[asterisk-dev] Asterisk 19.0.0-rc1 Now Available

Asterisk Development Team asteriskteam at digium.com
Wed Oct 13 06:51:14 CDT 2021

The Asterisk Development Team would like to announce the first
release candidate of Asterisk 19.0.0.
This release candidate is available for immediate download at 

The release of Asterisk 19.0.0-rc1 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release candidate:

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

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

Security bugs fixed in this release:
 * ASTERISK-29381 - chan_pjsip: Remote denial of service by an
      authenticated user
      (Reported by Ivan Poddubny)
 * ASTERISK-29415 - Crash in PJSIP TLS transport 
      (Reported by Andrew Yager)
 * 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)

New Features made in this release:
 * 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)

Bugs fixed in this release:
 * 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-29654 - pjproject includes trailing whitespace in
      sdp format attributes
      (Reported by George Joseph)
 * ASTERISK-29635 - MP3Player don' t work with actual mpg123
      (Reported by Carlos Oliva)
 * ASTERISK-29629 - ARI external media channel creation doesn't
      set option data
      (Reported by sungtae kim)
 * ASTERISK-27176 - test_abstract_jb: frames leak
      (Reported by Corey Farrell)
 * ASTERISK-29634 - res_snmp:  gcc 11 needs -fPIC to compile
      (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-27871 - Remote URL in playback must end with file
      (Reported by Caesar)
 * ASTERISK-29507 - STUN timeout is silently delaying calls
      (Reported by S��bastien Duthil)
 * 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-29372 - file.c switch does not account for flash
      (Reported by N A)
 * 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-29358 - chan_pjsip: Trace message for progress is
      output even if frame is not queued
      (Reported by Michael
 * ASTERISK-29030 - res_rtp_asterisk: Additional RTP-frame (with
      wrong SSRC) gets inserted when switching from progress to
      (Reported by Matthias Hensler)
 * ASTERISK-29407 - chan_local: Filtering audio formats should
      not occur on removed streams
      (Reported by Joshua C. Colp)
 * 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-26614 - app_queue: updatecdr option in queues.conf
      does effectively nothing
      (Reported by Alexander Gonchiy)
 * ASTERISK-24631 - Incorrect description of option "context" in
      (Reported by Etienne Lessard)
 * 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-28452 - pjsip: <sess-version> of SDP is not
      incremented though SDP may be changed on reinvite without SDP
      (Reported by Michael Maier)
 * ASTERISK-29311 - res_odbc_transaction sets forcecommit
      default value based on isolation level instead of forcecommit
      (Reported by Jaco Kroon)
 * ASTERISK-29303 - pjsip: Re-invite occurs when it shouldn't
      (Reported by Benjamin Keith Ford)
 * ASTERISK-29293 - res_config_pgsql: Limit realtime_pgsql() to
      return one (no more) record
      (Reported by Boris P. Korzun)
 * ASTERISK-28369 - app_queue: Member device state "invalid"
      when second call is ringing and hint is used
      (Reported by
      Boolah )
 * ASTERISK-29287 - app.h: C++ compatibility broken
      (Reported by Jean Aunis - Prescom)
 * 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-29259 - channel: 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-29258 - chan_sip: Audio stream rejected, Other
      stream present: Invalid SDP.
      (Reported by Alexander Traud)
 * ASTERISK-29248 - res_pjsip_session: res sometimes
      uninitialized reported by compiler Clang.
      (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-29229 - Stasis/messaging: text messages not
      dispatched to all subscribers when using generic subscription
      (Reported by Jean Aunis - Prescom)
 * 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-29240 - chan_pjsip: Incoming PJSIP calls set global
      SIPDOMAIN instead of a channel variable
      (Reported by Ivan
 * 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-28947 - Segmentation fault in mixmonitor_ds_destroy

      (Reported by Robert Sutton)
 * ASTERISK-29201 - Crash occurs when Transfer and execute
      Hangup before the Transfer result 
      (Reported by Dan Cropp)
 * 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-29209 - Debug messages printed by scope trace might
      be missing newlines
      (Reported by Alexander Traud)
 * ASTERISK-29024 - pjsip: Route Header in Cancel request
      incorrectly set
      (Reported by Flole Systems)
 * 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-29146 - GCC Warnings: ���%s��� directive argument is
      (Reported by Alexander Traud)
 * ASTERISK-29145 - GCC Warnings with OPTIMIZE=-Os make
      (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-29097 - res_pjsip_config_wizard: Crash when freeing
      string when failing to add extension
      (Reported by Vieri)
 * ASTERISK-26424 - app_voicemail: Undocumented behavior from
      (Reported by Eric Smith)
 * 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-29091 - Crash when ast_translator_build_path fails
      (Reported by Jasper van der Neut)
 * ASTERISK-28311 - dsp: ast_dsp_silence_noise_with_energy wrong
      judgment of frame format
      (Reported by ���������)
 * ASTERISK-29085 - func_curl: Segmentation fault when using
      CURL after setting httpheader CURLOPT
      (Reported by P��ter
 * ASTERISK-24329 - Music On Hold announcement cuts intro of
      music the first time it is played
      (Reported by Thomas
 * 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-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-29034 - Lastpause of realtime members is reseting
      (Reported by Evandro C��sar Arruda)
 * 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)

Improvements made in this release:
 * 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-29527 - res_http_media_cache: Cleanup audio format
      lookup in HTTP requests
      (Reported by Sean Bright)
 * 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-29460 - Recognize application/hook-flash in PJSIP
      (Reported by N A)
 * ASTERISK-29459 - Missing configuration from PJSIP to SIP
      conversion script
      (Reported by N A)
 * ASTERISK-29434 - Asterisk reveals pjproject version in STUN
      (Reported by Jeremy Lain��)
 * ASTERISK-29380 - Add Flash AMI event to handle flash events
      (Reported by N A)
 * ASTERISK-29349 - Silent voicemail option is not completely
      (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-29335 - xml: Embed module information into core XML
      (Reported by Joshua C. Colp)
 * ASTERISK-29336 - documentation: Fix inconsistent support
      (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-29083 - Do not build chan_sip by default as it is
      now deprecated
      (Reported by Sean Bright)
 * 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)

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

Thank you for your continued support of Asterisk!
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