The Asterisk Development Team would like to announce the release of Asterisk 
19.0.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

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

Thank you!

The following issues are resolved in this release:

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

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

Security bugs fixed in this release:
-----------------------------------
 * ASTERISK-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
      load
      (Reported by Sandro Gauci)

New Features made in this release:
-----------------------------------
 * 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)

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
      Kern)
 * ASTERISK-29673 - app_read: Fix null pointer crash regression

      (Reported by N A)
 * ASTERISK-29671 - res_rtp_asterisk: memory leak
     
      (Reported by Jean Aunis - Prescom)
 * ASTERISK-29668 - ari: Listing bridges fails when dialing
      bridge exists
      (Reported by Joshua C. Colp)
 * ASTERISK-29663 - messaging: AMI MessageSend does not support
      same parameters as dialplan application
      (Reported by Brian
      J. Murrell)
 * ASTERISK-29578 - app_queue: Custom device state using
      included hints do not update
      (Reported by N A)
 * ASTERISK-29660 - Build failure when disabling PJSIP support
 
      (Reported by Guido Falsi)
 * ASTERISK-29654 - pjproject includes trailing whitespace in
      sdp format attributes
      (Reported by George Joseph)
 * ASTERISK-29635 - MP3Player don' t work with actual mpg123
      versions
      (Reported by Carlos Oliva)
 * ASTERISK-29629 - ARI external media channel creation doesn't
      set option data
      (Reported by sungtae kim)
 * ASTERISK-27176 - test_abstract_jb: frames leak
     
      (Reported by Corey Farrell)
 * ASTERISK-29634 - res_snmp:  gcc 11 needs -fPIC to compile
      correctly
      (Reported by George Joseph)
 * ASTERISK-29630 - Asterisk is unable to read extended number
      format terminfo files
      (Reported by Sean Bright)
 * ASTERISK-28004 - dns: Core ast_dns_get_nameservers does not
      support configured IPv6 servers
      (Reported by Isaac
      McDonald)
 * ASTERISK-29618 - ConfBridge errors on creation conference
      room
      (Reported by Alexander Zharov)
 * ASTERISK-29622 - ARI: external media create doesn't use body
      parameter
      (Reported by sungtae kim)
 * ASTERISK-29614 - app_agent_pool: XML Doc: unterminated entity
      reference
      (Reported by Alexander Traud)
 * ASTERISK-29609 - Subsequent 'ael reload' will cause a lock
      up
      (Reported by Mark Murawski)
 * ASTERISK-28701 - app_queue: Core reload resets queue stats,
      even when keepstats=yes
      (Reported by Luke Escude)
 * ASTERISK-29616 - res_rtp_asterisk: sqrt(.) requires the
      header math.h.
      (Reported by Alexander Traud)
 * ASTERISK-29518 - sig_analog: FCG_CAMA fails to signal ANI
      spill when using MF signaling
      (Reported by Sarah Autumn)
 * ASTERISK-29582 - res_pjproject: Can't map pjproject log
      messages to Asterisk TRACE
      (Reported by George Joseph)
 * ASTERISK-29575 - app_milliwatt: Milliwatt application doesn't
      use the proper timings
      (Reported by N A)
 * ASTERISK-20339 - chan_mgcp, resp_pktccops ast_debug support
 
      (Reported by Tomas Maldonado)
 * ASTERISK-29540 - aelparse: include of context with timings
      fails
      (Reported by Alexander Traud)
 * ASTERISK-29539 - Segmentation fault at ast_writestream() when
      write handler not defined (happens with OGG/Speex)
     
      (Reported by Ernani José 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-27871 - Remote URL in playback must end with file
      extension
      (Reported by Caesar)
 * ASTERISK-29507 - STUN timeout is silently delaying calls
    
      (Reported by Sébastien Duthil)
 * 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-29372 - file.c switch does not account for flash
      events
      (Reported by N A)
 * 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-29358 - chan_pjsip: Trace message for progress is
      output even if frame is not queued
      (Reported by Michael
      Maier)
 * ASTERISK-29030 - res_rtp_asterisk: Additional RTP-frame (with
      wrong SSRC) gets inserted when switching from progress to
      established
      (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
      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-26614 - app_queue: updatecdr option in queues.conf
      does effectively nothing
      (Reported by Alexander Gonchiy)
 * ASTERISK-24631 - Incorrect description of option "context" in
      queues.conf.sample
      (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
      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-28452 - pjsip: <sess-version> of SDP is not
      incremented though SDP may be changed on reinvite without SDP
      offer
      (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
      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-29259 - channel: 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-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
      Cripps)
 * 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
      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-29240 - chan_pjsip: Incoming PJSIP calls set global
      SIPDOMAIN instead of a channel variable
      (Reported by Ivan
      Poddubny)
 * 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-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
      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-29146 - GCC Warnings: ‘%s’ directive argument is
      null.
      (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
      VMSayName
      (Reported by Eric Smith)
 * 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-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 周家建)
 * ASTERISK-29085 - func_curl: Segmentation fault when using
      CURL after setting httpheader CURLOPT
      (Reported by Péter
      Juhász)
 * ASTERISK-24329 - Music On Hold announcement cuts intro of
      music the first time it is played
      (Reported by Thomas
      Frederiksen)
 * 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-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-29034 - Lastpause of realtime members is reseting
  
      (Reported by Evandro César 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
      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)

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
      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 Sébastien Duthil)
 * ASTERISK-29612 - bridge_basic: Don't throw warning if
      attended transfer is cancelled
      (Reported by N A)
 * ASTERISK-29544 - Media Cache - Delayed remote sound file
      retrieve delays all playbacks
      (Reported by Andre Barbosa)
 * ASTERISK-29495 - Return integer instead of float if response
      is a whole number
      (Reported by N A)
 * ASTERISK-29541 - app_morsecode: Add American Morse code
     
      (Reported by N A)
 * ASTERISK-29543 - app_originate: Allow specifying codec(s) to
      use
      (Reported by N A)
 * ASTERISK-29528 - Add support for multiple files for agent
      announcements
      (Reported by N A)
 * ASTERISK-29527 - res_http_media_cache: Cleanup audio format
      lookup in HTTP requests
      (Reported by Sean Bright)
 * 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-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
      packets
      (Reported by Jeremy Lainé)
 * ASTERISK-29380 - Add Flash AMI event to handle flash events
 
      (Reported by N A)
 * ASTERISK-29349 - Silent voicemail option is not completely
      silent
      (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-29335 - xml: Embed module information into core XML
      documentation.
      (Reported by Joshua C. Colp)
 * ASTERISK-29336 - documentation: Fix inconsistent support
      levels
      (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 Sébastien 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-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, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0

Thank you for your continued support of Asterisk!
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