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

Asterisk Development Team asteriskteam at digium.com
Thu Apr 28 08:43:51 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
https://downloads.asterisk.org/pub/telephony/certified-asterisk

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
      19
      (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
      19
      (Reported by Joshua C. Colp)
 * ASTERISK-29560 - app_url: Deprecated in 16, to be removed in
      19
      (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
      19
      (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
      21
      (Reported by Joshua C. Colp)
 * ASTERISK-29568 - chan_nbs: Deprecated in 16, to be removed in
      19
      (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
      19
      (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
      19
      (Reported by Joshua C. Colp)
 * ASTERISK-29574 - muted: Deprecated in 16, to be removed in
      19
      (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
      load
      (Reported by Sandro Gauci)
 * ASTERISK-28589 - chan_sip: Depending on configuration an
      INVITE can alter Addr of a peer
      (Reported by Andrey  V.
      T.)
 * ASTERISK-28580 - Bypass SYSTEM write permission in manager
      action allows system commands execution
      (Reported by Eliel
      Sarda��ons)
 * ASTERISK-28495 - res_pjsip_t38: 200 OK with SDP answer with
      declined stream causes crash
      (Reported by Alexei
      Gradinari)
 * 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
      reINVITE
      (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
      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-11 - AGI channel_status failure
      (Reported by
      bbawkon)
 * ASTERISK-29477 - Function to asynchronously store digits
      dialed
      (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
      header
      (Reported by Martin Tomec)
 * ASTERISK-28533 - func_jitterbuffer: Add support for video
      synchronization
      (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
      /ari/channels/{channelid}/rtp_statistics
      (Reported by
      sungtae kim)
 * ASTERISK-28267 - res_stasis: Add ability to switch
      applications
      (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
      Searle)
 * 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
      honored
      (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
      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-29764 - chan_misdn: Fix for Doxygen
      (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-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
      (Reported
      by Alexander Traud)
 * ASTERISK-29737 - chan_iax2: Fix for Doxygen
      (Reported
      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
      labels
      (Reported by Alexander Traud)
 * ASTERISK-29734 - progdocs: Use Doxygen \example correctly
   
      (Reported by Alexander Traud)
 * ASTERISK-29744 - app_morsecode: Fix deadlock
      (Reported
      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
      distributions.
      (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-29691 - stun: Not all users provide a dst to
      ast_stun_request
      (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
      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-29635 - MP3Player don' t work with actual mpg123
      versions
      (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
      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)
 * ASTERISK-29507 - STUN timeout is silently delaying calls
    
      (Reported by S��bastien Duthil)
 * ASTERISK-27871 - Remote URL in playback must end with file
      extension
      (Reported by Caesar)
 * ASTERISK-29514 - ari: Audiosocket segfault when no data
      specified
      (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.
      Colp)
 * ASTERISK-29479 - [patch] Channels are not put on hold for
      Session Progress with inactive audio
      (Reported by Bernd
      Zobl)
 * 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
      Chris)
 * ASTERISK-29397 - pjsip: Asterisk isn't tolerant of RFC8760
      UASs
      (Reported by George Joseph)
 * ASTERISK-24601 - [patch]Missing RFC4235 tags and attributes
      in PJSIP NOTIFY event: dialog  XML body
      (Reported by Marco
      Paland)
 * ASTERISK-29370 - chan_sip does not recognize
      application/hook-flash
      (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
      events
      (Reported by N A)
 * ASTERISK-29358 - chan_pjsip: Trace message for progress is
      output even if frame is not queued
      (Reported by Michael
      Maier)
 * 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
      established
      (Reported by Matthias Hensler)
 * ASTERISK-29328 - translate.c: possible buffer overflow when
      upsampling
      (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
      miscalculation 
      (Reported by Kevin Harwell)
 * ASTERISK-29373 - res_rtp_asterisk: Flash events are
      duplicated
      (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
      queues.conf.sample
      (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
      faxing
      (Reported by Matthias Hensler)
 * ASTERISK-29354 - res_pjsip: Allow partial reloading of
      transports
      (Reported by Joshua C. Colp)
 * ASTERISK-29348 - menuselect doesn't return errors in many
      cases
      (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
      (Reported
      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
      Damm)
 * 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
      session
      (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
      French)
 * 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
      Gradinari)
 * ASTERISK-16799 - Callee declined when 'beep' audio file does
      not exist
      (Reported by IAMJames_)
 * ASTERISK-29313 - res_pjsip_refer:  Segfault in progress
      notify
      (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
      offer
      (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
      Vidmar)
 * ASTERISK-29196 - res_pjsip: Segmentation fault
     
      (Reported by Mauri de Souza Meneguzzo (3CPlus))
 * ASTERISK-29280 - chan_sip: Allow peers without audio
      (text+video).
      (Reported by Alexander Traud)
 * ASTERISK-29265 - chan_sip: Allow text+video media streams,
      again.
      (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,
      again.
      (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
      Cripps)
 * 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
      Poddubny)
 * 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
      disabled.
      (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
      responses
      (Reported by George Joseph)
 * ASTERISK-28016 - PJSIP sends duplicate 183 Progress
      responses
      (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
      registered
      (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
      counts
      (Reported by Sean Bright)
 * ASTERISK-29173 - Media cache URL requests allow infinite
      redirects
      (Reported by Sean Bright)
 * ASTERISK-29175 - res_pjsip_stir_shaken: Fix module
      description
      (Reported by Stanislav Abramenkov)
 * ASTERISK-29148 - AST_MODULE_INFO no, MODULEINFO depend
     
      (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
      server.
      (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
      Doekes)
 * 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
      Damm)
 * ASTERISK-29142 - sip_to_pjsip.py: doesn't read globbed
      includes
      (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
      null.
      (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
      VMSayName
      (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
      (Reported
      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
      Frederiksen)
 * ASTERISK-29085 - func_curl: Segmentation fault when using
      CURL after setting httpheader CURLOPT
      (Reported by P��ter
      Juh��sz)
 * 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
      slin
      (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
      events
      (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
      copied
      (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
      responses
      (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
      command
      (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
      reload
      (Reported by Dennis)
 * ASTERISK-29021 - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on
      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
      external_media_address)
      (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
      Doekes)
 * 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
      dahdi-channels.conf
      (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
      channel
      (Reported by Walter Doekes)
 * ASTERISK-28644 - Stale comment in app_queue about ring_entry
      exception
      (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.
      Colp)
 * 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
      Massel)
 * ASTERISK-28929 - pjproject_bundled: Honor
      --without-pjproject.
      (Reported by Alexander Traud)
 * ASTERISK-28932 - res_pjsip_logger writing too big packets
   
      (Reported by nappsoft)
 * ASTERISK-28920 - bridge show all causes crash
      (Reported
      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
      printing
      (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
      transport=transport-udp6
      (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
      truncation.
      (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
      secure_bridge_signaling
      (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
      (TCP)
      (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)
 * ASTERISK-28838 - AST_MODULE_INFO requires, MODULEINFO does
      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
      --without-pjproject.
      (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
      Traud)
 * 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
      Traud)
 * 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
      writes
      (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
      Kroon)
 * 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.
      Colp)
 * 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
      Hold
      (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
      refreshes
      (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
      Segfaults
      (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
      (Reported
      by Ross Beer)
 * ASTERISK-28679 - stasis application is destroyed after its
      creation
      (Reported by Francois Blackburn)
 * ASTERISK-25421 - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in
      spite of the error when sending
      (Reported by Dmitriy
      Serov)
 * 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
      calls
      (Reported by Maciej Michno)
 * ASTERISK-28702 - chan_dahdi: holding a channel via flash to
      dialtone times out after 0:16:40
      (Reported by Andrew
      Siplas)
 * 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
      Meggelen)
 * 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
      event
      (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
      AvayaXAsterisk)
 * ASTERISK-28682 - app_record: Lack of `beep` audio file causes
      application to return error and hangup
      (Reported by Corey
      Farrell)
 * 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
      Wendland)
 * 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
      sip_to_pjsip.py
      (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
      documentation
      (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
      ptime.
      (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
      received  
      (Reported by Salah Ahmed)
 * ASTERISK-28624 - res_pjsip_outbound_registration: add SRV
      failover
      (Reported by Kevin Harwell)
 * ASTERISK-28608 - app_amd: Use time calculation to calculate
      timeout
      (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
      Elson)
 * ASTERISK-26481 - FILE function grabs garbage along with read
      data when target line has no newline
      (Reported by Jonathan
      Harris)
 * 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
      res_calendar_icalendar
      (Reported by Yoooooo Ha)
 * ASTERISK-28585 - ari/resource_events: Crash in event session
      cleanup
      (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
      command
      (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
      16.5
      (Reported by Niklas Larsson)
 * ASTERISK-28561 - Asterisk Deadlocks
      (Reported by
      Aheliotech)
 * 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
      call(s)
      (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
      (Reported
      by Cyril Rami��re)
 * ASTERISK-28536 - Asterisk release candidates fail to build on
      FreeBSD
      (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
      readsql
      (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
      FOLL)
 * 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
      re-register
      (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
      (Reported
      by Alexei Gradinari)
 * ASTERISK-28487 - compile menuselect on gentoo
      (Reported
      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
      entries
      (Reported by Ian Jones)
 * ASTERISK-28483 - packet lost on UDPTL wrap around
     
      (Reported by Torrey Searle)
 * ASTERISK-28477 - Crash when not specifying "dbfile" in
      res_config_sqlite3.conf
      (Reported by Dennis)
 * ASTERISK-28478 - Crash performing "core reload" with modified
      res_config_sqlite3.conf
      (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
      logs
      (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
      suppression
      (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
      gatewaying
      (Reported by pasandev)
 * ASTERISK-28427 - new mwi.h include missing from some dahdi
      source files, causes build failure
      (Reported by Guido
      Falsi)
 * ASTERISK-28421 - Wrong type used for timestamp in
      res_rtp_asterisk
      (Reported by Morten Tryfoss)
 * ASTERISK-28161 - Removal of Previous Patch Causes PJSIP Timer
      Issues
      (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
      queuing
      (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
      registrar_find_contact
      (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
      expected
      (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
      enabled
      (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
      Mendes)
 * ASTERISK-27935 - app_voicemail: emailbody per user can't
      contain commas
      (Reported by S��bastien Duthil)
 * ASTERISK-17695 - 1.8.3.2 extenpatternmatchnew=yes cannot find
      extensions with '-' in them
      (Reported by test011)
 * ASTERISK-17799 - AEL reload causes loss of control in a
      macro
      (Reported by Kirill Katsnelson)
 * ASTERISK-18593 - AEL for loops use Macro app and pipe
      delimiter
      (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
      Disabled
      (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
      Seratti)
 * 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
      Odrov)
 * 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
      Michael)
 * 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
      Harris)
 * 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
      applications
      (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
      Searle)
 * ASTERISK-28251 - CI: Fix CI so it reverifies commit message
      changes
      (Reported by George Joseph)
 * ASTERISK-28277 - database: Add some basic logging
     
      (Reported by Joshua C. Colp)
 * ASTERISK-28181 - ari: Originating overwrites channel start
      time
      (Reported by sungtae kim)
 * ASTERISK-28173 - Deadlock in chan_sip handling subscribe
      request during res_parking reload
      (Reported by Giuseppe
      Sucameli)
 * ASTERISK-28104 - AstriCon Feedback:  Automatically create a 1
      line dialplan context for stasis apps
      (Reported by George
      Joseph)
 * ASTERISK-28271 - Opensuse Leap 15 --with-jannson-bundled will
      not compile
      (Reported by David Wilcox)
 * ASTERISK-28238 - PJSIP realtime. getcontext not working with
      DUNDI
      (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
      arm-linux-gnueabihf
      (Reported by Jean Aunis - Prescom)
 * ASTERISK-28252 - HangupHandler manager events are never
      thrown
      (Reported by Gerald Schnabel)
 * ASTERISK-28231 - res_http_websocket: Not responding to
      Connection Close Frame (opcode 8)
      (Reported by Jeremy
      Lain��)
 * ASTERISK-28249 - res_monitor: Segfault with
      Monitor(wav,file,i)
      (Reported by Valentin Vidi��)
 * ASTERISK-28244 - stasis: Filter messages at publishing to
      AMI/ARI
      (Reported by Joshua C. Colp)
 * ASTERISK-28197 - stasis: ast_endpoint struct holds the
      channel_ids of channels past destruction in certain cases
     
      (Reported by Mohit Dhiman)
 * ASTERISK-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
      issue
      (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
      mode
      (Reported by Joshua C. Colp)
 * ASTERISK-28222 - Regression: MWI polling no longer works
    
      (Reported by abelbeck)
 * ASTERISK-28221 - Bug in ast_coredumper
      (Reported by
      Andrew Nagy)
 * ASTERISK-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
      (Reported
      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
      Joseph)
 * ASTERISK-28182 - chan_pjsip: When connected_line_method is
      set to invite, asterisk is not trying UPDATE
      (Reported by
      nappsoft)
 * 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
      dump
      (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
      (Reported
      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
      (Reported
      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
      (Reported
      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
      AMI
      (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
      2.8
      (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
      class
      (Reported by laszlovl)
 * ASTERISK-28049 - res_pjproject build failure
      (Reported
      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
      boolean
      (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
      BUU)
 * 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
      Floimair)
 * 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
      offer
      (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
      LEAVEEMPTY
      (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
      (Reported
      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
      10/gcc-4.1.1
      (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.
      Colp)
 * ASTERISK-27968 - systemd: asterisk.service
      (Reported by
      seanchann.zhou)

Improvements made in this release:
-----------------------------------
 * 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-29501 - ARI - Stasis Playback doesn't hangup call
      when processing a list of invalid files
      (Reported by Andre
      Barbosa)
 * 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
      packets
      (Reported by Jeremy Lain��)
 * ASTERISK-29349 - Silent voicemail option is not completely
      silent
      (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
      modules!
      (Reported by Joshua C. Colp)
 * ASTERISK-29337 - menuselect: Add ability to set deprecated in
      and removed in versions for modules
      (Reported by Joshua C.
      Colp)
 * ASTERISK-29336 - documentation: Fix inconsistent support
      levels
      (Reported by Joshua C. Colp)
 * ASTERISK-29335 - xml: Embed module information into core XML
      documentation.
      (Reported by Joshua C. Colp)
 * ASTERISK-29321 - sorcery: Add support for more intelligent
      reloading.
      (Reported by Joshua C. Colp)
 * ASTERISK-29325 - res_pjsip_registrar: Include source IP
      address and port in log messages
      (Reported by Joshua C.
      Colp)
 * ASTERISK-29326 - asterisk: Update copyright/company
     
      (Reported by Joshua C. Colp)
 * ASTERISK-29244 - Add MixMonitorStart / Stop / Mute AMI
      events
      (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
      Liu)
 * ASTERISK-29216 - contrib: systemd asterisk service for
      centos8 or other newer linux versions
      (Reported by Mark
      Petersen)
 * 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
      Castro)
 * 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
      client
      (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
      system
      (Reported by Guido Falsi)
 * ASTERISK-28866 - third-party/pjproject/configure.m4 contains
      bashisms
      (Reported by Guido Falsi)
 * ASTERISK-28853 - Missing include on FreeBSD
      (Reported
      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 -
      Prescom)
 * ASTERISK-28777 - Codec Negotiation: add
      outgoing_call_offer_prefs option
      (Reported by Kevin
      Harwell)
 * ASTERISK-27946 - dial (API): Storage of dialed target uses
      AST_MAX_EXTENSION when it shouldn't
      (Reported by Joshua
      Elson)
 * 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
      Harwell)
 * ASTERISK-28750 - TLS/SSL Key too small error
      (Reported
      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
      Afchain)
 * 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
      ChanIsAvail
      (Reported by cmaj)
 * ASTERISK-28673 - GET FULL VARIABLE documentation
      clarification
      (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
      startup.
      (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)
      codec
      (Reported by Florian Floimair)
 * ASTERISK-28443 - app_voicemail: remove dependency on stasis
      cache
      (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
      settings
      (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
      variants
      (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
      Wheeler)
 * 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
      member
      (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:
https://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-18.9-cert1

Thank you for your continued support of Asterisk!
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