[asterisk-dev] Certified Asterisk 18.9-cert1 Now Available

Asterisk Development Team asteriskteam at digium.com
Thu Apr 28 08:43:49 CDT 2022

The Asterisk Development Team would like to announce the release of Certified Asterisk 18.9-cert1.
This release is available for immediate download at

The release of Certified Asterisk 18.9-cert1 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-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-29554 - cdr_mysql: Deprecated in 1.8, to be removed
      in 19
      (Reported by Joshua C. Colp)
 * ASTERISK-29555 - app_mysql: Deprecated in 1.8, to be removed
      in 19
      (Reported by Joshua C. Colp)
 * ASTERISK-29557 - app_ices: Deprecated in 16, to be removed in
      (Reported by Joshua C. Colp)
 * ASTERISK-29558 - app_macro: Deprecated in 16, to be removed
      in 21
      (Reported by Joshua C. Colp)
 * ASTERISK-29559 - app_fax: Deprecated in 16, to be removed in
      (Reported by Joshua C. Colp)
 * ASTERISK-29560 - app_url: Deprecated in 16, to be removed in
      (Reported by Joshua C. Colp)
 * ASTERISK-29561 - app_image: Deprecated in 16, to be removed
      in 19
      (Reported by Joshua C. Colp)
 * ASTERISK-29562 - app_nbscat: Deprecated in 16, to be removed
      in 19
      (Reported by Joshua C. Colp)
 * ASTERISK-29563 - app_dahdiras: Deprecated in 16, to be
      removed in 19
      (Reported by Joshua C. Colp)
 * ASTERISK-29564 - cdr_syslog: Deprecated in 16, to be removed
      in 19
      (Reported by Joshua C. Colp)
 * ASTERISK-29565 - chan_oss: Deprecated in 16, to be removed in
      (Reported by Joshua C. Colp)
 * ASTERISK-29566 - chan_phone: Deprecated in 16, to be removed
      in 19
      (Reported by Joshua C. Colp)
 * ASTERISK-29567 - chan_sip: Deprecated in 17, to be removed in
      (Reported by Joshua C. Colp)
 * ASTERISK-29568 - chan_nbs: Deprecated in 16, to be removed in
      (Reported by Joshua C. Colp)
 * ASTERISK-29569 - chan_misdn: Deprecated in 16, to be removed
      in 19
      (Reported by Joshua C. Colp)
 * ASTERISK-29570 - chan_vpb: Deprecated in 16, to be removed in
      (Reported by Joshua C. Colp)
 * ASTERISK-29571 - res_config_sqlite: Deprecated in 16, to be
      removed in 19
      (Reported by Joshua C. Colp)
 * ASTERISK-29572 - res_monitor: Deprecated in 16, to be removed
      in 21
      (Reported by Joshua C. Colp)
 * ASTERISK-29573 - conf2ael: Deprecated in 16, to be removed in
      (Reported by Joshua C. Colp)
 * ASTERISK-29574 - muted: Deprecated in 16, to be removed in
      (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-29945 - pjproject: Security fixes for things
      (Reported by Kevin Harwell)
 * 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)
 * ASTERISK-29305 - ASTERISK-29203 / AST-2021-002 -- Another
      scenario is causing a crash
      (Reported by Gregory Massel)
 * ASTERISK-29260 - sRTP Replay Protection ignored; even tears
      down long calls
      (Reported by Alexander Traud)
 * ASTERISK-29227 - res_pjsip_diversion: sending multiple 181
      responses causes memory corruption and crash
      (Reported by
      Ivan Poddubny)
 * ASTERISK-29219 - res_pjsip_diversion: Crash if Tel URI
      contains History-Info
      (Reported by Torrey Searle)
 * ASTERISK-29057 - pjsip: Crash on call rejection during high
      (Reported by Sandro Gauci)
 * ASTERISK-28589 - chan_sip: Depending on configuration an
      INVITE can alter Addr of a peer
      (Reported by Andrey  V.
 * ASTERISK-28580 - Bypass SYSTEM write permission in manager
      action allows system commands execution
      (Reported by Eliel
 * ASTERISK-28495 - res_pjsip_t38: 200 OK with SDP answer with
      declined stream causes crash
      (Reported by Alexei
 * ASTERISK-28447 - res_pjsip_messaging: In-dialog MESSAGE with
      no body causes crash
      (Reported by Gil Richard)
 * ASTERISK-28465 - Broken SDP can cause a segfault in a T.38
      (Reported by Francesco Castellano)
 * ASTERISK-28260 - Asterisk segfault when rtp negotiation is
      wrong or fails
      (Reported by Sotiris Ganouris)
 * ASTERISK-28127 - Buffer overflow for DNS SRV/NAPTR records
      (Reported by Jan Hoffmann)
 * ASTERISK-28013 - res_http_websocket: Crash when reading HTTP
      Upgrade requests
      (Reported by Sean Bright)

New Features made in this release:
 * ASTERISK-29720 - res_tonedetect: Add call progress tone
      (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
      by N A)
 * ASTERISK-29496 - Add SendMF application
      (Reported by N
 * ASTERISK-29627 - Add STRBETWEEN function
      (Reported by N
 * ASTERISK-29628 - Add file and directory functions
      (Reported by N A)
 * ASTERISK-29531 - Add SAYFILES function
      (Reported by N
 * 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
      (Reported by N A)
 * ASTERISK-29389 - Add PJSIP_HEADERS() and ability to read
      header by pattern
      (Reported by Igor Goncharovsky)
 * ASTERISK-11 - AGI channel_status failure
      (Reported by
 * ASTERISK-29477 - Function to asynchronously store digits
      (Reported by N A)
 * ASTERISK-29454 - New application to reload modules
      (Reported by N A)
 * ASTERISK-29444 - Add application to wait for condition
      (Reported by N A)
 * ASTERISK-29442 - app_dial: Expand A option to allow
      announcement playback to caller
      (Reported by N A)
 * ASTERISK-29446 - app_confbridge: New ConfKick application
      (Reported by N A)
 * ASTERISK-29440 - app_confbridge: Allow ConfBridge answer to
      be suppressed
      (Reported by N A)
 * ASTERISK-29431 - Minimum and maximum dialplan functions
      (Reported by N A)
 * ASTERISK-29439 - func_volume: Volume function can't be read
      (Reported by N A)
 * ASTERISK-27477 - Chan_pjsip does not support unauthenticated
      OPTIONS ping
      (Reported by Ross Beer)
 * ASTERISK-29027 - Implement support for History-Info
      (Reported by Torrey Searle)
 * ASTERISK-6863 - [patch] allow Asterisk to set high ToS bits
      as non-root on Linux
      (Reported by Matt Addison)
 * ASTERISK-17491 - CURLOPT() needs a "followlocation" parameter
      / "maxredirs" doesn't do anything
      (Reported by candrews)
 * ASTERISK-28639 - res_pjsip_endpoint_identifier_ip: Add
      ability to match on source port
      (Reported by Sean Bright)
 * ASTERISK-28614 - app_senddtmf: Allow "receiving" DTMF with
      PlayDTMF instead of only "sending"
      (Reported by laszlovl)
 * ASTERISK-28613 - func_curl: CURLOPT cannot set Content-Type
      (Reported by Martin Tomec)
 * ASTERISK-28533 - func_jitterbuffer: Add support for video
      (Reported by Joshua C. Colp)
 * ASTERISK-17808 - [patch] Unregister a realtime moh class
      (Reported by Byron Clark)
 * ASTERISK-28489 - Channel variable SIPFROMDOMAIN for
      chan_pjsip to setup From header URI domain
      (Reported by
      Stas Kobzar)
 * ASTERISK-28403 - Add native Prometheus support to Asterisk
      (Reported by Matt Jordan)
 * ASTERISK-28375 - res_pjsip: New configuration setting to
      allow disabling norefersub
      (Reported by Dan Cropp)
 * ASTERISK-28320 - Added ARI resource
      (Reported by
      sungtae kim)
 * ASTERISK-28267 - res_stasis: Add ability to switch
      (Reported by Benjamin Keith Ford)
 * ASTERISK-28087 - add flag to allow CALLERID(num) to be placed
      in Contact header in chan_pjsip
      (Reported by Torrey
 * ASTERISK-27971 - res_pjsip: Implement additional SIP RFCs for
      Google Voice trunk compatability
      (Reported by Nick French)

Bugs fixed in this release:
 * ASTERISK-30024 - Failed to sign STIR/SHAKEN payload with
      functionality not enabled
      (Reported by Claude Diderich)
 * ASTERISK-29859 - VoiceMailMain() fails when encountering
      non-numeric CALLERID(num)
      (Reported by Mark Murawski)
 * ASTERISK-29816 - SAY_DTMF_INTERRUPT channel variable is not
      (Reported by Sean Bright)
 * ASTERISK-29821 - Deadlock in bridge_channel_internal_join()
      on local channels.
      (Reported by Krzysztof Trempala)
 * ASTERISK-29779 - progdocs: Hidden code sections with syntax
      (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-29764 - chan_misdn: Fix for Doxygen
      by Alexander Traud)
 * ASTERISK-29773 - progdocs: doxyref.h outdated
      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-29755 - frame: Fix for Doxygen
      (Reported by
      Alexander Traud)
 * ASTERISK-29756 - res_ari: Fix for Doxygen
      (Reported by
      Alexander Traud)
 * ASTERISK-29751 - channel: 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-29742 - addons: Fix for Doxygen.
      (Reported by
      Alexander Traud)
 * ASTERISK-29747 - res_pjsip: Fix for Doxygen
      by Alexander Traud)
 * ASTERISK-29737 - chan_iax2: Fix for Doxygen
      by Alexander Traud)
 * ASTERISK-29743 - bridges: Fix for Doxygen
      (Reported by
      Alexander Traud)
 * ASTERISK-29741 - tests: Fix for Doxygen
      (Reported by
      Alexander Traud)
 * ASTERISK-29740 - apps: Fix for Doxygen
      (Reported by
      Alexander Traud)
 * ASTERISK-29733 - progdocs: Avoid name with Doxygen \file
      (Reported by Alexander Traud)
 * ASTERISK-29736 - bridge_channel: Fix for Doxygen
      (Reported by Alexander Traud)
 * ASTERISK-29735 - progdocs: Avoid multiple use of section
      (Reported by Alexander Traud)
 * ASTERISK-29734 - progdocs: Use Doxygen \example correctly
      (Reported by Alexander Traud)
 * ASTERISK-29744 - app_morsecode: Fix deadlock
      by N A)
 * ASTERISK-29703 - res_pjsip_callerid: Fix OLI parsing
      (Reported by N A)
 * ASTERISK-29705 - app_read: Fix custom terminator
      functionality regression
      (Reported by N A)
 * ASTERISK-29724 - BuildSystem: In POSIX sh, == in place of =
      is undefined.
      (Reported by Alexander Traud)
 * ASTERISK-29702 - sig_analog: Fix truncated buffer copy
      (Reported by N A)
 * ASTERISK-28040 - pbx: "dialplan reload" is removing minus
      symbol from dynamic hints
      (Reported by Daniel Zanutti)
 * ASTERISK-29391 - VoiceMail does not cancel recording on
      rerecord hangup
      (Reported by N A)
 * ASTERISK-29709 - res_snmp: Not build on recent Debian
      (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
      (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
      (Reported by Moritz Fain)
 * ASTERISK-29691 - stun: Not all users provide a dst to
      (Reported by Dennis Haney)
 * ASTERISK-26497 - make install downloads x86_32 variants of
      external modules on non Intel architectures
      (Reported by
      Corey Farrell)
 * 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
 * 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-29635 - MP3Player don' t work with actual mpg123
      (Reported by Carlos Oliva)
 * ASTERISK-29654 - pjproject includes trailing whitespace in
      sdp format attributes
      (Reported by George Joseph)
 * 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
      (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
 * ASTERISK-29618 - ConfBridge errors on creation conference
      (Reported by Alexander Zharov)
 * ASTERISK-29622 - ARI: external media create doesn't use body
      (Reported by sungtae kim)
 * ASTERISK-29614 - app_agent_pool: XML Doc: unterminated entity
      (Reported by Alexander Traud)
 * ASTERISK-29609 - Subsequent 'ael reload' will cause a lock
      (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
      (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
      (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
      (Reported by under)
 * ASTERISK-29392 - chan_iax2: Asterisk crashes when queueing
      video with format
      (Reported by Michael Welk)
 * ASTERISK-29507 - STUN timeout is silently delaying calls
      (Reported by S��bastien Duthil)
 * ASTERISK-27871 - Remote URL in playback must end with file
      (Reported by Caesar)
 * ASTERISK-29514 - ari: Audiosocket segfault when no data
      (Reported by Igor Goncharovsky)
 * ASTERISK-29503 - Updated identify/match syntax not supported
      by config wizard
      (Reported by Sean Bright)
 * ASTERISK-29480 - fixedjitterbuffer contains an un-wrappered
      assert that triggers on a negative time slew
      (Reported by
      Dan Cropp)
 * ASTERISK-29485 - core: Inband generation of tones for Busy()
      and Congestion() may not occur
      (Reported by Joshua C.
 * ASTERISK-29479 - [patch] Channels are not put on hold for
      Session Progress with inactive audio
      (Reported by Bernd
 * ASTERISK-29475 - SayNumber triggers WARNING if caller hangs
      up during application execution
      (Reported by N A)
 * ASTERISK-29404 - Consolidate res_pjsip_messaging fixes for
      domain name
      (Reported by George Joseph)
 * ASTERISK-29441 - Core reload making TCP endpoints go offline

      (Reported by Luke Escude)
 * ASTERISK-28237 - "FRACK!, Failed assertion bad magic number"
      happens when unsubscribe an application from an event source
      (Reported by Lucas Tardioli Silveira)
 * ASTERISK-28393 - Multidomain support issue
      (Reported by
      Andrea Sannucci)
 * ASTERISK-29433 - res_rtp_asterisk: Server reflexive
      candidates use incorrect raddr for RTCP
      (Reported by
 * ASTERISK-29397 - pjsip: Asterisk isn't tolerant of RFC8760
      (Reported by George Joseph)
 * ASTERISK-24601 - [patch]Missing RFC4235 tags and attributes
      in PJSIP NOTIFY event: dialog  XML body
      (Reported by Marco
 * ASTERISK-29370 - chan_sip does not recognize
      (Reported by N A)
 * ASTERISK-29377 - cpool_release_pool "double free or
      corruption (out)"
      (Reported by Robert Sutton)
 * ASTERISK-29372 - file.c switch does not account for flash
      (Reported by N A)
 * ASTERISK-29358 - chan_pjsip: Trace message for progress is
      output even if frame is not queued
      (Reported by Michael
 * ASTERISK-29407 - chan_local: Filtering audio formats should
      not occur on removed streams
      (Reported by Joshua C. Colp)
 * ASTERISK-29030 - res_rtp_asterisk: Additional RTP-frame (with
      wrong SSRC) gets inserted when switching from progress to
      (Reported by Matthias Hensler)
 * ASTERISK-29328 - translate.c: possible buffer overflow when
      (Reported by Jean Aunis - Prescom)
 * ASTERISK-29379 - Segfault - ast_channel_is_multistream
      (chan=0x0) at channel_internal_api.c:1590
      (Reported by
      Ross Beer)
 * ASTERISK-29130 - prometheus: Crash when scraping bridge
      (Reported by Francisco Correia)
 * ASTERISK-29364 - res_rtp_asterisk: standard deviation
      (Reported by Kevin Harwell)
 * ASTERISK-29373 - res_rtp_asterisk: Flash events are
      (Reported by N A)
 * ASTERISK-28356 - app_queue: CLI set ringinuse for realtime
      member not working
      (Reported by Michael)
 * ASTERISK-24434 - Fix differing usage of assignment operators
      in modules.conf
      (Reported by Rusty Newton)
 * ASTERISK-24631 - Incorrect description of option "context" in
      (Reported by Etienne Lessard)
 * ASTERISK-26614 - app_queue: updatecdr option in queues.conf
      does effectively nothing
      (Reported by Alexander Gonchiy)
 * ASTERISK-25358 - dateformat not read from logger.conf by
      remote console
      (Reported by Igor Liferenko)
 * ASTERISK-27542 - app_queue: When "queue show" CLI command is
      executed a crash occurs
      (Reported by Miguel Sanz)
 * ASTERISK-29215 - res_pjsip_session: NULL active_media_state
      topology caused asterisk crash
      (Reported by sungtae kim)
 * ASTERISK-29355 - app_queue: Queue member status message sent
      even if status doesn't change
      (Reported by Roman Pertsev)
 * ASTERISK-29035 - chan_local: Multistream support breaks T.38
      (Reported by Matthias Hensler)
 * ASTERISK-29354 - res_pjsip: Allow partial reloading of
      (Reported by Joshua C. Colp)
 * ASTERISK-29348 - menuselect doesn't return errors in many
      (Reported by George Joseph)
 * ASTERISK-29352 - res_rtp_asterisk: Fix frame delivery time
      when SSRC changes
      (Reported by Joshua C. Colp)
 * ASTERISK-29071 - app_confbridge: Memory rises when
      jitterbuffer enabled and muting over AMI occurs
      by Stefan Ruf)
 * ASTERISK-29329 - app_dial: DTMF to 'D' option gets duplicated
      if there are multiple progress events
      (Reported by N A)
 * ASTERISK-29306 - strings: Incorrect use of
      __attribute__((pure)) in ast_str_to_lower definition
      (Reported by Vitezslav Novy)
 * ASTERISK-29300 - res_rtp_asterisk: When native local bridging
      the remote SSRC becomes permanent
      (Reported by Sebastian
 * ASTERISK-29235 - res_pjsip_nat: Contact is rewritten on
      REGISTER responses with external_signaling_address
      (Reported by Brian Paboojian)
 * ASTERISK-29266 - ICE Role conflict with an unauthorized
      (Reported by Salah Ahmed)
 * ASTERISK-29105 - chan_pjsip: 180 Ringing with SDP not changed
      into progress
      (Reported by Sebastian Damm)
 * ASTERISK-29297 - say: Y2021 problem ��� Asterisk cannot say
      year 2021 in Dutch
      (Reported by Jacek Konieczny)
 * ASTERISK-29315 - res_pjsip: re-registration gets stuck if
      setting initial auth credentials fails
      (Reported by Nick
 * ASTERISK-29312 - res_fax: asterisk fails to publish the
      Stasis and ReceiveFax status messages if the remote Station ID
      contains invalid UTF-8 characters
      (Reported by Alexei
 * ASTERISK-16799 - Callee declined when 'beep' audio file does
      not exist
      (Reported by IAMJames_)
 * ASTERISK-29313 - res_pjsip_refer:  Segfault in progress
      (Reported by George Joseph)
 * ASTERISK-29293 - res_config_pgsql: Limit realtime_pgsql() to
      return one (no more) record
      (Reported by Boris P. Korzun)
 * ASTERISK-29303 - pjsip: Re-invite occurs when it shouldn't
      (Reported by Benjamin Keith Ford)
 * ASTERISK-29311 - res_odbc_transaction sets forcecommit
      default value based on isolation level instead of forcecommit
      (Reported by Jaco Kroon)
 * ASTERISK-28452 - pjsip: <sess-version> of SDP is not
      incremented though SDP may be changed on reinvite without SDP
      (Reported by Michael Maier)
 * ASTERISK-29287 - app.h: C++ compatibility broken
      (Reported by Jean Aunis - Prescom)
 * ASTERISK-28369 - app_queue: Member device state "invalid"
      when second call is ringing and hint is used
      (Reported by
      Boolah )
 * ASTERISK-29203 - res_pjsip_t38: Crash when changing state
      (Reported by Gregory Massel)
 * ASTERISK-29205 - res_rtp_asterisk: Asterisk crashes when
      making hold/unhold from webrtc client
      (Reported by Edvin
 * ASTERISK-29196 - res_pjsip: Segmentation fault
      (Reported by Mauri de Souza Meneguzzo (3CPlus))
 * ASTERISK-29280 - chan_sip: Allow peers without audio
      (Reported by Alexander Traud)
 * ASTERISK-29265 - chan_sip: Allow text+video media streams,
      (Reported by Alexander Traud)
 * ASTERISK-29261 - res_pjsip: user=phone validation fail for
      isup numbers containing *#
      (Reported by Mark Petersen)
 * ASTERISK-29259 - channel: Allow text+video media streams,
      (Reported by Alexander Traud)
 * ASTERISK-29258 - chan_sip: Audio stream rejected, Other
      stream present: Invalid SDP.
      (Reported by Alexander Traud)
 * ASTERISK-29220 - After T38 reinvite response of 488 a
      subsequent G711 reinvite is not processed correctly. Instead the
      previous T38 session media is used
      (Reported by Robert
 * ASTERISK-29248 - res_pjsip_session: res sometimes
      uninitialized reported by compiler Clang.
      (Reported by
      Alexander Traud)
 * ASTERISK-29229 - Stasis/messaging: text messages not
      dispatched to all subscribers when using generic subscription
      (Reported by Jean Aunis - Prescom)
 * ASTERISK-29240 - chan_pjsip: Incoming PJSIP calls set global
      SIPDOMAIN instead of a channel variable
      (Reported by Ivan
 * ASTERISK-29238 - chan_sip: SDP: Offers without any enabled
      stream are accepted.
      (Reported by Alexander Traud)
 * ASTERISK-29237 - chan_sip: SDP: m=video is parsed even when
      (Reported by Alexander Traud)
 * ASTERISK-29222 - chan_sip: Hold/Resume an sRTP call on a
      video enabled user-agent.
      (Reported by Alexander Traud)
 * ASTERISK-27902 - chan_pjsip isn't updating hangupcause on 4XX
      (Reported by George Joseph)
 * ASTERISK-28016 - PJSIP sends duplicate 183 Progress
      (Reported by Alex Hermann)
 * ASTERISK-28185 - chan_pjsip: Subsequent same responses are
      not stopped
      (Reported by Julien)
 * ASTERISK-29230 - pjsip: Asterisk goes crazy and massively
      spams logfile if registration can't be send
      (Reported by
      Michael Maier)
 * ASTERISK-29231 - pjsip: SIGSEGV in CLI if no trunk is
      (Reported by Michael Maier)
 * ASTERISK-29217 - LOCK() can grant the same lock to multiple
      channels spuriously
      (Reported by Jaco Kroon)
 * ASTERISK-29201 - Crash occurs when Transfer and execute
      Hangup before the Transfer result 
      (Reported by Dan Cropp)
 * ASTERISK-28947 - Segmentation fault in mixmonitor_ds_destroy

      (Reported by Robert Sutton)
 * ASTERISK-29168 - Asterisk crashes during call transfer
      (Reported by Dalius Mockevicius)
 * ASTERISK-29210 - res_pjsip: Crash when examining transport
      (Reported by N GM )
 * ASTERISK-29191 - tel: URI in Diversion header causes crash
      (Reported by Mikhail Ivanov)
 * ASTERISK-28883 - Spyee information ist missing in ChanSpyStop
      AMI Event
      (Reported by Hendrik Wedhorn)
 * ASTERISK-29188 - null media causing the Asterisk crash
      (Reported by sungtae kim)
 * ASTERISK-29024 - pjsip: Route Header in Cancel request
      incorrectly set
      (Reported by Flole Systems)
 * ASTERISK-29209 - Debug messages printed by scope trace might
      be missing newlines
      (Reported by Alexander Traud)
 * ASTERISK-29211 - res_musiconhold: Segfault on realtime music
      on hold without entries
      (Reported by Nathan Bruning)
 * ASTERISK-29022 - Crash when manipulating PJSIP invite dlg ref
      (Reported by Sean Bright)
 * ASTERISK-29173 - Media cache URL requests allow infinite
      (Reported by Sean Bright)
 * ASTERISK-29175 - res_pjsip_stir_shaken: Fix module
      (Reported by Stanislav Abramenkov)
      (Reported by Alexander Traud)
 * ASTERISK-29165 - res_pjsip: malformed header Accept-Encoding
      in OPTIONS response
      (Reported by Alexander Greiner-Baer)
 * ASTERISK-28798 - [patch] chan_sip: TCP/TLS client without
      (Reported by Alexander Traud)
 * ASTERISK-29161 - Incorrect setup of recall channels
      (Reported by Boris P. Korzun)
 * ASTERISK-29155 - app_queue: Deadlock between queues container
      and individual queues
      (Reported by George Joseph)
 * ASTERISK-28933 - res_pjsip.so fails to load when bundled
      pjproject is compiled without libssl
      (Reported by Walter
 * ASTERISK-28825 - Any curl response checks out as valid even
      if 404 is returned.
      (Reported by dovid)
 * ASTERISK-29013 - res_pjsip: Asterisk doesn't stop sending
      invites (with auth) on 407 replies
      (Reported by Sebastian
 * ASTERISK-29142 - sip_to_pjsip.py: doesn't read globbed
      (Reported by Michael Newton)
 * ASTERISK-29144 - GCC Warnings with OPTIMIZE=-Og make
      (Reported by Alexander Traud)
 * ASTERISK-29145 - GCC Warnings with OPTIMIZE=-Os make
      (Reported by Alexander Traud)
 * ASTERISK-29146 - GCC Warnings: ���%s��� directive argument is
      (Reported by Alexander Traud)
 * ASTERISK-29124 - res_pjsip: flow transport broken for
      outbound requests
      (Reported by Nick French)
 * ASTERISK-29136 - config: Sample features.conf incorrectly
      includes " around sound files
      (Reported by Benjamin M.)
 * ASTERISK-29123 - logger.conf.sample missing comment mark on
      line 115
      (Reported by Andrew Siplas)
 * ASTERISK-29109 - res_pjsip_session: Asterisk 18 does not
      progress calls due to codec negotiation after upgrading from
      Asterisk 16
      (Reported by Ross Beer)
 * ASTERISK-28430 - res_rtp_asterisk.c: FRACK!, Failed assertion
      errno != EBADF
      (Reported by under)
 * ASTERISK-29108 - resource_endpoints.c : Memory leak if
      endpoint not found
      (Reported by Jean Aunis - Prescom)
 * ASTERISK-26424 - app_voicemail: Undocumented behavior from
      (Reported by Eric Smith)
 * ASTERISK-29097 - res_pjsip_config_wizard: Crash when freeing
      string when failing to add extension
      (Reported by Vieri)
 * ASTERISK-29091 - Crash when ast_translator_build_path fails
      (Reported by Jasper van der Neut)
 * ASTERISK-29051 - res_pjsip_sdp_rtp: Does not set correct
      values on RTP instance when "auto" DTMF is used
      by Sebastian Damm)
 * ASTERISK-29099 - res_musiconhold: Realtime MOH only loads a
      single entry
      (Reported by laszlovl)
 * ASTERISK-28311 - dsp: ast_dsp_silence_noise_with_energy wrong
      judgment of frame format
      (Reported by ���������)
 * ASTERISK-24329 - Music On Hold announcement cuts intro of
      music the first time it is played
      (Reported by Thomas
 * ASTERISK-29085 - func_curl: Segmentation fault when using
      CURL after setting httpheader CURLOPT
      (Reported by P��ter
 * ASTERISK-29089 - RTP Ports not cleared after hangup
      (Reported by Ross Beer)
 * ASTERISK-29081 - res_stasis: Add compare function for bridges
      moh container
      (Reported by Hajek Michal)
 * ASTERISK-28416 - Unable to get rtp codec payload code for
      (Reported by Brian J. Murrell)
 * ASTERISK-29014 - res_pjsip_session: Re-INVITE collisions
      aren't handled correctly
      (Reported by George Joseph)
 * ASTERISK-25665 - Duplicate logging in queue log for EXITEMPTY
      (Reported by Ove Aursand)
 * ASTERISK-29043 - app_queue: Leave empty sometimes not
      recorded as abandoned
      (Reported by Kfir Itzhak)
 * ASTERISK-29042 - res_parking: Parker UUID is no longer
      (Reported by Misha Vodsedalek)
 * ASTERISK-28878 - chan_pjsip: PJSIP_MEDIA_OFFER Broken
      asterisk 16
      (Reported by Joseph Ades)
 * ASTERISK-29046 - pbx: Deadlock when doing a reload, while
      simultaneously doing an ExtensionState on a pattern match hint
      that ends up adding an extension
      (Reported by Ramarajan)
 * ASTERISK-29040 - res_speech: Assertion on format
      (Reported by Nickolay V. Shmyrev)
 * ASTERISK-29001 - chan_pjsip does not process or forward 181
      (Reported by Torrey Searle)
 * ASTERISK-29034 - Lastpause of realtime members is reseting
      (Reported by Evandro C��sar Arruda)
 * ASTERISK-27273 - app_voicemail: When a voicemail is marked as
      "Urgent", it is not sent by email/processed by the mailcmd
      (Reported by Leandro Dardini)
 * ASTERISK-29033 - res_pjsip_session: Aggressively terminates
      session on failed re-INVITE
      (Reported by Joshua C. Colp)
 * ASTERISK-28974 - res_rtp_asterisk: T.140 messages have
      appended RTP string to each message block.
      (Reported by
      Thomas Johnson)
 * ASTERISK-29011 - chan_sip: ToHost property not cleared on
      (Reported by Dennis)
      certified versions
      (Reported by cmaj)
 * ASTERISK-28927 - Asterisk crash in music on hold
      (Reported by David Cunningham)
 * ASTERISK-28973 - Malformed IP address in SDP of 2nd SIP timer
      triggered INVITE when NAT is active (UDP transport with
      (Reported by Michael Neuhauser)
 * ASTERISK-28995 - res_pjsip_registrar: Expires on statically
      configured contacts is not correct
      (Reported by tootai)
 * ASTERISK-28987 - BridgeCreated ARI event shows wrong
      video_mode info
      (Reported by sungtae kim)
 * ASTERISK-28978 - acl: named_acl rule misconfiguration results
      in segfault on reading rule from realtime
      (Reported by
      Andrew Yager)
 * ASTERISK-28975 - res_http_websocket: Text payload data
      doesn't necessary include trailing zero
      (Reported by
      Nickolay V. Shmyrev)
 * ASTERISK-28951 - Inconsistent behaviour queues.conf when
      there is (not) a [general] section
      (Reported by Walter
 * ASTERISK-28965 - res_pjsip: Apply outbound proxy to static
      contacts on AOR
      (Reported by Joshua C. Colp)
 * ASTERISK-28930 - ./configure --without-ssl build failure
      (Reported by Jaco Kroon)
 * ASTERISK-28957 - chan_sip: chan_sip does not process 400
      response to an INVITE.
      (Reported by Frederic LE FOLL)
 * ASTERISK-28886 - chan_pjsip: PJSIP_SC_NULL does not exist in
      pjproject 2.7.2
      (Reported by Jared Smith)
 * ASTERISK-28888 - res_corosync: causes asterisk crash in huge
      distributed environment.
      (Reported by Universit�� di
      Bologna - CESIA VoIP)
 * ASTERISK-28954 - StreamEcho() only returns 1 active stream
      (Reported by Bill Kervaski)
 * ASTERISK-28955 - "setvar" doesn't work properly in
      (Reported by Marin Odrljin)
 * ASTERISK-28953 - res_pjsip_session: Preserve stream label
      (Reported by Joshua C. Colp)
 * ASTERISK-28942 - res_sorcery_memory_cache: Individual object
      expiration behaves unexpectedly with full backend caching
      (Reported by Joshua C. Colp)
 * ASTERISK-28950 - Stale code in app_queue to check untouched
      (Reported by Walter Doekes)
 * ASTERISK-28644 - Stale comment in app_queue about ring_entry
      (Reported by Walter Doekes)
 * ASTERISK-28952 - Queue wrapuptime sometimes not respected
      (based on stale lastcall time)
      (Reported by Walter Doekes)
 * ASTERISK-28938 - core_unreal / core_local: Add support for
      multistream and re-negotiation
      (Reported by Joshua C.
 * ASTERISK-28948 - ARI channel create doesn't referencing the
      channel_id parameter
      (Reported by sungtae kim)
 * ASTERISK-28939 - res_rtp_asterisk: Don't have send/receive
      buffers on non-WebRTC
      (Reported by Joshua C. Colp)
 * ASTERISK-28944 - bridge_softmix: Transitioning a stream from
      inactive -> sendrecv/sendonly doesn't re-negotiation
      (Reported by Joshua C. Colp)
 * ASTERISK-28923 - T.38 Segfaults in chan_pjsip_queryoption
      (Reported by Yury Kirsanov)
 * ASTERISK-28940 - /channels/create doesn't get any parameters
      from the body
      (Reported by sungtae kim)
 * ASTERISK-28936 - res_pjsip: crash when dialing non-sip uri
      (Reported by Walter Doekes)
 * ASTERISK-28900 - res_fax: Double frame free when gateway in
      use with off-nominal format usage
      (Reported by Gregory
 * ASTERISK-28929 - pjproject_bundled: Honor
      (Reported by Alexander Traud)
 * ASTERISK-28932 - res_pjsip_logger writing too big packets
      (Reported by nappsoft)
 * ASTERISK-28920 - bridge show all causes crash
      by sungtae kim)
 * ASTERISK-28921 - Wrong return value check for fwrite when
      writing to pcap file
      (Reported by nappsoft)
 * ASTERISK-28794 - res_pjsip: Crash when escaping during URI
      (Reported by nappsoft)
 * ASTERISK-28884 - x-ast-orig-host not filtered out from
      request URI and To header
      (Reported by nappsoft)
 * ASTERISK-28871 - res_pjsip_session: Unnecessary re-Invite on
      call answer
      (Reported by Alexei Gradinari)
 * ASTERISK-28903 - res_srtp: Answered Crypto Suite might be
      wrong in SDP/SDES.
      (Reported by Alexander Traud)
 * ASTERISK-28898 - bridge_softmix: Conference bridge not
      passing silent rtp packets
      (Reported by Jonathan Hunter)
 * ASTERISK-28892 - res_musiconhold: Module res_musiconhold
      throws false warning
      (Reported by Nicholas John Koch)
 * ASTERISK-28904 - RTP ICE leaks the memory
      (Reported by
      sungtae kim)
 * ASTERISK-26780 - res_pjsip: PJSIP Registration Fails when
      (Reported by Peter Sokolov)
 * ASTERISK-28854 - SIGSEGV when pjsip show history encounters
      IPV6 address
      (Reported by Roger James)
 * ASTERISK-28797 - [patch] tcptls: Fix notice when TLS is
      enabled but not configured.
      (Reported by Alexander Traud)
 * ASTERISK-28804 - [patch] app_osplookup.c: Avoid a format
      (Reported by Alexander Traud)
 * ASTERISK-28776 - Non async-signal-safe syscalls used after
      fork before exec
      (Reported by nappsoft)
 * ASTERISK-28870 - streams: One memory leak and one issue
      cloning streams
      (Reported by George Joseph)
 * ASTERISK-28829 - app_queue: leaking stasis subscription when
      Redirecting call 
      (Reported by laszlovl)
 * ASTERISK-25844 - app_queue: Ghost channels in "core show
      channels" output
      (Reported by Etienne Lessard)
 * ASTERISK-28859 - pjsip: Increase maximum candidate count
      (Reported by Joshua C. Colp)
 * ASTERISK-22920 - Crash while Forwarding from TLS extension
      with CHANNEL args secure_bridge_media and
      (Reported by Shlomi Gutman)
 * ASTERISK-28852 - Unprotected access to nochecksums variable,
      causes build failures
      (Reported by Guido Falsi)
 * ASTERISK-28848 - app_fax: Compile.
      (Reported by
      Alexander Traud)
 * ASTERISK-28846 - stream: Enforce formats immutability
      (Reported by Joshua C. Colp)
 * ASTERISK-28847 - ARI channels cuts the endpoint string over
      80 characters
      (Reported by sungtae kim)
 * ASTERISK-28811 - Crash occurs when fax session switches from
      T.38 to audio
      (Reported by Alexey Vasilyev)
 * ASTERISK-28839 - Sporadic crashes with Segmentation fault
      (Reported by Joeran Vinzens)
 * ASTERISK-28835 - IPv6 addresses in SDP incorrectly formatted

      (Reported by Daniel Heckl)
 * ASTERISK-28372 - Asterisk REPLY Wrong Contact header port
      (Reported by Anton Satskiy)
 * ASTERISK-24428 - Document that Asterisk will use the default
      SIP ports (5060 for TCP, 5061 for TLS) if the extern option
      variants aren't used
      (Reported by sstream)
      not mention
      (Reported by Alexander Traud)
 * ASTERISK-28841 - app_confbridge: Add support for disabling
      text messaging for a user
      (Reported by Joshua C. Colp)
 * ASTERISK-28837 - pjproject_bundled: Honor
      (Reported by Alexander Traud)
 * ASTERISK-28827 - res_rtp_asterisk: Loop when receive buffer
      is flushed by a received packet that is also in receive buffer
      with NACK
      (Reported by nappsoft)
 * ASTERISK-27195 - chan_sip: only sets ToS bits on UDP socket,
      ignoring TCP and TLS sockets
      (Reported by Joshua Roys)
 * ASTERISK-28826 - res_rtp_asterisk: Duplicate seqnos being
      added to send buffer with NACK
      (Reported by nappsoft)
 * ASTERISK-28812 - First DTMF is not get
      (Reported by
      Bernard Merindol)
 * ASTERISK-28758 - pjsip startup errors when using "with-ssl"
      configure option
      (Reported by Patrick Wakano)
 * ASTERISK-28824 - BuildSystem: Search for Python/C API when
      possibly needed only.
      (Reported by Alexander Traud)
 * ASTERISK-27717 - [patch] BuildSystem: In NetBSD, the Python
      Programming Language is python-2.7.
      (Reported by Alexander
 * ASTERISK-28817 - chan_pjsip: constant DTMF tone if RTP is not
      setup yet
      (Reported by Kevin Harwell)
 * ASTERISK-28819 - [patch] bridge_softmix_binaural: Show state
      in menuselect.
      (Reported by Alexander Traud)
 * ASTERISK-28816 - [patch] BuildSystem: Remove doc/tex and
      doc/pdf leftovers.
      (Reported by Alexander Traud)
 * ASTERISK-28818 - [patch] BuildSystem: Allow space in path.
      (Reported by Alexander Traud)
 * ASTERISK-28809 - [patch] res_rtp_asterisk: Avoid absolute
      value on unsigned subtraction.
      (Reported by Alexander
 * ASTERISK-28796 - func_channel: cannot read fields exten,
      context, userfield, channame from dialplan
      (Reported by
      S��bastien Duthil)
 * ASTERISK-28803 - [patch] chan_unistim: Avoid tautological
      warnings with clang.
      (Reported by Alexander Traud)
 * ASTERISK-28808 - [patch] test_stasis: Avoid always true
      warning with clang.
      (Reported by Alexander Traud)
 * ASTERISK-28056 - res_pjsip: Incorrect endpoint status after
      endpoint synchronization for a specific AOR
      (Reported by
      Jason Hord)
 * ASTERISK-28795 - channel: write to a stream on multi-frame
      (Reported by Kevin Harwell)
 * ASTERISK-28789 - test_utils: incorrectly printing error
      'declined to load'
      (Reported by Alexander Traud)
 * ASTERISK-28788 - func_aes: incorrectly printing error
      'declined to load'
      (Reported by Alexander Traud)
 * ASTERISK-28790 - Crash during conference call using
      confbridge and video
      (Reported by Pascal Cadotte Michaud)
 * ASTERISK-16676 - DAHDIRAS fails to properly initiate pppd
      unless asterisk is running as root
      (Reported by Jaco
 * ASTERISK-21205 - [patch] dundi_read_result crash due to
      negative number
      (Reported by Jaco Kroon)
 * ASTERISK-28784 - res_pjsip_sdp_rtp: Only do hold/unhold on
      first audio stream
      (Reported by Joshua C. Colp)
 * ASTERISK-28743 - Asterisk is crashing if the 200 OK with SDP

      (Reported by sungtae kim)
 * ASTERISK-28783 - res_pjsip_session: Allow default non-audio
      streams to have reflected state
      (Reported by Joshua C.
 * ASTERISK-28774 - chan_pjsip's rtptimeout is erroneously
      triggered during direct-media (native_rtp) bridge
      (Reported by Michael Neuhauser)
 * ASTERISK-20325 - Comments in configs/func_odbc.conf.sample
      are not consistent with examples. Missing examples.
      (Reported by Olivier Krief)
 * ASTERISK-28780 - app_mixmonitor: Memory leak due to race
      condition between AMI MixMonitor and hangup
      (Reported by
      Joshua C. Colp)
 * ASTERISK-28773 - Incorrect Sender SSRC in RTCP when p2p rtp
      bridge is active
      (Reported by Torrey Searle)
 * ASTERISK-28769 - DTLS Handshake Fails to Occur if ice_support
      is enabled but not used
      (Reported by Torrey Searle)
 * ASTERISK-28759 - A non negotiated rtp frame causes call
      disconnection when there is a SSRC change
      (Reported by
      Paulo Vicentini)
 * ASTERISK-26711 - func_enum: ENUM code wrong case
      (Reported by Vitold)
 * ASTERISK-23407 - Fix the FSF address in the headers of lots
      of pjproject files
      (Reported by Jared Smith)
 * ASTERISK-19460 - [patch] Function TXTCIDNAME never actually
      makes DNS calls and always returns an empty string
      (Reported by George Joseph)
 * ASTERISK-28766 - PJSIP blind transfer not completed after
      using Proceeding()
      (Reported by laszlovl)
 * ASTERISK-28764 - res_rtp_asterisk: Improve NACK support and
      seqno handling
      (Reported by Joshua C. Colp)
 * ASTERISK-28755 - SIP/Stasis: SIP headers not transmitted in
      the "variables" field
      (Reported by Jean Aunis - Prescom)
 * ASTERISK-28685 - check_expr2: linking (when hardening) and
      cross-compiling troubles
      (Reported by Sebastian Kemper)
 * ASTERISK-28754 - ASTERISK-28738 Causes Audio Issue After
      (Reported by Ross Beer)
 * ASTERISK-28697 - res_pjsip: Named ACL does not update on
      reload if changed
      (Reported by Timothy Vanderaerden)
 * ASTERISK-28746 - res_pjsip_outbound_registration keeps
      retrying the first entry in a SRV record set
      (Reported by
      George Joseph)
 * ASTERISK-28716 - ICE: pjnath shouldn't wait for ICE to
      complete before allowing sending
      (Reported by Benjamin
      Keith Ford)
 * ASTERISK-28738 - Incorrect state machine used when
      MOH_PASSTHRU is used
      (Reported by Torrey Searle)
 * ASTERISK-28742 - res_rtp_asterisk: static for audio due to
      incomplete dtls/srtp setup
      (Reported by Kevin Harwell)
 * ASTERISK-28735 - Realtime MoH Unknown format '' -- defaulting
      to SLIN
      (Reported by Ross Beer)
 * ASTERISK-28730 - res_pjsip_session: Fix out of order session
      (Reported by Joshua C. Colp)
 * ASTERISK-26955 - pjsip: SIP Packets with Via "received="
      Containing IPv6 Address Delimited by "[]" Rejected
      (Reported by Peter Sokolov)
 * ASTERISK-28718 - chan_sip: Returns 403 if RTP ports are
      depleted, should return 503
      (Reported by Walter Doekes)
 * ASTERISK-28713 - res_stasis_playback: Error building JSON
      (Reported by S��bastien Duthil)
 * ASTERISK-28714 - REGRESSION: Feature
      subscription_persistence_recreate (ASTERISK-27759) Causes
      (Reported by Ross Beer)
 * ASTERISK-26082 - res_pjsip_messaging: MessageSend
      Content-Type can't be changed
      (Reported by Alex)
 * ASTERISK-28423 - ARI causes STASIS Deadlock
      by Ross Beer)
 * ASTERISK-28679 - stasis application is destroyed after its
      (Reported by Francois Blackburn)
      spite of the error when sending
      (Reported by Dmitriy
 * ASTERISK-28686 - chan_sip strictrtp=yes fails when media
      source is changed: no audio
      (Reported by Walter Doekes)
 * ASTERISK-28139 - RTP Stream Incorrect Payload Type Causes
      Asterisk To Drop Calls
      (Reported by Paul Brooks)
 * ASTERISK-28677 - CDR billsec is always 0 for transferred
      (Reported by Maciej Michno)
 * ASTERISK-28702 - chan_dahdi: holding a channel via flash to
      dialtone times out after 0:16:40
      (Reported by Andrew
 * ASTERISK-24484 - Update documentation for statsd module -
      usage requirements unclear
      (Reported by Dan Jenkins)
 * ASTERISK-28706 - silk 24hHz doesn't show up in 'core show
      translation' output
      (Reported by Sean Bright)
 * ASTERISK-28695 - core: minmemfree watermark uses free RAM,
      not available RAM
      (Reported by Kevin Flyn)
 * ASTERISK-28693 - chan_sip: SIP MESSAGE beginning with a
      whitespace appears empty in the dialplan
      (Reported by
      Frank Matano)
 * ASTERISK-23739 - [patch]Segfault forwarding voicemail with
      ODBC storage enabled and realtime voicemail_data is used
      (Reported by Stas Kobzar)
 * ASTERISK-27622 - empty voicemail.conf required for ARA
      (realtime) voicemail to leave message
      (Reported by Jim Van
 * ASTERISK-21794 - CLI command 'realtime update2' syntax
      failure when using according to usage help
      (Reported by
      Cedric BASSAGET)
 * ASTERISK-28349 - Pause reason not reported in QueueMember AMI
      (Reported by Niksa Baldun)
 * ASTERISK-25429 - res_pjsip_endpoint_identifier_ip: Document
      support for hostnames
      (Reported by Joshua C. Colp)
 * ASTERISK-27775 - res_pjsip_notify: Multiple Event headers can
      be present instead of just one
      (Reported by
 * ASTERISK-28682 - app_record: Lack of `beep` audio file causes
      application to return error and hangup
      (Reported by Corey
 * ASTERISK-28507 - Wiki docs missing for MessageWaiting
      (Reported by David M. Lee)
 * ASTERISK-27759 - res_pjsip_pubsub: Subscription persistence
      does not preserve XML <dialog-info> version number
      (Reported by Bryan Nelson)
 * ASTERISK-28605 - chan_dahdi: Deadlock in Hangup Scenarios
      with concurrent command pri show span X
      (Reported by Dirk
 * ASTERISK-28633 - stasis bridge topic leak
      (Reported by
      Joeran Vinzens)
 * ASTERISK-28492 - pjsip reload not reloading wizard
      endpoint/pickup_group endpoint/call_group
      (Reported by
      Jean-Denis Girard)
 * ASTERISK-28562 - SIP WSS message not processed until next
      frame arrives
      (Reported by Robert Sutton)
 * ASTERISK-28667 - Asterisk ignores parsing of config files if
      a Byte order mark is present
      (Reported by Robin Leffmann)
 * ASTERISK-28625 - Playback of local files impacted by large
      media cache
      (Reported by Kevin Reeves)
 * ASTERISK-27243 - contrib: valgrind.supp doesn't suppress what
      it's supposed to due to invalid syntax
      (Reported by
      Richard Kenner)
 * ASTERISK-28664 - "trustrpid" is misspelled in
      (Reported by Pascal Cadotte Michaud)
 * ASTERISK-28636 - app_chanisavail+cdr: ChanIsAvail sometimes
      fails to deactivate CDR.
      (Reported by Frederic LE FOLL)
 * ASTERISK-28604 - app_meetme, chan_ooh323 and cdr_mysql don't
      build on 17.0.0
      (Reported by George Joseph)
 * ASTERISK-28659 - res_pjsip_sdp_rtp: Bundle includes
      non-existent media stream if codecs create additional streams
      and offer does not have them
      (Reported by nappsoft)
 * ASTERISK-28660 - res_fax: wrap Asterisk initiated negotiation
      with config option
      (Reported by Kevin Harwell)
 * ASTERISK-28626 - Missing arguments in PJSIP_CONTACT function
      (Reported by Pascal Cadotte Michaud)
 * ASTERISK-28609 - Memory Leak in res_rtp_asterisk.c
      (Reported by Ted G)
 * ASTERISK-28651 - chan_sip logs errors on tx to non-existent
      TCP connections
      (Reported by Jaco Kroon)
 * ASTERISK-28502 - chan_pjsip incorrectly re-writes REGISTER
      200 Response Contact
      (Reported by Ross Beer)
 * ASTERISK-28641 - res_pjsip Segfaults when realtime
      configuration to an AOR points to a not existent AOR
      (Reported by Ross Beer)
 * ASTERISK-28647 - chan_sip: RTP frames not transmitted after
      emitting a COLP
      (Reported by Jean Aunis - Prescom)
 * ASTERISK-28637 - chan_sip+native_bridge_rtp: directmedia
      compatibility check failure when negociated ptime is not default
      (Reported by Frederic LE FOLL)
 * ASTERISK-28445 - res_pjsip_session: ast_json_vpack: Invalid
      UTF-8 string on hangup when TEST_FRAMEWORK enabled
      (Reported by Bernhard Schmidt)
 * ASTERISK-28631 - res_parking: Doesn't park when parkee and
      parker are the same
      (Reported by Ross Beer)
 * ASTERISK-28621 - Enforce T.38 error correction mode at 200 ok
      (Reported by Salah Ahmed)
 * ASTERISK-28624 - res_pjsip_outbound_registration: add SRV
      (Reported by Kevin Harwell)
 * ASTERISK-28608 - app_amd: Use time calculation to calculate
      (Reported by Michael Cargile)
 * ASTERISK-28615 - chan_dahdi: PRI span status may stay "Down,
      Active" after a short alarm
      (Reported by Frederic LE FOLL)
 * ASTERISK-28576 - res_rtp_asterisk: ICE Completion Crash when
      sent packet length doesn't match
      (Reported by Joshua
 * ASTERISK-26481 - FILE function grabs garbage along with read
      data when target line has no newline
      (Reported by Jonathan
 * ASTERISK-28618 - bridge_softmix: hold not cleared when
      joining a softmix bridge
      (Reported by Kevin Harwell)
 * ASTERISK-28616 - parking: Deadlock when multi call parking
      (Reported by Joshua C. Colp)
 * ASTERISK-28572 - Memory leaks in res_calendar_exchange and
      (Reported by Yoooooo Ha)
 * ASTERISK-28585 - ari/resource_events: Crash in event session
      (Reported by Kevin Harwell)
 * ASTERISK-28590 - utils.c throws repeated warnings;
      "pthread_attr_setstacksize: Invalid argument"
      (Reported by
      Speed Dial Dave)
 * ASTERISK-28578 - race condition on pjsip channelstats
      (Reported by Salah Ahmed)
 * ASTERISK-28571 - cdr_pgsql: accesses obsolete (and finally
      removed) column
      (Reported by Christoph Moench-Tegeder)
 * ASTERISK-28575 - MWI Send Notify Crash on 16.6
      (Reported by Joshua Elson)
 * ASTERISK-28574 - pjproject fails to build on 16.6.0, works on
      (Reported by Niklas Larsson)
 * ASTERISK-28561 - Asterisk Deadlocks
      (Reported by
 * ASTERISK-28086 - chan_pjsip: Crash when initiating PlayDTMF
      over AMI
      (Reported by Jeremiah Gadd)
 * ASTERISK-28552 - res_pjsip_mwi: Frack during unload on
      unsolicited_mwi container
      (Reported by Kevin Harwell)
 * ASTERISK-28566 - CDR backend unload problem during active
      (Reported by Marian Piater)
 * ASTERISK-28553 - stasis.c: Crash during unload
      (Reported by Kevin Harwell)
 * ASTERISK-28544 - Wrong contact representation in ipv6 mode
      (Reported by J��rgen H)
 * ASTERISK-28534 - Segmentation fault when there is no priority
      for an extension
      (Reported by Timothy Vanderaerden)
 * ASTERISK-28463 - res_pjsip_path: Crash when invalid contact
      is configured
      (Reported by Juan Martin)
 * ASTERISK-28521 - pjsip: Memory Leak
      (Reported by Mark)
 * ASTERISK-28523 - Asterisk 16.5.0 Memory leak
      by Cyril Rami��re)
 * ASTERISK-28536 - Asterisk release candidates fail to build on
      (Reported by Guido Falsi)
 * ASTERISK-28538 - chan_pjsip: Deadlock on fax detection
      (Reported by Joshua C. Colp)
 * ASTERISK-28497 - func_odbc: truncating Unicode string on
      (Reported by Boris P. Korzun)
 * ASTERISK-23756 - setvar directive when used in template and a
      child of said template, results in duplicate variable names
      (Reported by Michael Goryainov)
 * ASTERISK-28527 - ChanIsAvail() creates a CDR if
      unanswered=yes is set in cdr.conf
      (Reported by Frederic LE
 * ASTERISK-28525 - chan_dahdi: set CHANNEL(hangupsource) when a
      PRI channel hangs up
      (Reported by Frederic LE FOLL)
 * ASTERISK-28511 - codec_resample: Bad sound quality when up
      sampling from SLIN16 to SLIN32
      (Reported by Ruddy G)
 * ASTERISK-28499 - translate: Crash when frame does not have a
      "src" field set
      (Reported by Gregory Massel)
 * ASTERISK-25592 - chan_unistim: Clang Warning: variable sized
      type not at end of a struct
      (Reported by Alexander Traud)
 * ASTERISK-28488 - pjsip mwi: n+1 sip notify's sent on
      (Reported by Chris Savinovich)
 * ASTERISK-28509 - PJSIP cnonce generated on Linux contains 36
      characters, NEC only supports up to 32 characters
      (Reported by Dan Cropp)
 * ASTERISK-28505 - app_voicemail/IMAP: segfault in
      leave_voicemail because not checking mailstream
      by Alexei Gradinari)
 * ASTERISK-28487 - compile menuselect on gentoo
      by Kilburn)
 * ASTERISK-28472 - Asterisk occasionally passes a NULL as
      srtp->session to srtp_protect/unprotect causing SEGV
      (Reported by Jonas Swiatek)
 * ASTERISK-28498 - cel / cdr: Event times may be incorrect
      (Reported by Joshua C. Colp)
 * ASTERISK-28480 - json integer overflow in ssrc and timestamp

      (Reported by Salah Ahmed)
 * ASTERISK-28228 - res_pjsip: pjsip show contacts prints double
      (Reported by Ian Jones)
 * ASTERISK-28483 - packet lost on UDPTL wrap around
      (Reported by Torrey Searle)
 * ASTERISK-28477 - Crash when not specifying "dbfile" in
      (Reported by Dennis)
 * ASTERISK-28478 - Crash performing "core reload" with modified
      (Reported by Dennis)
 * ASTERISK-28282 - AST_SCHED_REPLACE_UNREF causes wait-on-self
      deadlocks (in chan_sip)
      (Reported by Walter Doekes)
 * ASTERISK-27121 - res_pjsip_mwi: Memory leak on reload
      (Reported by Sergej Kasumovic)
 * ASTERISK-28457 - [patch] Fix crash in chan_dahdi on 32-bit
      systems caused by ASTERISK-28317
      (Reported by abelbeck)
 * ASTERISK-28458 - res_pjsip_sdp_rtp: Remove unused variable
      (Reported by Michael Maier)
 * ASTERISK-26006 - Show offending IP for TLS setup failures in
      (Reported by Oleksandr Natalenko)
 * ASTERISK-28444 - chan_pjsip: Peer IP for SSL handshake errors
      not logged
      (Reported by Bernhard Schmidt)
 * ASTERISK-26968 - chan_pjsip: Transfer() does not result in
      TRANSFERSTATUS reflecting SIP response to transfer
      (Reported by Dan Cropp)
 * ASTERISK-28419 - app_amd: Does not work with silence
      (Reported by Nasir Iqbal)
 * ASTERISK-28018 - IP Fragmentation happening instead of DTLS
      fragmentation on handshake server hello certificate
      (Reported by vijay kumar)
 * ASTERISK-25371 - Crash in hangup at chan_pjsip.c:1749 when
      Asterisk attempts to generate hangup event
      (Reported by
      Abhay Gupta)
 * ASTERISK-28435 - cdr_pgsql: Unix socket doesn't work
      (Reported by Dmitry Svyatogorov)
 * ASTERISK-27981 - res_fax: Fax session leak with fax
      (Reported by pasandev)
 * ASTERISK-28427 - new mwi.h include missing from some dahdi
      source files, causes build failure
      (Reported by Guido
 * ASTERISK-28421 - Wrong type used for timestamp in
      (Reported by Morten Tryfoss)
 * ASTERISK-28161 - Removal of Previous Patch Causes PJSIP Timer
      (Reported by Ross Beer)
 * ASTERISK-27994 - PJSIP: Early media ringback not indicated
      after Progress()
      (Reported by Gregory Massel)
 * ASTERISK-28412 - GCC 9 catches more string formatting issues

      (Reported by George Joseph)
 * ASTERISK-28379 - pjsip: show channelstats incorrect
      information output
      (Reported by Vyrva Igor)
 * ASTERISK-28399 - channel.c: Exceptionally long queue length
      (Reported by Abhay Gupta)
 * ASTERISK-28392 - The no-partial-inlining flag isn't passed to
      the bundled pjproject or jansson builds
      (Reported by
      George Joseph)
 * ASTERISK-28402 - res_pjsip_registrar: SEGV in
      (Reported by Ross Beer)
 * ASTERISK-27756 - bridge: Failure to impart a channel results
      in bad data causing crash
      (Reported by Abhay Gupta)
 * ASTERISK-26718 - ARI: Bridge destroying doesn't work as
      (Reported by Marin Odrljin)
 * ASTERISK-28143 - app_amd: Infinite loop on silent calls 
      (Reported by Abhay Gupta)
 * ASTERISK-28353 - stasis: Crash at shutdown when statistics
      (Reported by Joshua C. Colp)
 * ASTERISK-28374 - latest asterisk unconditionally launch gcc
      --version, even if the compiler is different
      (Reported by
      Guido Falsi)
 * ASTERISK-28391 - res_indications: Crash requesting
      autocomplete on indications cli command
      (Reported by Lucas
 * ASTERISK-27935 - app_voicemail: emailbody per user can't
      contain commas
      (Reported by S��bastien Duthil)
 * ASTERISK-17695 - extenpatternmatchnew=yes cannot find
      extensions with '-' in them
      (Reported by test011)
 * ASTERISK-17799 - AEL reload causes loss of control in a
      (Reported by Kirill Katsnelson)
 * ASTERISK-18593 - AEL for loops use Macro app and pipe
      (Reported by Luke-Jr)
 * ASTERISK-14939 - AEL parsers does not find existing label
      (Reported by klaus3000)
 * ASTERISK-20182 - Parsing a label beginning with a numeric
      character in all Goto/GotoIf/GotoIfTime application causes
      unexpected behavior
      (Reported by Janu)
 * ASTERISK-28348 - Failed to initialize OOH323 endpoint-OOH323
      (Reported by Dmitry Shubin)
 * ASTERISK-28371 - chan_pjsip: DTMF Mode auto_info fallback
      lead to both inband and info
      (Reported by Salah Ahmed)
 * ASTERISK-28319 - musl: Crash on startup when loading modules

      (Reported by Sebastian Kemper)
 * ASTERISK-28362 - strtok_r() makes gcc compile warning
      (Reported by sungtae kim)
 * ASTERISK-28255 - res_rtp_asterisk: REMB RTCP packet sending
      may be incorrect
      (Reported by Joshua C. Colp)
 * ASTERISK-27541 - app_queue: Queue paused reason was (big
      number) secs ago when reason is set
      (Reported by C��sar
      Benjam��n Garc��a Mart��nez)
 * ASTERISK-20986 - QUEUE_MEMBER 's description is inaccurate
      (Reported by Olivier Krief)
 * ASTERISK-28350 - manager: Stasis backed up due to locking
      (Reported by Joshua C. Colp)
 * ASTERISK-25792 - chan_sip: qualifygap bounds checking
      (Reported by Paul Sandys)
 * ASTERISK-28341 - res_config_odbc eliminates empty custom (���@���
      prefix) variables 
      (Reported by Alexei Gradinari)
 * ASTERISK-28333 - StasisEnd event makes wrong timestamp value

      (Reported by sungtae kim)
 * ASTERISK-28306 - res_pjsip_mwi: MWI NOTIFY occasionally takes
      minutes to be sent
      (Reported by Jared Hull)
 * ASTERISK-28332 - Variable ALTCONF ignored when service is
      used in Debian
      (Reported by Cirillo Ferreira)
 * ASTERISK-27964 - app_queue: ring_entry accesses nativeformats
      without channel lock or reference
      (Reported by Francisco
 * ASTERISK-28335 - stasis: Make topic and maybe subscription
      names unique and more useful
      (Reported by Joshua C. Colp)
 * ASTERISK-28321 - res_rtp_asterisk: Fixing possible divide by
      zero for rtcp stat calculation
      (Reported by sungtae kim)
 * ASTERISK-28322 - chan_pjsip: Add option to allow ignoring of
      183 without SDP
      (Reported by Torrey Searle)
 * ASTERISK-28328 - MeetMe global non-admin mute is muting
      admins that subsequently join
      (Reported by Philip Mott)
 * ASTERISK-28168 - app_queue: Adding a blank entry into sql
      queue_members crashes asterisk.
      (Reported by Michael)
 * ASTERISK-28323 - pjsip: sip.conf to pjsip.conf conversion
      script fails
      (Reported by Guido Weckwerth)
 * ASTERISK-28272 - The basic-pbx config samples don't produce a
      running asterisk
      (Reported by George Joseph)
 * ASTERISK-28312 - res_pjsip_diversion: Corrupted SIP Diversion
      field after handling a 302 redirect
      (Reported by Alex
 * ASTERISK-24173 - File menuselect/menuselect_gtk.c has no
      license header
      (Reported by Jeremy Lain��)
 * ASTERISK-28166 - app_voicemail: Asterisk unresponsive after
      changing voicemail password with ODBC
      (Reported by
 * ASTERISK-28309 - res_pjsip: Wrong Contact and Via fields with
      multiple UDP interfaces
      (Reported by Nikolay shakin)
 * ASTERISK-27992 - PJSIP: Adding `sends_registrations = yes` to
      pjsip_wizard.conf  causes crash
      (Reported by Jonathan
 * ASTERISK-28213 - res_pjsip: Threads pile up needlessly when
      AOR is blocked
      (Reported by Ross Beer)
 * ASTERISK-28301 - Allow voicemail boxes to be subscribed to
      with a presence event package
      (Reported by George Joseph)
 * ASTERISK-28303 - res_rtp_asterisk: Interaction between
      smoother and DTMF can cause out of order timestamps
      (Reported by Torrey Searle)
 * ASTERISK-28302 - ARI: "Error destroying mutex" when listing
      all ARI applications
      (Reported by Stefan Repke)
 * ASTERISK-28300 - AST_PBX_MAX_STACK is too low for some
      (Reported by George Joseph)
 * ASTERISK-28106 - Astricon Feedback: Unable to filter ARI
      events when GETting causes overload of events
      (Reported by
      George Joseph)
 * ASTERISK-28284 - switching between native_bridge and
      simple_bridge can cause one way audio
      (Reported by Torrey
 * ASTERISK-28251 - CI: Fix CI so it reverifies commit message
      (Reported by George Joseph)
 * ASTERISK-28277 - database: Add some basic logging
      (Reported by Joshua C. Colp)
 * ASTERISK-28181 - ari: Originating overwrites channel start
      (Reported by sungtae kim)
 * ASTERISK-28173 - Deadlock in chan_sip handling subscribe
      request during res_parking reload
      (Reported by Giuseppe
 * ASTERISK-28104 - AstriCon Feedback:  Automatically create a 1
      line dialplan context for stasis apps
      (Reported by George
 * ASTERISK-28271 - Opensuse Leap 15 --with-jannson-bundled will
      not compile
      (Reported by David Wilcox)
 * ASTERISK-28238 - PJSIP realtime. getcontext not working with
      (Reported by Ray)
 * ASTERISK-28263 - codec_opus: errors setting max_playback_rate
      and bitrate to "sdp"
      (Reported by Gianluca Merlo)
 * ASTERISK-28257 - res_http_websocket: PING / PONG opcodes
      break data reception
      (Reported by Jeremy Lain��)
 * ASTERISK-28250 - build: Cross-compilation fails for target
      (Reported by Jean Aunis - Prescom)
 * ASTERISK-28252 - HangupHandler manager events are never
      (Reported by Gerald Schnabel)
 * ASTERISK-28231 - res_http_websocket: Not responding to
      Connection Close Frame (opcode 8)
      (Reported by Jeremy
 * ASTERISK-28249 - res_monitor: Segfault with
      (Reported by Valentin Vidi��)
 * ASTERISK-28244 - stasis: Filter messages at publishing to
      (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-28230 - res_rtp_asterisk: abs-send-time extension
      added with Asterisk 15.5.0 breaks GXV3140 video telephony
      (Reported by David Kuehling)
 * ASTERISK-28232 - core: RAII using clang use-after-scope
      (Reported by Diederik de Groot)
 * 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
      (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
      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-28157 - Asterisk crashes when the res_pjsip_*
      modules unload
      (Reported by sungtae kim)
 * ASTERISK-28125 - app_queue: Revert broken queue channel
      reference patch
      (Reported by laszlovl)
 * ASTERISK-27095 - chan_pjsip: When connected_line_method is
      set to invite, we're not trying UPDATE
      (Reported by George
 * ASTERISK-28182 - chan_pjsip: When connected_line_method is
      set to invite, asterisk is not trying UPDATE
      (Reported by
 * ASTERISK-28151 - app_voicemail: MWI fails with
      mailboxes=##@device instead of mailboxes=##@default
      (Reported by Ronald Raikes)
 * ASTERISK-28119 - stasis: Segment channel snapshot to reduce
      creation cost
      (Reported by Joshua C. Colp)
 * ASTERISK-28102 - stasis: Use implementation specific cache
      for channel snapshots
      (Reported by Joshua C. Colp)
 * ASTERISK-28159 - SIGABRT caused by stack corruption in
      hashkeys_read when no matching keys present
      (Reported by
      Michael Walton)
 * ASTERISK-28140 - repeated segmentation faults 
      (Reported by Eyal Hasson)
 * ASTERISK-28103 - stasis: Filter messages at publishing to
      reduce work done
      (Reported by Joshua C. Colp)
 * ASTERISK-28169 - ARI /channels/create handler causes core
      (Reported by sungtae kim)
 * ASTERISK-28129 - Incorrect Behavior for rewrite_contact when
      Re-Invite omits routset
      (Reported by Torrey Searle)
 * ASTERISK-28158 - Some conditions prevent running of el_end,
      break the terminal.
      (Reported by Corey Farrell)
 * ASTERISK-28110 - rtp: Incorrect Packetization
      by Robert Cripps)
 * ASTERISK-28146 - pbx_config: Only the first [globals] section
      is processed.
      (Reported by Corey Farrell)
 * ASTERISK-28150 - Formatting error in documentation
      (Reported by Scott Griepentrog)
 * ASTERISK-28081 - chan_sip: Asterisk 12+ chan_sip doesn't
      report AST_CEL_PICKUP in handle_invite_replaces
      by Luit van Drongelen)
 * ASTERISK-28137 - res_pjsip_notify: improve realtime
      performance on CLI completion on the endpoint
      (Reported by
      Alexei Gradinari)
 * ASTERISK-27980 - Caller ID cannot be changed on Attended
      Transfer before dialing out
      (Reported by Alexei Gradinari)
 * ASTERISK-28107 - app_confbridge:  Participant info labels
      aren't being added to the SDPs
      (Reported by George Joseph)
 * ASTERISK-28089 - function ast_sendtext() create RTP realtime
      packets with a trailing null byte in the payload
      by Emmanuel BUU)
 * ASTERISK-28076 - bridging: Asterisk crashes when receiving an
      empty realtime text frame
      (Reported by Emmanuel BUU)
 * ASTERISK-28084 - app_queue: QueueMemberStatus Event flooding
      (Reported by Andrej)
 * ASTERISK-28077 - res_pjsip: improve realtime performance on
      CLI 'pjsip show contacts'
      (Reported by Alexei Gradinari)
 * ASTERISK-27920 - app_queue: Queue member considered inuse
      after immediately hanging up during dialing.
      (Reported by
      Cao Minh Hiep)
 * ASTERISK-26094 - stasis: Playing MOH to bridge with ARI does
      not work
      (Reported by Cameron)
 * ASTERISK-28065 - res_odbc: missing SQL error diagnostic
      (Reported by Alexei Gradinari)
 * ASTERISK-28057 - chan_sip: SipNotify via AMI behaves
      differently to CLI
      (Reported by Peter Katzmann)
 * ASTERISK-28045 - configure script does not enforce
      libunbound2 version
      (Reported by Samuel Galarneau)
 * ASTERISK-28070 - testsuite: Sniffer assumes pjmedia will use
      ports below 10000
      (Reported by Joshua C. Colp)
 * ASTERISK-27854 - rtp: Crash in off-nominal case where RTP
      instance can't be set up
      (Reported by Lei Fu)
 * ASTERISK-28034 - chan_sip unstable with TLS after asterisk
      start or reloads
      (Reported by David Hajek)
 * ASTERISK-28059 - PJSIP: Update bundled PJPROJECT to version
      (Reported by Joshua C. Colp)
 * ASTERISK-28047 - chan_pjsip: Declined video stream is added
      when no video codecs configured and session refresh with removed
      video stream occurs
      (Reported by Will)
 * ASTERISK-28033 - AMI event "NewExten" is set to the wrong
      (Reported by laszlovl)
 * ASTERISK-28049 - res_pjproject build failure
      by Jaco Kroon)
 * ASTERISK-28029 - [patch] res_musiconhold : music on hold will
      not start if previous hold just reached end of file
      (Reported by Frederic LE FOLL)
 * ASTERISK-28005 - channel.c: ARI ring only once
      (Reported by Hajek Michal)
 * ASTERISK-28032 - Realtime queuemembers are not updated during
      retry phase
      (Reported by laszlovl)
 * ASTERISK-27988 - alembic: PJSIP
      "mwi_subscribe_replaces_unsolicited" field is integer not
      (Reported by Joshua C. Colp)
 * ASTERISK-28020 - res_pjsip_transport_websocket: Properly set
      'received' for IPv6
      (Reported by Sean Bright)
 * ASTERISK-28002 - When T.140 realtime text is negociated, a
      lot of debug traces are generated
      (Reported by Emmanuel
 * ASTERISK-27881 - PBX calls via chan_sip TCP trunk now get
      authentification error
      (Reported by Ian Gilmour)
 * ASTERISK-28022 - res_pjsip realtime: uri column in
      ps_contacts table can be too short
      (Reported by Florian
 * ASTERISK-27944 - res_pjsip_t38: Crash receiving 1xx responses
      other than 100 before 200 for T.38 reINVITE
      (Reported by
      Joshua Elson)
 * ASTERISK-28007 - rtcp-mux is put in SDP answer regardless of
      (Reported by Torrey Searle)
 * ASTERISK-27398 - No joint capabilities with video and
      audio-only streams
      (Reported by Benjamin Keith Ford)
 * ASTERISK-27973 - app_queue: QUEUESTATUS = CONTINUE instead
      (Reported by Valentin Safonov)
 * ASTERISK-27997 - pjproject_bundled: Fix for Solaris builds.
      Do not undef s_addr.
      (Reported by Alexander Traud)
 * ASTERISK-27999 - Wrong SRTP use status report
      by Salah Ahmed)
 * ASTERISK-28001 - res_pjsip_registrar: Improve performance of
      inbound handling
      (Reported by Joshua C. Colp)
 * ASTERISK-27966 - pjsip: Race condition in 183 re transmission
      can result in a deadlock
      (Reported by Torrey Searle)
 * ASTERISK-15331 - make menuselect fails due to undefined
      symbols (initscr32, w32addch) in menuselect_curses.o
      (Reported by Majdi Bsoul)
 * ASTERISK-14935 - [regression] menuselect compilation failure
      on Solaris 10
      (Reported by Samuel Owens)
 * ASTERISK-12382 - menuselect compilation failure on Solaris 10
      / gcc 3.4.3
      (Reported by rleasure)
 * ASTERISK-9107 - menuselect compilation failure on Solaris
      (Reported by Bob Atkins)
 * ASTERISK-27991 - BuildSystem: Enable Jansson in Solaris 11.
      (Reported by Alexander Traud)
 * ASTERISK-27548 - res_pjsip_endpoint_identifier_ip only
      matches against "generic string" headers
      (Reported by
      George Joseph)
 * ASTERISK-27990 - res_rtp_asterisk: Requires OpenSSL in
      Developer Mode.
      (Reported by Alexander Traud)
 * ASTERISK-27591 - Frack errors in stasis.c and memory leakage

      (Reported by Siruja Maharjan)
 * ASTERISK-27978 - res_pjsip: Change default transport
      keepalive to preserve behavior
      (Reported by Joshua C.
 * ASTERISK-27968 - systemd: asterisk.service
      (Reported by

Improvements made in this release:
 * ASTERISK-29777 - documentation: Standardize example syntax
      (Reported by N A)
 * ASTERISK-29715 - app_voicemail: Refactor email generation
      (Reported by N A)
 * ASTERISK-29727 - Add type for JSON stasis message RTCP Report
      (Reported by Boris P. Korzun)
 * ASTERISK-29714 - Spelling errors
      (Reported by Josh
 * 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
      (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
      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
      (Reported by N A)
 * ASTERISK-29528 - Add support for multiple files for agent
      (Reported by N A)
 * ASTERISK-29501 - ARI - Stasis Playback doesn't hangup call
      when processing a list of invalid files
      (Reported by Andre
 * ASTERISK-29464 - ARI - PlaybackFinish skip error events
      (Reported by Andre Barbosa)
 * ASTERISK-29450 - Allow setting channel variables using
      Originate application
      (Reported by N A)
 * ASTERISK-29459 - Missing configuration from PJSIP to SIP
      conversion script
      (Reported by N A)
 * ASTERISK-29460 - Recognize application/hook-flash in PJSIP
      (Reported by N A)
 * ASTERISK-29434 - Asterisk reveals pjproject version in STUN
      (Reported by Jeremy Lain��)
 * ASTERISK-29349 - Silent voicemail option is not completely
      (Reported by N A)
 * ASTERISK-29380 - Add Flash AMI event to handle flash events
      (Reported by N A)
 * ASTERISK-29339 - loader: Let's output warnings for deprecated
      (Reported by Joshua C. Colp)
 * ASTERISK-29337 - menuselect: Add ability to set deprecated in
      and removed in versions for modules
      (Reported by Joshua C.
 * ASTERISK-29336 - documentation: Fix inconsistent support
      (Reported by Joshua C. Colp)
 * ASTERISK-29335 - xml: Embed module information into core XML
      (Reported by Joshua C. Colp)
 * ASTERISK-29321 - sorcery: Add support for more intelligent
      (Reported by Joshua C. Colp)
 * ASTERISK-29325 - res_pjsip_registrar: Include source IP
      address and port in log messages
      (Reported by Joshua C.
 * ASTERISK-29326 - asterisk: Update copyright/company
      (Reported by Joshua C. Colp)
 * ASTERISK-29244 - Add MixMonitorStart / Stop / Mute AMI
      (Reported by S��bastien Duthil)
 * ASTERISK-29252 - TRANSFERSTATUSPROTOCOL variable to report
      Transfer (REFER) failure SIP code
      (Reported by Dan Cropp)
 * ASTERISK-29262 - Support of various URL-schemes by MoH
      (Reported by Boris P. Korzun)
 * ASTERISK-28549 - Two repeated 183
      (Reported by Gant
 * ASTERISK-29216 - contrib: systemd asterisk service for
      centos8 or other newer linux versions
      (Reported by Mark
 * ASTERISK-29143 - res_http_media_cache: HTTP media cache
      stored hardcoded in /tmp
      (Reported by laszlovl)
 * ASTERISK-29118 - VoiceMail() should have an option to play
      greetings as Early Media
      (Reported by Juan Carlos Castro y
 * ASTERISK-29054 - Logger: Add debug logging categories
      (Reported by Kevin Harwell)
 * ASTERISK-29056 - Increase reg_server column size for
      ps_contacts table realtime
      (Reported by sungtae kim)
 * ASTERISK-29055 - Create a Bridge with video_single mode
      (Reported by sungtae kim)
 * ASTERISK-28959 - res_pjsip: Added option for disable rport
      parameter set
      (Reported by sungtae kim)
 * ASTERISK-28958 - Continue reading string when ping received
      by websocket
      (Reported by Nickolay V. Shmyrev)
 * ASTERISK-28945 - AMI SendText - add Content-Type parameter
      (Reported by Kevin Harwell)
 * ASTERISK-28949 - res_http_websocket: Add masking to websocket
      (Reported by Moises Silva)
 * ASTERISK-28899 - Upgrade Asterisk to bundled pjproject 2.10
      (Reported by Kevin Harwell)
 * ASTERISK-28895 - res_pjsip_logger: Add tons'o'functionality
      (Reported by Joshua C. Colp)
 * ASTERISK-28896 - ari: Add support for specifying variables on
      channel create
      (Reported by Joshua C. Colp)
 * ASTERISK-28879 - pjproject has race conditions in it's build
      (Reported by Guido Falsi)
 * ASTERISK-28866 - third-party/pjproject/configure.m4 contains
      (Reported by Guido Falsi)
 * ASTERISK-28853 - Missing include on FreeBSD
      by Guido Falsi)
 * ASTERISK-28832 - chan_mobile creates PCMA streams that make
      some VoIP clients crash or not render received audio
      (Reported by Peter Turczak)
 * ASTERISK-28813 - func_volume: Allow decimal numbers as
      parameter to improve granularity
      (Reported by Jean Aunis -
 * ASTERISK-28777 - Codec Negotiation: add
      outgoing_call_offer_prefs option
      (Reported by Kevin
 * ASTERISK-27946 - dial (API): Storage of dialed target uses
      AST_MAX_EXTENSION when it shouldn't
      (Reported by Joshua
 * ASTERISK-28782 - Add support for Content-Disposition header
      in multi-part INVITES
      (Reported by Torrey Searle)
 * ASTERISK-28787 - res_pjsip_session: Decide more intelligently
      when to add video
      (Reported by Joshua C. Colp)
 * ASTERISK-28756 - Codec Negotiation: add
      incoming_call_offer_pref option
      (Reported by Kevin
 * ASTERISK-28750 - TLS/SSL Key too small error
      by Martin Zeh)
 * ASTERISK-28733 - stream: Add support for adding/removing
      streams during SFU/calls
      (Reported by Joshua C. Colp)
 * ASTERISK-24798 - Documentation - Clarify That Format Is Set
      By File Name Extension In MixMonitor
      (Reported by xrobau)
 * ASTERISK-28726 - install_prereq script uses the interactive
      mode when installing aptitude
      (Reported by Sylvain
 * ASTERISK-28710 - Should be able to disable the /httpstatus
      URI in the built-in HTTP server
      (Reported by Sean Bright)
 * ASTERISK-28484 - Add AudioSocket support
      (Reported by
      Se��n C. McCord)
 * ASTERISK-28638 - Simplify dialplan for Dial, Page, and
      (Reported by cmaj)
 * ASTERISK-28673 - GET FULL VARIABLE documentation
      (Reported by Jonathan Harris)
 * ASTERISK-28629 - [patch] Add an "inhibitCOLP" flag to the
      bridges REST API
      (Reported by Jean Aunis - Prescom)
 * ASTERISK-28658 - app_confbridge: Add support for setting
      maximum sample rate
      (Reported by Joshua C. Colp)
 * ASTERISK-28602 - res_pjsip_outbound_registration: Maximum
      retries reached
      (Reported by Daniel)
 * ASTERISK-28586 - Typo in README-SERIOUSLY.bestpractices.md
      (Reported by Sam Banks)
 * ASTERISK-22192 - [patch] Allow voicemail forwards with ODBC
      backend when format differs from attachfmt column
      (Reported by cmaj)
 * ASTERISK-28567 - Problem with ASTERISK-20207: Asterisk should
      clear out any .lock files in the voice mail directory on
      (Reported by Michael)
 * ASTERISK-28542 - [patch] add the ability for asterisk to
      generate on-hold re-invites
      (Reported by Torrey Searle)
 * ASTERISK-28512 - Add pass-through support for H.265 (HEVC)
      (Reported by Florian Floimair)
 * ASTERISK-28443 - app_voicemail: remove dependency on stasis
      (Reported by Kevin Harwell)
 * ASTERISK-28442 - stasis_state: Create a stasis module to
      cache last known state
      (Reported by Kevin Harwell)
 * ASTERISK-28385 - res_ari_channels: Added detail hangup code
      (Reported by sungtae kim)
 * ASTERISK-28234 - pbx_dundi: Add IPv4/IPv6 dual bind support
      for DUNDi
      (Reported by Kirsty Tyerman)
 * ASTERISK-28401 - app_confbridge: Add *_all remb behavior
      (Reported by Joshua C. Colp)
 * ASTERISK-28400 - res_rtp_asterisk / res_pjsip_sdp_rtp: Add
      support for transport-cc
      (Reported by Joshua C. Colp)
 * ASTERISK-28363 - Millisecond-resolution call stats including
      PDD in channel variables
      (Reported by Antoni Goldstein)
 * ASTERISK-28378 - Added detail subscriber/subscription info
      for stasis show app cli
      (Reported by sungtae kim)
 * ASTERISK-20207 - Asterisk should clear out any .lock files in
      the voice mail directory on startup.
      (Reported by Steven
 * ASTERISK-28111 - build: CHANGES/UPGRADE are irritating to
      work with.
      (Reported by Corey Farrell)
 * ASTERISK-28264 - Added topic_all container
      (Reported by
      sungtae kim)
 * ASTERISK-28343 - Added app_name, app_data to channel type
      (Reported by sungtae kim)
 * ASTERISK-28326 - ari: Added timestamp for some ari events.
      (Reported by sungtae kim)
 * ASTERISK-28317 - Add logical group at DAHDIChannel event and
      create "dahdi_group" at CHANNEL function
      (Reported by
      Cirillo Ferreira)
 * ASTERISK-28279 - Added creation timestamp for bridge
      (Reported by sungtae kim)
 * ASTERISK-27483 - Allow wrapuptime to be set for each queue
      (Reported by Rodrigo Ramirez Norambuena)
 * ASTERISK-28055 - app_queue: Per-member wrapup time missing
      from AddQueueMember application
      (Reported by Niksa Baldun)
 * ASTERISK-28292 - Changed to show all channel stats including
      wrong media
      (Reported by sungtae kim)
 * ASTERISK-28253 - res_pjsip_session: Adding rtcp stats result
      into the session
      (Reported by sungtae kim)
 * 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)
 * ASTERISK-28144 - [patch] New function PJSIP_PARSE_URI to
      parse an URI and return a specified part of the URI
      (Reported by Alexei Gradinari)
 * ASTERISK-28136 - Allow the sip_to_pjsip script to be used in
      a pipe
      (Reported by Pascal Cadotte Michaud)
 * ASTERISK-28046 - Remove stale nonoptreq references
      (Reported by Walter Doekes)
 * ASTERISK-27164 - [patch] Add IPv6 Support for DUNDi
      (Reported by Adam Secombe)
 * ASTERISK-28006 - PJSIP: Missing
      "party=calling"/"party=called" in Remote-Party-ID
      (Reported by Eric Dantie)
 * ASTERISK-27995 - pjproject_bundled: Find shared libraries in
      root --with-ssl=PATH.
      (Reported by Alexander Traud)
 * ASTERISK-27993 - pjsip_wizard example gives wrong info about
      unsupported SRV records
      (Reported by Jonathan Harris)
 * ASTERISK-27970 - res_rtp_asterisk: T.140 packets containing
      backspace or end of line are merged with regular text and it
      causes some UA to break
      (Reported by Emmanuel BUU)

For a full list of changes in this release, please see the ChangeLog:

Thank you for your continued support of Asterisk!
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