix for Doxygen
      (Reported
      by Alexander Traud)
 * ASTERISK-29743 - bridges: Fix for Doxygen
      (Reported by
      Alexander Traud)
 * ASTERISK-29742 - addons: Fix for Doxygen.
      (Reported by
      Alexander Traud)
 * ASTERISK-29740 - apps: Fix for Doxygen
      (Reported by
      Alexander Traud)
 * ASTERISK-29741 - tests: Fix for Doxygen
      (Reported by
      Alexander Traud)
 * ASTERISK-29735 - progdocs: Avoid multiple use of section
      labels
      (Reported by Alexander Traud)
 * ASTERISK-29734 - progdocs: Use Doxygen \example correctly
   
      (Reported by Alexander Traud)
 * ASTERISK-29736 - bridge_channel: Fix for Doxygen
     
      (Reported by Alexander Traud)
 * ASTERISK-29733 - progdocs: Avoid name with Doxygen \file
    
      (Reported by Alexander Traud)
 * ASTERISK-29744 - app_morsecode: Fix deadlock
      (Reported
      by N A)
 * ASTERISK-29705 - app_read: Fix custom terminator
      functionality regression
      (Reported by N A)
 * ASTERISK-29703 - res_pjsip_callerid: Fix OLI parsing
     
      (Reported by N A)
 * ASTERISK-29724 - BuildSystem: In POSIX sh, == in place of =
      is undefined.
      (Reported by Alexander Traud)
 * ASTERISK-28040 - pbx: "dialplan reload" is removing minus
      symbol from dynamic hints
      (Reported by Daniel Zanutti)
 * ASTERISK-29702 - sig_analog: Fix truncated buffer copy
     
      (Reported by N A)
 * ASTERISK-29391 - VoiceMail does not cancel recording on
      rerecord hangup
      (Reported by N A)
 * ASTERISK-29709 - res_snmp: Not build on recent Debian
      distributions.
      (Reported by Alexander Traud)
 * ASTERISK-29717 - res_config_sqlite: not removed in
      makeopts.in
      (Reported by Alexander Traud)
 * ASTERISK-29710 - stasis: Clang 13 warns about the unused but
      set variable dispatched.
      (Reported by Alexander Traud)
 * ASTERISK-29711 - aelparse: GCC 11.2 found two maybe
      uninitialized
      (Reported by Alexander Traud)
 * ASTERISK-29713 - GCC 11.2: two stringop-overread
     
      (Reported by Alexander Traud)
 * ASTERISK-29682 - Squash compiler issues generated by gcc 11
 
      (Reported by George Joseph)
 * ASTERISK-29693 - Using --with-crypto and --with-ssl fails on
      a recompile
      (Reported by George Joseph)
 * ASTERISK-27816 - func_talkdetect's logic is completely
      broken
      (Reported by Moritz Fain)
 * ASTERISK-26497 - make install downloads x86_32 variants of
      external modules on non Intel architectures
      (Reported by
      Corey Farrell)
 * ASTERISK-29691 - stun: Not all users provide a dst to
      ast_stun_request
      (Reported by Dennis Haney)
 * ASTERISK-20219 - [patch] - IAX2 Call Encryption Fails with
      RSA authentication
      (Reported by Michael Munger)
 * ASTERISK-29402 - res_pjsip_t38: Socket is bound to IPv4/IPv6
      but platform does not support it
      (Reported by Matthew
      Kern)
 * ASTERISK-29673 - app_read: Fix null pointer crash regression

      (Reported by N A)
 * ASTERISK-29671 - res_rtp_asterisk: memory leak
     
      (Reported by Jean Aunis - Prescom)
 * ASTERISK-29668 - ari: Listing bridges fails when dialing
      bridge exists
      (Reported by Joshua C. Colp)
 * ASTERISK-29663 - messaging: AMI MessageSend does not support
      same parameters as dialplan application
      (Reported by Brian
      J. Murrell)
 * ASTERISK-29578 - app_queue: Custom device state using
      included hints do not update
      (Reported by N A)
 * ASTERISK-29660 - Build failure when disabling PJSIP support
 
      (Reported by Guido Falsi)
 * ASTERISK-29654 - pjproject includes trailing whitespace in
      sdp format attributes
      (Reported by George Joseph)
 * ASTERISK-29635 - MP3Player don' t work with actual mpg123
      versions
      (Reported by Carlos Oliva)
 * ASTERISK-29629 - ARI external media channel creation doesn't
      set option data
      (Reported by sungtae kim)
 * ASTERISK-27176 - test_abstract_jb: frames leak
     
      (Reported by Corey Farrell)
 * ASTERISK-29634 - res_snmp:  gcc 11 needs -fPIC to compile
      correctly
      (Reported by George Joseph)
 * ASTERISK-29630 - Asterisk is unable to read extended number
      format terminfo files
      (Reported by Sean Bright)
 * ASTERISK-28004 - dns: Core ast_dns_get_nameservers does not
      support configured IPv6 servers
      (Reported by Isaac
      McDonald)
 * ASTERISK-29618 - ConfBridge errors on creation conference
      room
      (Reported by Alexander Zharov)
 * ASTERISK-29622 - ARI: external media create doesn't use body
      parameter
      (Reported by sungtae kim)
 * ASTERISK-29614 - app_agent_pool: XML Doc: unterminated entity
      reference
      (Reported by Alexander Traud)
 * ASTERISK-29609 - Subsequent 'ael reload' will cause a lock
      up
      (Reported by Mark Murawski)
 * ASTERISK-28701 - app_queue: Core reload resets queue stats,
      even when keepstats=yes
      (Reported by Luke Escude)
 * ASTERISK-29616 - res_rtp_asterisk: sqrt(.) requires the
      header math.h.
      (Reported by Alexander Traud)
 * ASTERISK-29518 - sig_analog: FCG_CAMA fails to signal ANI
      spill when using MF signaling
      (Reported by Sarah Autumn)
 * ASTERISK-29582 - res_pjproject: Can't map pjproject log
      messages to Asterisk TRACE
      (Reported by George Joseph)
 * ASTERISK-29575 - app_milliwatt: Milliwatt application doesn't
      use the proper timings
      (Reported by N A)
 * ASTERISK-20339 - chan_mgcp, resp_pktccops ast_debug support
 
      (Reported by Tomas Maldonado)
 * ASTERISK-29540 - aelparse: include of context with timings
      fails
      (Reported by Alexander Traud)
 * ASTERISK-29539 - Segmentation fault at ast_writestream() when
      write handler not defined (happens with OGG/Speex)
     
      (Reported by Ernani Jos�� Camargo Azevedo)
 * ASTERISK-29494 - cdr_adaptive_odbc: Prevent throwing warnings
      if CDR filtering is used
      (Reported by N A)
 * ASTERISK-29513 - statsd: Remove non-standard metric type
      Meter
      (Reported by Rijnhard Hessel)
 * ASTERISK-12 - app_voicemail2 became a bit silent, lately
    
      (Reported by siggi)
 * ASTERISK-29526 - G729 audio gets corrupted by Asterisk due to
      smoother
      (Reported by under)
 * ASTERISK-29392 - chan_iax2: Asterisk crashes when queueing
      video with format
      (Reported by Michael Welk)

Improvements made in this release:
-----------------------------------
 * ASTERISK-30241 - res_pjsip_gelocation: Downgrade some NOTICE
      scope trace debugs to DEBUG level
      (Reported by N A)
 * ASTERISK-30178 - extend user_eq_phone behavior to local
      uri's
      (Reported by Michael Bradeen)
 * ASTERISK-30046 - Reimplement res/res_crypto.c internals with
      EVP_PKEY interface to Openssl API's
      (Reported by Philip
      Prindeville)
 * ASTERISK-30045 - Add test coverage to res/res_crypto.c
      functionality
      (Reported by Philip Prindeville)
 * ASTERISK-30209 - pbx_variables: Use const char for
      pbx_substitute_variables_helper_full_location
      (Reported by
      N A)
 * ASTERISK-30185 - res_geolocation: Allow location parameters
      to be specified in profiles
      (Reported by George Joseph)
 * ASTERISK-30177 - res_geolocation:  Add option to suppress
      empty elements
      (Reported by George Joseph)
 * ASTERISK-30182 - res_geolocation: Add built-in profiles to
      use in fully dynamic configurations
      (Reported by George
      Joseph)
 * ASTERISK-29906 - [patch] update RLS to reflect the changes to
      the lists
      (Reported by Alexei Gradinari)
 * ASTERISK-30163 - general: fix minor formatting issues
     
      (Reported by N A)
 * ASTERISK-30164 - chan_iax2: Add missing option documentation

      (Reported by N A)
 * ASTERISK-30153 - logger: Improve log levels
      (Reported
      by N A)
 * ASTERISK-30160 - cdr.conf: Remove obsolete app_mysql
      reference
      (Reported by N A)
 * ASTERISK-30159 - general: Remove obsolete SVN references
    
      (Reported by N A)
 * ASTERISK-30128 - Create PJSIP interface module for
      Geolocation
      (Reported by George Joseph)
 * ASTERISK-30127 - Create core Geolocation capability for
      Asterisk
      (Reported by George Joseph)
 * ASTERISK-30089 - general: fix typos
      (Reported by N A)
 * ASTERISK-30050 - Upgrade Asterisk to bundled pjproject
      2.12.1
      (Reported by Stanislav Abramenkov)
 * ASTERISK-30090 - xmldocs: Use example tags for examples
     
      (Reported by N A)
 * ASTERISK-29891 - [patch] provide a display name for RLS
      subscriptions
      (Reported by Alexei Gradinari)
 * ASTERISK-30086 - res_parking: Warn when invalid parking space
      requested
      (Reported by N A)
 * ASTERISK-30058 - Evaluate dialplan functions and variables in
      agi exec
      (Reported by Shloime Rosenblum)
 * ASTERISK-30027 - ari: expose channel driver's unique id (i.e.
      Call-ID for chan_sip/chan_pjsip) in ARI channel resource
     
      (Reported by Moritz Fain)
 * ASTERISK-29845 - res_pjsip_outbound_registration: Show time
      remaining until registration lapses
      (Reported by N A)
 * ASTERISK-24827 - Missing documentation for chan_dahdi dial
      string ring cadences
      (Reported by Scott Griepentrog)
 * ASTERISK-29940 - general: Add since tags to xmldocs
     
      (Reported by N A)
 * ASTERISK-30008 - samples: Remove obsolete config files
     
      (Reported by N A)
 * ASTERISK-29726 - Add Asterisk External Application Protocol
      (AEAP) implementation
      (Reported by Kevin Harwell)
 * ASTERISK-29951 - app_mf, app_sf: Return -1 on hangup
     
      (Reported by N A)
 * ASTERISK-29954 - app_meetme: Emit warning if conference not
      found
      (Reported by N A)
 * ASTERISK-29935 - build: Remove leftover build references
    
      (Reported by N A)
 * ASTERISK-29351 - Qualify pjproject 2.12 for Asterisk
     
      (Reported by George Joseph)
 * ASTERISK-29976 - Should Readme include information about
      install_prereq script?
      (Reported by Marcel Wagner)
 * ASTERISK-29970 - Use pkg-config to find libxml2 headers and
      libraries
      (Reported by Hugh McMaster)
 * ASTERISK-25716 - Documentation: Document explanations and
      examples for possible values of DIALSTATUS
      (Reported by
      Rusty Newton)
 * ASTERISK-29980 - build: External binary modules don't use
      https
      (Reported by INVADE International Ltd.)
 * ASTERISK-29967 - pbx_builtins: Add missing documentation
    
      (Reported by N A)
 * ASTERISK-29909 - app_queue: Add support for withdrawing a
      call
      (Reported by Kfir Itzhak)
 * ASTERISK-29353 - Qualify jansson 2.14 for asterisk
     
      (Reported by George Joseph)
 * ASTERISK-29897 - channels: Increase core debug levels for
      chatty debugs
      (Reported by N A)
 * ASTERISK-29896 - xmldocs: Add since tag
      (Reported by N
      A)
 * ASTERISK-29861 - asterisk.h: add macro for curl user agent
  
      (Reported by N A)
 * ASTERISK-29809 - curl, stir_shaken: refactor curl code
     
      (Reported by N A)
 * ASTERISK-29920 - app_voicemail: Warn if trying to manage
      nonexistent mailbox
      (Reported by N A)
 * ASTERISK-29925 - func_db: Warn about malformed key names
    
      (Reported by N A)
 * ASTERISK-29866 - cli: add core dump information to core show
      settings
      (Reported by N A)
 * ASTERISK-29898 - documentation: Add default attributes to
      documentation
      (Reported by N A)
 * ASTERISK-29900 - app_mp3: Document and warn about https
      incompatibility
      (Reported by N A)
 * ASTERISK-29877 - app_mf: Allow reading a maximum number of
      digits
      (Reported by N A)
 * 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)
 * ASTERISK-29777 - documentation: Standardize example syntax
  
      (Reported by N A)
 * ASTERISK-29715 - app_voicemail: Refactor email generation
      functions
      (Reported by N A)
 * ASTERISK-29727 - Add type for JSON stasis message RTCP Report
      Received/Sent
      (Reported by Boris P. Korzun)
 * ASTERISK-29714 - Spelling errors
      (Reported by Josh
      Soref)
 * ASTERISK-29707 - chan_iax2: Allow both key and secret to be
      specified at dial time
      (Reported by N A)
 * ASTERISK-29662 - Add mix option to Playback application for
      say and filename
      (Reported by Shloime Rosenblum)
 * ASTERISK-29637 - Add support for future dates in Say.c
     
      (Reported by Shloime Rosenblum)
 * ASTERISK-29525 - PJSIP remove_existing unavailable contacts
 
      (Reported by Joseph Nadiv)
 * ASTERISK-29661 - func_vmcount: Add support for multiple
      mailboxes
      (Reported by N A)
 * ASTERISK-29275 - Support of MIME-type for wav16
     
      (Reported by Boris P. Korzun)
 * ASTERISK-29529 - Add custom logging level
      (Reported by
      N A)
 * ASTERISK-29472 - res_pjsip: OLI/ANI2 support missing
     
      (Reported by N A)
 * ASTERISK-29626 - app_stack: Include calling location if
      attempting to branch to nonexistent location
      (Reported by
      N A)
 * ASTERISK-29632 - Add option to Application_VoiceMail to
      suppress instructions only when a custom greeting is present
   
      (Reported by Charlie Smurthwaite)
 * ASTERISK-29605 - chan_iax2: Add ANI2
      (Reported by N A)
 * ASTERISK-29508 - STUN server address refresh
      (Reported
      by S��bastien Duthil)
 * ASTERISK-29612 - bridge_basic: Don't throw warning if
      attended transfer is cancelled
      (Reported by N A)
 * ASTERISK-29544 - Media Cache - Delayed remote sound file
      retrieve delays all playbacks
      (Reported by Andre Barbosa)
 * ASTERISK-29495 - Return integer instead of float if response
      is a whole number
      (Reported by N A)
 * ASTERISK-29541 - app_morsecode: Add American Morse code
     
      (Reported by N A)
 * ASTERISK-29543 - app_originate: Allow specifying codec(s) to
      use
      (Reported by N A)
 * ASTERISK-29528 - Add support for multiple files for agent
      announcements
      (Reported by N A)
 * ASTERISK-29527 - res_http_media_cache: Cleanup audio format
      lookup in HTTP requests
      (Reported by Sean Bright)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0

Thank you for your continued support of Asterisk!
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